From royce3 at gmail.com  Sun Mar  1 09:45:24 2015
From: royce3 at gmail.com (Royce Mitchell III)
Date: Sun, 1 Mar 2015 00:45:24 -0600
Subject: [Freeswitch-users] originating a call from event socket
In-Reply-To: <CACOYK=gchwSXqaj1soKqOwONG1BstjQ6izb7yWB3x_PYi=mfHg@mail.gmail.com>
References: <CAHwzD9kieX4Yzv1vk0Z3sasRyGQadTHUu=41aieiaoK23dB2Gw@mail.gmail.com>
	<CACOYK=gchwSXqaj1soKqOwONG1BstjQ6izb7yWB3x_PYi=mfHg@mail.gmail.com>
Message-ID: <CAHwzD9n853bXqoEMgxTG4uz=+O=77OY9sb7DTu0tyD+6_vA7ng@mail.gmail.com>

okay, I got some ideas from your sample scripts, thanks!

My challenge is the agent is already on a call in a callcenter queue. so
initiating a new call to the agent isn't going to work.

I just tried this, and it looks like it works:

<new-uuid> = create_uuid
originate {origination_uuid=<new-uuid>}loopback/<agent> &park()
uuid_setvar <new-uuid> toll_allow domestic,local
uuid_transfer <new-uuid> <destination>

next step, which I will test tomorrow, will be to add this to the above:

uuid_setvar <agent-uuid> transfer_after_bridge 7777
bridge <agent-uuid> <new-uuid>

I hear that loopback is evil. I will analyze the cdr and see just how evil
it is. Is there a cleaner way to do this? 7777 is supposed to put the agent
back into the call queue when the call ends, which is working fine for
inbound calls




Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com

On Sat, Feb 28, 2015 at 1:41 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> here's my script that does this:
> https://github.com/voxserv/freeswitch-helper-scripts/tree/master/esl
>
> It does:
> uuid_create
> originate with &park()
> uuid_transfer
>
>
>
> On Sat, Feb 28, 2015 at 1:38 AM, Royce Mitchell III <royce3 at gmail.com>
> wrote:
> > I need to initiate a call on an agent's behalf, but I can't seem to get
> it
> > to work.
> >
> > This is the closest I've gotten ( 7777 puts agent back in the call-queue
> )
> >
> > uuid_setvar <agent-uuid> transfer_after_bridge 7777
> >
> > uuid_transfer <agent-uuid> <pstn-number>
> >
> > I have 2 problems:
> >
> > 1) the transfer_after_bridge isn't happening, or it's not going to the
> right
> > place
> >
> > 2) I have to wait forever to get a uuid to control that call.
> >
> > I want to be able to allocate a uuid for the outbound leg before I
> initiate
> > the call, but I can't figure out the syntax to make it work.
> >
> > I've tried this, but I get a fast busy:
> >
> > uuid_transfer <agent-uuid> {origination_uuid=<new-uuid>}<pstn-number>
> >
> > I've also tried this and it doesn't work either:
> >
> > originate {origination_uuid=<new-uuid>}sofia/external/<pstn-number>
> > &bridge(<agent-extension>)
> >
> > Please help, thanks
> >
> >
> > 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://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/59f535b7/attachment-0001.html 

From idokan at gmail.com  Sun Mar  1 11:19:57 2015
From: idokan at gmail.com (ik)
Date: Sun, 1 Mar 2015 10:19:57 +0200
Subject: [Freeswitch-users] supporting multiple dtmf protocols
In-Reply-To: <54F0682C.7090603@freeswitch.org>
References: <CAOccsKDpCDX8_fcee8K5PpHvuHF-+ZGzb5ujnO-Ustay3PWXoQ@mail.gmail.com>
	<54F0682C.7090603@freeswitch.org>
Message-ID: <CAOccsKCBju2qM8Nfpq5BzKNshhyVNDe4mUxFMdYjY7HXma9OVw@mail.gmail.com>

On Fri, Feb 27, 2015 at 2:50 PM, I put the Who? in Mishehu <
mishehu at freeswitch.org> wrote:

>  I see after all these years it's still the popular thing to think that
> Bezeq is the most terrible thing in the world.  :-)  Be thankful you don't
> have to deal with some American incumbent providers...  Bezeq can be a walk
> in the park...  :-)
>

You mean like answering with 200 OK on INVITE, getting 404 instead of
"authentication require", dropping requests because I offer both Alaw and
Ulaw and other horror stories around other SIP providers that I have worked
with ? :P


>
> DTMF handling can be a fun little source of headache in general.
>
> To clarify, are you saying that midway through a call that the DTMF mode
> changes?  Or are you saying that the DTMF mode for one call may be 2833 and
> the next call can randomly be inband audio?  If you are saying that the
> mode changes during the call, what indication do you receive that this has
> happened?
>

Different types of payloads in RTP, you can see it also in wireshark when
capturing with tcpdump




>
> You can try the "liberal-dtmf" setting and see if that fixes the issue for
> you, but I believe that only allows SIP NOTIFY and 2833, and I don't
> believe it handles inband.  (Hopefully somebody else will correct me if I'm
> mistaken.)
>

I'll try this
Thank you

Ido


> --
> Yossi Neiman
>
> On 02/27/2015 05:08 AM, ik wrote:
>
>     Hello,
>
>  I have the misfortune of forced to use a telco named Bezeq - the biggest
> telco in Israel.
>  The SIP trunk they provide is very problematic.
>
>  The one that I cannot overcome is that sometimes doing the call, it
> switches DTMF between Inband and rfc2833, and sometimes the whole DTMF
> sending is either inband or rfc2833 doing the entire call.
>
>  I cannot make them to be more stable in this matter (I have tried talking
> with them), and I cannot replace them (I have tried to do so as well).
>
> How can I deal with such mess ?
>
>  Thanks,
> Ido
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/8ce7b2bf/attachment.html 

From groysem at gmail.com  Sun Mar  1 15:20:56 2015
From: groysem at gmail.com (Shai Perelman)
Date: Sun, 1 Mar 2015 14:20:56 +0200
Subject: [Freeswitch-users] Announce position in queue
Message-ID: <CAJfexNE-kc8rP58u3Z997uf3qs_-HVBY2fbGuOk47dy6aQgrDQ@mail.gmail.com>

I found this code for adding position annoucement ability to fusionpbx.
https://code.google.com/p/fusionpbx/issues/detail?id=658

can some one point me the steps to integrate it to my fpbx installation?

it looks like the code files is replacing some original code , is that good?
what about version updates, isnt it going to overwrite it?
are there other , better options to acheive this?

Im new to this so forgive my newbie questions,
I find this list to be the only solution trying to learn fs and fusion, as
the documentation is very limited.
Thanks
Shai
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/6009c646/attachment.html 

From max at nysolutions.com  Sun Mar  1 17:28:16 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 1 Mar 2015 14:28:16 +0000
Subject: [Freeswitch-users] Announce position in queue
In-Reply-To: <CAJfexNE-kc8rP58u3Z997uf3qs_-HVBY2fbGuOk47dy6aQgrDQ@mail.gmail.com>
References: <CAJfexNE-kc8rP58u3Z997uf3qs_-HVBY2fbGuOk47dy6aQgrDQ@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315CD12C8@ex2.connex.local>

For FusionPBX you would be best served using their IRC channel or their Google Code issues.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shai Perelman
Sent: Sunday, March 1, 2015 7:21 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Announce position in queue

I found this code for adding position annoucement ability to fusionpbx.
https://code.google.com/p/fusionpbx/issues/detail?id=658

can some one point me the steps to integrate it to my fpbx installation?

it looks like the code files is replacing some original code , is that good?
what about version updates, isnt it going to overwrite it?
are there other , better options to acheive this?

Im new to this so forgive my newbie questions,
I find this list to be the only solution trying to learn fs and fusion, as the documentation is very limited.
Thanks
Shai
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/04ad37fb/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/04ad37fb/attachment-0001.jpg 

From brian at freeswitch.org  Sun Mar  1 17:32:19 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2015 08:32:19 -0600
Subject: [Freeswitch-users] FreeSWITCH Cookbook Free Today
Message-ID: <CAEJMVkD+CbOXdQG=fX30EAHL+c_zYQuH0FzKt7nABXWzu4Jz1Q@mail.gmail.com>

https://twitter.com/packtpub/status/572041101076381696

Go get it!

Start your FreeSWITCHing today!

/b


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/b01467d8/attachment.html 

From brian at freeswitch.org  Sun Mar  1 17:52:52 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2015 08:52:52 -0600
Subject: [Freeswitch-users] supporting multiple dtmf protocols
In-Reply-To: <CAOccsKCBju2qM8Nfpq5BzKNshhyVNDe4mUxFMdYjY7HXma9OVw@mail.gmail.com>
References: <CAOccsKDpCDX8_fcee8K5PpHvuHF-+ZGzb5ujnO-Ustay3PWXoQ@mail.gmail.com>
	<54F0682C.7090603@freeswitch.org>
	<CAOccsKCBju2qM8Nfpq5BzKNshhyVNDe4mUxFMdYjY7HXma9OVw@mail.gmail.com>
Message-ID: <CAEJMVkBmYLed01oqJG-gFeXDchGwZtqXEiteU4ms4wJFb374_g@mail.gmail.com>

Why do we keep letting these providers get away with this? Smh

On Sunday, March 1, 2015, ik <idokan at gmail.com> wrote:

>
>
> On Fri, Feb 27, 2015 at 2:50 PM, I put the Who? in Mishehu <
> mishehu at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','mishehu at freeswitch.org');>> wrote:
>
>>  I see after all these years it's still the popular thing to think that
>> Bezeq is the most terrible thing in the world.  :-)  Be thankful you don't
>> have to deal with some American incumbent providers...  Bezeq can be a walk
>> in the park...  :-)
>>
>
> You mean like answering with 200 OK on INVITE, getting 404 instead of
> "authentication require", dropping requests because I offer both Alaw and
> Ulaw and other horror stories around other SIP providers that I have worked
> with ? :P
>
>
>>
>> DTMF handling can be a fun little source of headache in general.
>>
>> To clarify, are you saying that midway through a call that the DTMF mode
>> changes?  Or are you saying that the DTMF mode for one call may be 2833 and
>> the next call can randomly be inband audio?  If you are saying that the
>> mode changes during the call, what indication do you receive that this has
>> happened?
>>
>
> Different types of payloads in RTP, you can see it also in wireshark when
> capturing with tcpdump
>
>
>
>
>>
>> You can try the "liberal-dtmf" setting and see if that fixes the issue
>> for you, but I believe that only allows SIP NOTIFY and 2833, and I don't
>> believe it handles inband.  (Hopefully somebody else will correct me if I'm
>> mistaken.)
>>
>
> I'll try this
> Thank you
>
> Ido
>
>
>> --
>> Yossi Neiman
>>
>> On 02/27/2015 05:08 AM, ik wrote:
>>
>>     Hello,
>>
>>  I have the misfortune of forced to use a telco named Bezeq - the biggest
>> telco in Israel.
>>  The SIP trunk they provide is very problematic.
>>
>>  The one that I cannot overcome is that sometimes doing the call, it
>> switches DTMF between Inband and rfc2833, and sometimes the whole DTMF
>> sending is either inband or rfc2833 doing the entire call.
>>
>>  I cannot make them to be more stable in this matter (I have tried
>> talking with them), and I cannot replace them (I have tried to do so as
>> well).
>>
>> How can I deal with such mess ?
>>
>>  Thanks,
>> Ido
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>http://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
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/a9898a9f/attachment.html 

From gmaruzz at gmail.com  Sun Mar  1 17:54:02 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 1 Mar 2015 15:54:02 +0100
Subject: [Freeswitch-users] FreeSWITCH Cookbook Free Today
In-Reply-To: <CAEJMVkD+CbOXdQG=fX30EAHL+c_zYQuH0FzKt7nABXWzu4Jz1Q@mail.gmail.com>
References: <CAEJMVkD+CbOXdQG=fX30EAHL+c_zYQuH0FzKt7nABXWzu4Jz1Q@mail.gmail.com>
Message-ID: <CALXCt0qrdDbVus2pnrHpu_YF7czfh35xgNfzTMOfdFxOATTOaw@mail.gmail.com>

Fast!
Until it last!!!

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Mar 1, 2015 3:32 PM, "Brian West" <brian at freeswitch.org> wrote:

> https://twitter.com/packtpub/status/572041101076381696
>
> Go get it!
>
> Start your FreeSWITCHing today!
>
> /b
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/28dafcb3/attachment.html 

From paul.atreides83 at googlemail.com  Sun Mar  1 18:05:09 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Sun, 1 Mar 2015 16:05:09 +0100
Subject: [Freeswitch-users] Early Dial / Sip 484
Message-ID: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>

Hi,

does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it up
in the dialplan?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/7e473bb2/attachment.html 

From brian at freeswitch.org  Sun Mar  1 18:05:45 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2015 09:05:45 -0600
Subject: [Freeswitch-users] FreeSWITCH Cookbook Free Today
In-Reply-To: <CALXCt0qrdDbVus2pnrHpu_YF7czfh35xgNfzTMOfdFxOATTOaw@mail.gmail.com>
References: <CAEJMVkD+CbOXdQG=fX30EAHL+c_zYQuH0FzKt7nABXWzu4Jz1Q@mail.gmail.com>
	<CALXCt0qrdDbVus2pnrHpu_YF7czfh35xgNfzTMOfdFxOATTOaw@mail.gmail.com>
Message-ID: <CAEJMVkCzx1-L_aRk0XrZfrt7fNE2bF_csgobVr1gX4JSMwB7eA@mail.gmail.com>

On Sunday, March 1, 2015, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Fast!
> Until it last!!!
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Mar 1, 2015 3:32 PM, "Brian West" <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>
>> https://twitter.com/packtpub/status/572041101076381696
>>
>> Go get it!
>>
>> Start your FreeSWITCHing today!
>>
>> /b
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/fb8a6c30/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: IMG_0127.JPG
Type: image/jpeg
Size: 49383 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/fb8a6c30/attachment-0001.jpe 

From brian at freeswitch.org  Sun Mar  1 18:07:50 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2015 09:07:50 -0600
Subject: [Freeswitch-users] Early Dial / Sip 484
In-Reply-To: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>
References: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>
Message-ID: <CAEJMVkA5HToXh-LQAyauL_d2fAmxg7tz9J3ZM11=KbKCfqE1tg@mail.gmail.com>

Yes, just use the respond app at the bottom of your dial plan with 484 as
the argument

On Sunday, March 1, 2015, Paul Atreides <paul.atreides83 at googlemail.com>
wrote:

> Hi,
>
> does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it
> up in the dialplan?
>
> Thanks
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/dd97df48/attachment.html 

From groysem at gmail.com  Sun Mar  1 18:27:49 2015
From: groysem at gmail.com (Shai Perelman)
Date: Sun, 1 Mar 2015 17:27:49 +0200
Subject: [Freeswitch-users] Announce position in queue
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315CD12C8@ex2.connex.local>
References: <CAJfexNE-kc8rP58u3Z997uf3qs_-HVBY2fbGuOk47dy6aQgrDQ@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315CD12C8@ex2.connex.local>
Message-ID: <CAJfexNHB6p9BREAL_R0DuKJmDKFNqapMnRXi-6ogE2oL0kGVSw@mail.gmail.com>

thanks, whats the irc channel?

On Sun, Mar 1, 2015 at 4:28 PM, Moishe Grunstein <max at nysolutions.com>
wrote:

>  For FusionPBX you would be best served using their IRC channel or their
> Google Code issues.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shai
> Perelman
> *Sent:* Sunday, March 1, 2015 7:21 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Announce position in queue
>
>
>
> I found this code for adding position annoucement ability to fusionpbx.
>
> https://code.google.com/p/fusionpbx/issues/detail?id=658
>
>
>
> can some one point me the steps to integrate it to my fpbx installation?
>
>
>
> it looks like the code files is replacing some original code , is that
> good?
>
> what about version updates, isnt it going to overwrite it?
>
> are there other , better options to acheive this?
>
>
>
> Im new to this so forgive my newbie questions,
>
> I find this list to be the only solution trying to learn fs and fusion, as
> the documentation is very limited.
>
> Thanks
>
> Shai
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
www.groyse.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/9066b086/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/9066b086/attachment.jpg 

From bpriddy at bryantschools.org  Sun Mar  1 18:40:34 2015
From: bpriddy at bryantschools.org (Blakelund Priddy)
Date: Sun, 01 Mar 2015 09:40:34 -0600
Subject: [Freeswitch-users] Announce position in queue
In-Reply-To: <CAJfexNHB6p9BREAL_R0DuKJmDKFNqapMnRXi-6ogE2oL0kGVSw@mail.gmail.com>
References: <CAJfexNE-kc8rP58u3Z997uf3qs_-HVBY2fbGuOk47dy6aQgrDQ@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315CD12C8@ex2.connex.local>
	<CAJfexNHB6p9BREAL_R0DuKJmDKFNqapMnRXi-6ogE2oL0kGVSw@mail.gmail.com>
Message-ID: <082dcbf5af04f76eeb3c9a62200327@ip-10-0-3-72>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/4572737e/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/4572737e/attachment-0001.jpg 

From bote_radio at botecomm.com  Sun Mar  1 19:41:28 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sun, 1 Mar 2015 11:41:28 -0500
Subject: [Freeswitch-users] supporting multiple dtmf protocols
In-Reply-To: <CAEJMVkBmYLed01oqJG-gFeXDchGwZtqXEiteU4ms4wJFb374_g@mail.gmail.com>
References: <CAOccsKDpCDX8_fcee8K5PpHvuHF-+ZGzb5ujnO-Ustay3PWXoQ@mail.gmail.com>	<54F0682C.7090603@freeswitch.org>	<CAOccsKCBju2qM8Nfpq5BzKNshhyVNDe4mUxFMdYjY7HXma9OVw@mail.gmail.com>
	<CAEJMVkBmYLed01oqJG-gFeXDchGwZtqXEiteU4ms4wJFb374_g@mail.gmail.com>
Message-ID: <037101d0543e$91065040$b312f0c0$@botecomm.com>

Experience tells me that it?s likely that some (most?) service providers don?t have the technical competence to know any better. Their plan is:

 

1)      Get the money

2)      Profit!

3)      What?s customer service??

 

Sadly.

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Sunday, 01 March, 2015 09:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] supporting multiple dtmf protocols

 

Why do we keep letting these providers get away with this? Smh

On Sunday, March 1, 2015, ik <idokan at gmail.com> wrote:

 

 

On Fri, Feb 27, 2015 at 2:50 PM, I put the Who? in Mishehu <mishehu at freeswitch.org <javascript:_e(%7B%7D,'cvml','mishehu at freeswitch.org');> > wrote:

I see after all these years it's still the popular thing to think that Bezeq is the most terrible thing in the world.  :-)  Be thankful you don't have to deal with some American incumbent providers...  Bezeq can be a walk in the park...  :-)

 

You mean like answering with 200 OK on INVITE, getting 404 instead of "authentication require", dropping requests because I offer both Alaw and Ulaw and other horror stories around other SIP providers that I have worked with ? :P

 


DTMF handling can be a fun little source of headache in general.

To clarify, are you saying that midway through a call that the DTMF mode changes?  Or are you saying that the DTMF mode for one call may be 2833 and the next call can randomly be inband audio?  If you are saying that the mode changes during the call, what indication do you receive that this has happened?

 

Different types of payloads in RTP, you can see it also in wireshark when capturing with tcpdump 

 

 

 


You can try the "liberal-dtmf" setting and see if that fixes the issue for you, but I believe that only allows SIP NOTIFY and 2833, and I don't believe it handles inband.  (Hopefully somebody else will correct me if I'm mistaken.)

 

I'll try this

Thank you

Ido

 

-- 
Yossi Neiman
 
On 02/27/2015 05:08 AM, ik wrote:

Hello,

I have the misfortune of forced to use a telco named Bezeq - the biggest telco in Israel.

The SIP trunk they provide is very problematic.

The one that I cannot overcome is that sometimes doing the call, it switches DTMF between Inband and rfc2833, and sometimes the whole DTMF sending is either inband or rfc2833 doing the entire call.

 

I cannot make them to be more stable in this matter (I have tried talking with them), and I cannot replace them (I have tried to do so as well).


How can I deal with such mess ?

Thanks,
Ido

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 



-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/71381a3b/attachment.html 

From vladget at gmail.com  Sun Mar  1 23:37:31 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Sun, 1 Mar 2015 22:37:31 +0200
Subject: [Freeswitch-users] inheritance parameters from sofia.conf.xml to
	sip_profiles
Message-ID: <CAMxcU5+2s2TOyP1_qRuBJs2fi1m1j6V=ysDCR6mPFM_zZaA2Dg@mail.gmail.com>

Hi Everyone!

Looks like inheritance of some parameters from sofia.conf.xml
<global_settings> to sip_profiles does not work.

I've faced to problem with next parameters which configured at
<global_settings> in sofia.conf.xml:
- rtp-autofix-timing
- user-agent-string
but has no effect.

Please advice.

-- 
Yours sincerely,
Vladimir Getmanshchuk


From steveayre at gmail.com  Mon Mar  2 00:16:38 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 1 Mar 2015 21:16:38 +0000
Subject: [Freeswitch-users] inheritance parameters from sofia.conf.xml
	to sip_profiles
In-Reply-To: <CAMxcU5+2s2TOyP1_qRuBJs2fi1m1j6V=ysDCR6mPFM_zZaA2Dg@mail.gmail.com>
References: <CAMxcU5+2s2TOyP1_qRuBJs2fi1m1j6V=ysDCR6mPFM_zZaA2Dg@mail.gmail.com>
Message-ID: <CAFiqYu=BGKG2z7PCmEm_i0eyUoEyPyf7kOyEre=Tcn6K5-33WQ@mail.gmail.com>

Profiles do not inherit parameters from global_settings.

The valid parameters for global_settings are:
log-level
tracelevel
debug-presence
debug-sla
max-reg-threads
auto-restart
rewrite-multicasted-fs-path
capture-server

On 1 March 2015 at 20:37, Vladimir Getmanshchuk <vladget at gmail.com> wrote:

> Hi Everyone!
>
> Looks like inheritance of some parameters from sofia.conf.xml
> <global_settings> to sip_profiles does not work.
>
> I've faced to problem with next parameters which configured at
> <global_settings> in sofia.conf.xml:
> - rtp-autofix-timing
> - user-agent-string
> but has no effect.
>
> Please advice.
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/14f62e64/attachment-0001.html 

From brian at freeswitch.org  Mon Mar  2 01:30:25 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2015 16:30:25 -0600
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
Message-ID: <CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>

ZRTP hash in the sdp will cause it to toggle on too!

On Saturday, February 28, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
wrote:

> Bote,
> When I said identical configuration I mean files at FS configuration
> directory.
> G.729 license? No, I use proxy-media mode with no transcoding.
>
> Brian,
> Both FS boxes configured for proxing media:
> # grep inbound-proxy-media /usr/local/freeswitch/conf
> /sip_profiles/internal.xml
> <param name="inbound-proxy-media" value="true"/>
>
> I do not understand why FS version 1.4.15 trying to hide actual
> read/write codecs and change it by "PROXY"?
>
> Thank you.
>
> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>
>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>
>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com
>> <javascript:_e(%7B%7D,'cvml','bote_radio at botecomm.com');>> wrote:
>>
>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>
>>> I would say that these are not "absolutely" identical. As the FreeSWITCH
>>> development team never sleeps it is likely that there are differences in
>>> the
>>> code that you now see. The first thing is to bring both machines up to
>>> the
>>> same release before comparing behaviors.
>>>
>>> Another suggestion is to confirm your G.729 license and configuration, if
>>> you are decoding that codec. Perhaps one machine has the necessary
>>> file(s)
>>> in the correct locations and the other machine does not?
>>>
>>> Hope this helps.
>>>
>>> Bote
>>>
>>>
>>> -----Original Message-----
>>> From: Vladimir Getmanshchuk
>>> Sent: Friday, 27 February, 2015 07:37
>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>> different FS
>>> versions...
>>>
>>> Hello Everyone!
>>>
>>> I have two installations of FS with absolutely identical configurations.
>>> Both has SIP profiles with proxy-media enabled.
>>>
>>> But on
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>
>>> I have values in read_codec/write_codec variables at CDRs:
>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>
>>> but on another one
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>
>>> I have:
>>> "read_codec":"G729","write_codec":"G729",
>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>
>>> So Why?
>>>
>>>
>>> --
>>> Yours sincerely,
>>> Vladimir Getmanshchuk
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/94e7bfd5/attachment.html 

From dujinfang at gmail.com  Mon Mar  2 04:19:23 2015
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 2 Mar 2015 09:19:23 +0800
Subject: [Freeswitch-users] Mod_shout libfacc support
In-Reply-To: <CAEJMVkCFnhUUFK4NMWDfwt9up=KK-BHEc677gzyPiQeFXHNFfA@mail.gmail.com>
References: <CAFZDoUA2OfH55BsC2xBJB3ZWLr=ssdTAvDApTROpqZOKr8=tWw@mail.gmail.com>
	<CAEJMVkCFnhUUFK4NMWDfwt9up=KK-BHEc677gzyPiQeFXHNFfA@mail.gmail.com>
Message-ID: <CAJs+-5o6XfhN+6PEC=YRSQg3XifGGnEN3YmUJbJAphrtADupmw@mail.gmail.com>

 m4a is decoded by vlc so you should can make it work if you can make you
vlc work with m4a.

btw: libfaac is for encoding so you never need it when playback. you may
mean libfaad witch is for decoding. Maybe a bounty can help you get what
you want? note both faac and faad are GPL and also available in commercial
license so that?s you need to figure out first.

-- 
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com
http://www.freeswitch.org.cn

Sent with Sparrow <http://www.sparrowmailapp.com/?sig>

On Wednesday, February 25, 2015 at 5:42 AM, Brian West wrote:

Currently isn't supported.

On Tue, Feb 24, 2015 at 11:55 AM, Eloy Coto Pereiro <eloy.coto at gmail.com>
wrote:

Hi,

I'm trying to play a libfaac[0] files into my Freeswitch, but I can't get
it working. Play mp3 files work ok, but encoding with libfacc doesn't work.

This is my config in mod_shout:

<action application="playback" data="shout://localhost/suite.mp4"/>

In the other hand, I tried mod_vlc to play mp4 files, but It didn't work
too. I compiled from sources, and tried with debian backports too.

<action application="playback" data="vlc://http://localhost/suite.m4a"/>

In both cases http request work, reply 200 ok and with data. Play mp3 with
both modules work ok.

In the other hand, mod_shout and mod_vlc are loaded correctly, and
module_exists return always True.

Any idea? Is libfaac supported?

[0] https://trac.ffmpeg.org/wiki/Encode/AAC#libfaac

Regards

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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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




-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
 _________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/119e8935/attachment.html 

From dujinfang at gmail.com  Mon Mar  2 04:20:37 2015
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 2 Mar 2015 09:20:37 +0800
Subject: [Freeswitch-users] freeswitch + flashphoner web call server
In-Reply-To: <CC9131B0-88E0-466D-8059-A095920A932D@jerris.com>
References: <12f8f87df41ead55280e0b4731aecd4c@spingine.com>
	<54EB5601.4070504@freeswitch.org>
	<fe83849ab88207bcf4cd28deb32446c1@spingine.com>
	<CC9131B0-88E0-466D-8059-A095920A932D@jerris.com>
Message-ID: <CAJs+-5oUf4ungXQB9aRPwtONwQbfY75nsn+ni+ER00+zvxWzMw@mail.gmail.com>

 WebRTC is now supported by many browsers natively or via plugins so it
should work and is the future.

If you still want legacy flash support I think you should read the mod_rtmp
related js and/or flash code to find out how it works. I had made at least
rtmplite and flash-videoio works with FS, and I had made a flowplayer
plugin which even I made it support video.

-- 
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com
http://www.freeswitch.org.cn

Sent with Sparrow <http://www.sparrowmailapp.com/?sig>

On Wednesday, February 25, 2015 at 12:58 AM, Michael Jerris wrote:

maybe webrtc and mod_verto would be a good solution for you?

On Feb 24, 2015, at 1:39 AM, jdayola at spingine.com wrote:

Actually I was able to make calls using the flashphoner client that I
registered to my freeswitch server. My problem is how to make it work
using my own flex client. Im not entirely sure if I even need
flashphoner web call server for my problem. Im using fusionpbx on top of
freeswitch. I already had a working callcenter setup using xlite or
zoiper for testing environment. When I started using a flex client as
replacement for the xlite/zoiper client the agents can no longer receive
the calls. I already enabled mod_rtmp but It looks like im missing
something. I've already read the mod_rtmp enty but Im still a bit
confused. Sorry Im still a noob at this.

On 2015-02-24 12:32 am, I put the Who? in Mishehu wrote:

Just curious to know if FreeSWITCH's mod_rtmp didn't fit your needs.
I've tested some of Flashphoner's products against FreeSWITCH (the
products sent SIP calls to my FreeSWITCH), but for my needs I never
needed nor desired the extra hop in there. Of course, if you meant
RTMFP instead of RTMP, then I could understand easily why you are using
the extra hop. I don't remember exactly though how it was that I set
that up as it was a couple years ago. My guess is that Flashphoner
registered each client as a SIP registration to my FreeSWITCH. If that
is the case, then you need to set up entries in the appropriate
directory include directory under directory/${directoryname}

--
Yossi Neiman

On 02/23/2015 08:09 AM, jdayola at spingine.com wrote:

Hi Guys,

Have anyone tried using flashphoner web call server? I already have a
freeswitch and a flashponer webcallserver running. My problem is how
to
make freeswitch use the flashphoner webcallserver. Which freeswitch
xml
file do I need to edit to make use of the flashphoner webcallserver.
Im
using a flex phone client for my flex desktop application. Do you have
any suggestions on the best approach for sip to rtmp gateway?

Regards,
jdayola



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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/14037a73/attachment-0001.html 

From abaci64 at gmail.com  Mon Mar  2 04:50:01 2015
From: abaci64 at gmail.com (Abaci B)
Date: Sun, 1 Mar 2015 20:50:01 -0500
Subject: [Freeswitch-users] freeswitch say application currency in
 multipal language
In-Reply-To: <CAKzWOxXA81JxyPaapvEdofGTUsRxYjFzK9yXgLVMC6mP2xECRQ@mail.gmail.com>
References: <CAG6TCM-uBQC-Ddh9ynLF1MZnCPT=ctszJgALrkGmqyWKNv4ZBw@mail.gmail.com>
	<CAKzWOxXA81JxyPaapvEdofGTUsRxYjFzK9yXgLVMC6mP2xECRQ@mail.gmail.com>
Message-ID: <CAFmZFiPcjW2fMDJGk-gxUMfrjkhOAwAGiN1bWBaXT0WHQwCutQ@mail.gmail.com>

Just wondering if support for multiple currencies was ever added, if not is
there any plans?

On Thu, Apr 4, 2013 at 2:14 PM, Michael Collins <msc at freeswitch.org> wrote:

> I don't believe that there is currently a way to do this easily right now.
> We just spoke about languages on yesterday's conference call and this is a
> prime example of the kinds of things that we will need to overcome.
>
> Additionally I don't believe that I have any currencies other than
> dollar.wav and dollars.wav for the English sounds. I'll be glad to get them
> ordered. Could the community at large send me some ideas for units of
> currency? Here are a few ideas:
>
> euro, euros
> franc, francs
> Canadian, Australian, US dollar/dollars
> pound, pounds
>
> Send me some more ideas and I will get them added to the to-be-recorded
> list.
>
> -MC
>
> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388 at gmail.com>
> wrote:
>
>> Hi all,
>> I use free switch and i want to play sounds file like if user has credit
>> in USD then doller.wav file play and EUR then another file will be play.
>>
>> Currently it play doller.wav by default in
>> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR
>> so how can this possible.
>>
>> Is that any easy way to do this thing in multi language currency play in
>> say application.
>>
>> i use this syntax in my free-switch dial plan
>> $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED
>> $credit_balance\"/>";
>>
>> Thanks In advance...
>>
>> --
>> Thanks,
>> Bhavik Patel
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150301/b55f576a/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar  2 12:36:46 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Mon, 2 Mar 2015 10:36:46 +0100
Subject: [Freeswitch-users] Changing the voicemail recording menu
Message-ID: <CAP6oyxEf-hVF2xa8o7e5-chm8h=w3BSy1P1CH8Vywy44sD2SAg@mail.gmail.com>

Hi

how do I change the menu when the voicemail answers for record a new
message?

I want it to play the

- greeting message
- record the message
- and then hangup after silence

Is is possible to change the menu when the user access its voicemail as
well?
I found the voicemail_ivr.conf.xml, but I cant find any documentation to it
in the wiki

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/6564fd89/attachment.html 

From idokan at gmail.com  Mon Mar  2 16:07:48 2015
From: idokan at gmail.com (ik)
Date: Mon, 2 Mar 2015 15:07:48 +0200
Subject: [Freeswitch-users] cherry pick calls
Message-ID: <CAOccsKCNn5m7+SOK9PtaGqqnAfYBMegXYbsWjxBhNiukEDi1pA@mail.gmail.com>

Hello,

I'm looking for a way to have some sort of queue that I can cherry pick a
specific caller that I wish to bridge with a specific member.

The only way I can think of, is by using valet parking, is there another
way to do it, that is simpler?

Thanks,
Ido
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/82c8ba95/attachment.html 

From aqsyounas at gmail.com  Mon Mar  2 15:15:46 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Mon, 2 Mar 2015 17:15:46 +0500
Subject: [Freeswitch-users] freeswitch got killed
Message-ID: <CAGVOjVXaz7PM-D=UDbV4yHfC6_xDyLoDGcb6_M0oZeMAvP_fVQ@mail.gmail.com>

Hi, users.

I am playing streams with mod_vlc, but some streams make my switch killed.
I am using the lasted git version.

FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git
4909cdb 2015-02-24 20:58:26Z 64bit)


Logs that i see are these, also log files is attached.

2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute
log(${cur})
2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute
set(episode=0${last_matching_digits})
2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute curl(
http://206.225.05.12/rd_api/api/inboundcampaign/get_extension post
ext=${episode}&did=${dst})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
log(${curl_response_data})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
set(error=No)
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
set(cur=${curl_response_data})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
log(${cur})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
transfer(${cur} XML play)
2015-03-02 05:37:49.594042 [NOTICE] switch_ivr.c:1861 Transfer
sofia/external/19546006100 at 69.27.168.33:5060 to XML[
95.81.147.3/rfimonde/all/rfimonde-64k.mp3 at play]
2015-03-02 05:37:50.094042 [INFO] mod_dialplan_xml.c:635 Processing
19546006100 <19546006100>->95.81.147.3/rfimonde/all/rfimonde-64k.mp3 in
context play
2015-03-02 05:37:50.094042 [INFO] switch_ivr_async.c:212 Digit parser
DPTOOLS: Setting realm to 'moderator'
2015-03-02 05:37:50.114043 [NOTICE] mod_vlc.c:192 VLC Path is http
http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3
[0x25e099b8] access_http access: Raw-audio server found, mp3 demuxer
selected
[0x7fa3fc2f69a8] mpgatofixed32 audio converter error: libmad error: bad
main_data_begin pointer
2015-03-02 05:45:38.494035 [NOTICE] sofia.c:952 Hangup sofia/external/
18034805839 at 69.27.168.71:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2015-03-02 05:45:38.534034 [INFO] mod_json_cdr.c:271 Process
[f7c5c71a-438b-4c56-938e-34cb19766fd6.cdr.json]
2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1641 Session 2591
(sofia/external/18034805839 at 69.27.168.71:5060) Ended
2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/external/18034805839 at 69.27.168.71:5060 [CS_DESTROY]
[0x7fa47bf55668] Killed

What is believe is that freeswitch must not be killed even if stream is bad.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/9211e2d5/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: freeswitch.log
Type: text/x-log
Size: 2806243 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/9211e2d5/attachment-0001.bin 

From max at nysolutions.com  Mon Mar  2 16:43:11 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Mon, 2 Mar 2015 13:43:11 +0000
Subject: [Freeswitch-users] freeswitch got killed
In-Reply-To: <CAGVOjVXaz7PM-D=UDbV4yHfC6_xDyLoDGcb6_M0oZeMAvP_fVQ@mail.gmail.com>
References: <CAGVOjVXaz7PM-D=UDbV4yHfC6_xDyLoDGcb6_M0oZeMAvP_fVQ@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315CD4A93@ex2.connex.local>

https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aqs Younas
Sent: Monday, March 2, 2015 7:16 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] freeswitch got killed

Hi, users.

I am playing streams with mod_vlc, but some streams make my switch killed.
I am using the lasted git version.
FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git 4909cdb 2015-02-24 20:58:26Z 64bit)

Logs that i see are these, also log files is attached.

2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute log(${cur})
2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute set(episode=0${last_matching_digits})
2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute curl(http://206.225.05.12/rd_api/api/inboundcampaign/get_extension post ext=${episode}&did=${dst})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute log(${curl_response_data})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute set(error=No)
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute set(cur=${curl_response_data})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute log(${cur})
2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute transfer(${cur} XML play)
2015-03-02 05:37:49.594042 [NOTICE] switch_ivr.c:1861 Transfer sofia/external/19546006100 at 69.27.168.33:5060<http://19546006100 at 69.27.168.33:5060> to XML[95.81.147.3/rfimonde/all/rfimonde-64k.mp3 at play<http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3 at play>]
2015-03-02 05:37:50.094042 [INFO] mod_dialplan_xml.c:635 Processing 19546006100 <19546006100>->95.81.147.3/rfimonde/all/rfimonde-64k.mp3<http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3> in context play
2015-03-02 05:37:50.094042 [INFO] switch_ivr_async.c:212 Digit parser DPTOOLS: Setting realm to 'moderator'
2015-03-02 05:37:50.114043 [NOTICE] mod_vlc.c:192 VLC Path is http http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3
[0x25e099b8] access_http access: Raw-audio server found, mp3 demuxer selected
[0x7fa3fc2f69a8] mpgatofixed32 audio converter error: libmad error: bad main_data_begin pointer
2015-03-02 05:45:38.494035 [NOTICE] sofia.c:952 Hangup sofia/external/18034805839 at 69.27.168.71:5060<http://18034805839 at 69.27.168.71:5060> [CS_EXECUTE] [NORMAL_CLEARING]
2015-03-02 05:45:38.534034 [INFO] mod_json_cdr.c:271 Process [f7c5c71a-438b-4c56-938e-34cb19766fd6.cdr.json]
2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1641 Session 2591 (sofia/external/18034805839 at 69.27.168.71:5060<http://18034805839 at 69.27.168.71:5060>) Ended
2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/18034805839 at 69.27.168.71:5060<http://18034805839 at 69.27.168.71:5060> [CS_DESTROY]
[0x7fa47bf55668] Killed
What is believe is that freeswitch must not be killed even if stream is bad.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/fae6d3f9/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/fae6d3f9/attachment.jpg 

From ssinyagin at gmail.com  Mon Mar  2 17:16:09 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 2 Mar 2015 15:16:09 +0100
Subject: [Freeswitch-users] cherry pick calls
In-Reply-To: <CAOccsKCNn5m7+SOK9PtaGqqnAfYBMegXYbsWjxBhNiukEDi1pA@mail.gmail.com>
References: <CAOccsKCNn5m7+SOK9PtaGqqnAfYBMegXYbsWjxBhNiukEDi1pA@mail.gmail.com>
Message-ID: <CACOYK=g5XzFN7s228rypSu2G1XkMpH9yRf=HxcEycRRVVfNT3w@mail.gmail.com>

you can let the inbound calls play MOH, and use ESL to uuid_break and
uuid_bridge the ones you need.

On Mon, Mar 2, 2015 at 2:07 PM, ik <idokan at gmail.com> wrote:
> Hello,
>
> I'm looking for a way to have some sort of queue that I can cherry pick a
> specific caller that I wish to bridge with a specific member.
>
> The only way I can think of, is by using valet parking, is there another way
> to do it, that is simpler?
>
> Thanks,
> Ido
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tfred31 at yahoo.com  Mon Mar  2 18:14:48 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Mon, 2 Mar 2015 07:14:48 -0800
Subject: [Freeswitch-users] Getting FS to place Outbound calls
Message-ID: <1425309288.93614.YahooMailBasic@web160202.mail.bf1.yahoo.com>

I have made some progress in learning FreeSWITCH, but now the next hurdle.

I have my in-house softphones connecting to each other and I have an in-house SIP hard-phone connecting to the various in-house extensions.
And I can place a call to my 'outside' number which utilizes the SIP line to connect to one of my in-house extensions.
That is progress!

Now I want to get one of my in-house extensions to be able to connect to an 'outside' number via my single inbound/outbound SIP line.

I have followed the advice found on the web, but it is not working.

1. My Firewall is open to port 5060

2. Within the directory:  conf\sip_profiles\external   I have created a new XML file    velocity_outbound.xml    and within it I configured a gateway 
    Since my SIP line provider indicates that:   No SIP authentication is required.    I set the parameters as follows
         <include>
               <gateway name="velocity-outbound">
                       <param name="username" value=""/>  <!-- No Username needed -->
                       <param name="password" value="" />  <!-- No Password needed -->
                       <param name="proxy" value="<my SIP line IP address here>"/>
                       <param name="register" value="false"/>  <!-- No Registration needed -->
   
                      <!-- which transport to use for register -->
                      <param name="register-transport" value="udp"/>
                </gateway>
          </include>

3. In the directory:  conf\dialplan\default\     I created a new file:   outbound_via_velocity.xml   in which I defined what to do:
         <include>
           <extension name="domestic.velocity.short">
              <condition field="${toll_allow}" expression="domestic"/>
              <condition field="destination_number" expression="^(\d{10})$">
                 <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
                 <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
                 <action application="bridge" data="sofia/gateway/velocity-outbound/$1"/>
              </condition>
          </extension>
 
          <extension name="domestic.velocity">
              <condition field="${toll_allow}" expression="domestic"/>
              <condition field="destination_number" expression="^(\d{11})$">
                  <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
                  <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
                 <action application="bridge" data="sofia/gateway/velocity-outbound$1"/>
              </condition>
           </extension>
        </include>

I attempt to place an outside call and I only get a BUSY.

I look at the freeswitch.log and I see that the new gateway file is accessed:
       Action bridge(sofia/gateway/velocity-outbound/<number dialed here>)
Later in the log I see
       (sofia/external-ipv6/<number dialled here>) State Change CS_INIT -> CS_ROUTING
Followed by:
       sofia/external-ipv6/<number dialled here> entering state [calling][0]
      [DEBUG] sofia.c:6403 Channel sofia/external-ipv6/<number dialled here> entering state [terminated][503]
      [NOTICE] sofia.c:7286 Hangup sofia/external-ipv6/<number dialled here> [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
      [DEBUG] switch_channel.c:3222 Send signal sofia/external-ipv6/<number dialled here> [KILL]
This is repeated a few times and finally
      (sofia/internal/1001@<my FS IP address here>:5060) Locked, Waiting on external entities
      (sofia/internal/1001@<my FS IP address here>:5060) Ended
      (sofia/internal/1001@<my FS IP address here>:5060) Running State Change CS_DESTROY
      (sofia/external-ipv6/<number dialled here>) Ended
      (sofia/external-ipv6/<number dialled here>) State DESTROY going to sleep

(Obviously due to the MANY lines of info in the log, I am only showing a few of them here)

I am not yet familiar enough with 'interpreting' what the log is trying to tell me with the exception of:  Something Did Not Work.

Have I not created and/or configured something wrong in order to get my call out working?

Any other suggestions/advice?

Thanks,
TF



From bote_radio at botecomm.com  Mon Mar  2 19:29:29 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 2 Mar 2015 11:29:29 -0500
Subject: [Freeswitch-users] Getting FS to place Outbound calls
In-Reply-To: <1425309288.93614.YahooMailBasic@web160202.mail.bf1.yahoo.com>
References: <1425309288.93614.YahooMailBasic@web160202.mail.bf1.yahoo.com>
Message-ID: <045001d05506$0f681af0$2e3850d0$@botecomm.com>

Ensure that the <gateway name=" velocity-outbound"> exactly, as that is what
FS looks for when you specify sofia/gateway/velocity-outbound in the bridge
line.


A bigger problem might be:
      [DEBUG] switch_channel.c:3222 Send signal sofia/external-ipv6/<number
dialed here> [KILL] This is repeated a few times and finally

This states that the call is being sent out the external-ipv6 profile, which
I'm guessing you're not using. 

Each sip_profile describes a unique i.p. address and port number combination
and typically small installations such as yours and mine in our homes need
no more than 2 profiles. What I do is simply rename all unneeded profiles
with a ".txt" extension so that FS won't see an .xml file under the
sip_profiles/ directory tree and not pick them up at all. 

Also, note at the bottom of the dialplan/default.xml is an 'include' command
that picks up files in the child directory, which is how it found your
outbound_via_velocity.xml dialplan. Since that is tacked on to the end of
the default.xml dialplan it's possible that an earlier extension condition
is matching your dialed digits and FS never even gets down to your included
dialplan; that might be how your test call got sent out the ipv6 profile. I
find it worthwhile to test for 

^9(1\d{10})$

which is convenient since the channel variable $1 will be stuffed with 1
plus the 10 digit destination number that my provider wants to see. Since I
have to dial 9 to make an outside call to my provider this leaves me with
wide flexibility for my internal dialplan. 

Of course, you can play with the dialplan to match your needs any which way
you see fit, that's the beauty of FS. Feel free to make a backup copy of the
original dialplan and rip out all the unneeded example extensions that come
with FS, it will make your debugging much easier not seeing all those tests
fly by in the logs. In fact, you could trim it down to only 1 or 2 extension
solely for the purpose of testing these outbound calls to your provider;
when you perfect that, add back what minimal lines you need to get the rest
done.

It looks like you've made substantial progress, you're almost there.

Bote




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of T Fred
Farmington
Sent: Monday, 02 March, 2015 10:15
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Getting FS to place Outbound calls


3. In the directory:  conf\dialplan\default\     I created a new file:
outbound_via_velocity.xml   in which I defined what to do:
         <include>
           <extension name="domestic.velocity.short">
              <condition field="${toll_allow}" expression="domestic"/>
              <condition field="destination_number" expression="^(\d{10})$">
                 <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
                 <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
                 <action application="bridge"
data="sofia/gateway/velocity-outbound/$1"/>
              </condition>
          </extension>
 
          <extension name="domestic.velocity">
              <condition field="${toll_allow}" expression="domestic"/>
              <condition field="destination_number" expression="^(\d{11})$">
                  <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
                  <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
                 <action application="bridge"
data="sofia/gateway/velocity-outbound$1"/>
              </condition>
           </extension>
        </include>

I attempt to place an outside call and I only get a BUSY.

I look at the freeswitch.log and I see that the new gateway file is
accessed:
       Action bridge(sofia/gateway/velocity-outbound/<number dialed here>)
Later in the log I see
       (sofia/external-ipv6/<number dialled here>) State Change CS_INIT ->
CS_ROUTING Followed by:
       sofia/external-ipv6/<number dialled here> entering state [calling][0]
      [DEBUG] sofia.c:6403 Channel sofia/external-ipv6/<number dialled here>
entering state [terminated][503]
      [NOTICE] sofia.c:7286 Hangup sofia/external-ipv6/<number dialled here>
[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
      [DEBUG] switch_channel.c:3222 Send signal sofia/external-ipv6/<number
dialled here> [KILL] This is repeated a few times and finally
      (sofia/internal/1001@<my FS IP address here>:5060) Locked, Waiting on
external entities
      (sofia/internal/1001@<my FS IP address here>:5060) Ended
      (sofia/internal/1001@<my FS IP address here>:5060) Running State
Change CS_DESTROY
      (sofia/external-ipv6/<number dialled here>) Ended
      (sofia/external-ipv6/<number dialled here>) State DESTROY going to
sleep

(Obviously due to the MANY lines of info in the log, I am only showing a few
of them here)

I am not yet familiar enough with 'interpreting' what the log is trying to
tell me with the exception of:  Something Did Not Work.

Have I not created and/or configured something wrong in order to get my call
out working?

Any other suggestions/advice?

Thanks,
TF


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Mon Mar  2 20:06:07 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Mon, 02 Mar 2015 11:06:07 -0600
Subject: [Freeswitch-users] Changing the voicemail recording menu
In-Reply-To: <CAP6oyxEf-hVF2xa8o7e5-chm8h=w3BSy1P1CH8Vywy44sD2SAg@mail.gmail.com>
References: <CAP6oyxEf-hVF2xa8o7e5-chm8h=w3BSy1P1CH8Vywy44sD2SAg@mail.gmail.com>
Message-ID: <54F4987F.6010205@freeswitch.org>

You can modify the recordings that are played by looking at the files in 
conf/lang/<LANGUAGE>/vm . Those contain the sound macros, and that's 
what you'd want to modify.

-- 
Yossi Neiman
  

On 03/02/2015 03:36 AM, Paul Atreides wrote:
> Hi
>
> how do I change the menu when the voicemail answers for record a new 
> message?
>
> I want it to play the
>
> - greeting message
> - record the message
> - and then hangup after silence
>
> Is is possible to change the menu when the user access its voicemail 
> as well?
> I found the voicemail_ivr.conf.xml, but I cant find any documentation 
> to it in the wiki
>
> Thanks
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/149151a1/attachment.html 

From aqsyounas at gmail.com  Mon Mar  2 20:21:23 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Mon, 2 Mar 2015 22:21:23 +0500
Subject: [Freeswitch-users] How can i create dynamic dialplan in freeswitch.
Message-ID: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>

Hi, user.

After working for more than 3 months while writing my dialplan in static
xml file,but now wants to know how can i effectively create dynamic
dialplan in freeswitch.

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/72e0b12e/attachment.html 

From vipkilla at gmail.com  Mon Mar  2 20:24:55 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 2 Mar 2015 12:24:55 -0500
Subject: [Freeswitch-users] How can i create dynamic dialplan in
	freeswitch.
In-Reply-To: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>
References: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>
Message-ID: <CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>

Hi,
Look at mod_xml_curl to do a 'dynamic' dialplan.
Thanks.

On Mon, Mar 2, 2015 at 12:21 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, user.
>
> After working for more than 3 months while writing my dialplan in static
> xml file,but now wants to know how can i effectively create dynamic
> dialplan in freeswitch.
>
> Thanks.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/a46ec847/attachment.html 

From Rob.Moore at aeriandi.com  Mon Mar  2 21:01:56 2015
From: Rob.Moore at aeriandi.com (Rob Moore)
Date: Mon, 2 Mar 2015 18:01:56 +0000
Subject: [Freeswitch-users] ICE SDP Params
Message-ID: <DB3PR05MB028E9043CA144DFE146793283100@DB3PR05MB028.eurprd05.prod.outlook.com>

Hi All,

After a recent upgrade of Freeswitch I've been seeing some unusual behaviour in codec choice being made by our clients hardware during late negotiation calls.
No other changes have  taken place that could have caused this unusual behaviour so we assume that it must be something the newer version of  freeswitch is doing in this situation.

Comparing traces before and after the upgrade, the SDP in the final 200 OK's from our freeswitch now contains more parameters relating to ICE and other source specific attributes. I expect these additional params are what is upsetting our clients hardware.

I've since attempted to disable all forms of NAT management in Freeswitch in an attempt to get rid of these extra SDP attributes but none of them seem to have had any effect:

Setting the following in the sip Profile to disable stun / NAT.
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-auto-disable
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-enabled

ensuring the following are set to the local ip of the server
<X-PRE-PROCESS cmd="set" data="external_sip_ip=localIP "/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=localIP "/>


Does anyone have any suggestions on how I can remove these additional SDP params?


I've included an example good and bad  200 ok in case I've missed anything else that's obvious.

Many thanks

Rob

Bad 200 ok:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.4.78;branch=z9hG4bK0ba4.7a1f3a47e3eb3a2a4e8e13fd7dda8ab8.0
Via: SIP/2.0/UDP 10.9.138.26:5060;rport=5060;branch=z9hG4bK9ivhm3hvpf5c7vrsu2gh7ou4o2
Record-Route: <sip: 172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes>
Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes>
From: <sip:8448224432 at 10.150.35.124>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33986537
To: <sip:00353876721075 at 10.9.138.9>;tag=r3ZZc86S529mD
Call-ID: b87ef500-4f012e3b-1f738-7c23960a at 10.150.35.124
CSeq: 101 INVITE
Contact: <sip:mod_sofia at 172.18.4.251:5060>
User-Agent: Aeriandi Tel Server
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 841
x-nt-location: -1

v=0
o=FreeSWITCH 1425007009 1425007010 IN IP4 172.18.4.251
s=FreeSWITCH
c=IN IP4 172.18.4.251
t=0 0
a=msid-semantic: WMS hGYoorbKxEnXtlapBPoffcf3QQg7ijgq
m=audio 19630 RTP/SAVPF 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256 18:6A:21:5F:BF:02:8B:52:29:96:85:6B:05:99:B2:9D:C7:B3:26:DB:F9:32:5A:90:51:62:E8:21:3E:C2:23:C3
a=rtcp-mux
a=rtcp:19630 IN IP4 172.18.4.251
a=ssrc:3641993291 cname:Q4E4mGFKbSxu4yJv
a=ssrc:3641993291 msid:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq a0
a=ssrc:3641993291 mslabel:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq
a=ssrc:3641993291 label:hGYoorbKxEnXtlapBPoffcf3QQg7ijgqa0
a=ice-ufrag:eYAj0GcHniFvsLL8
a=ice-pwd:Gg3SMysNgP8bdIwhwXqnttUH
a=candidate:1204810811 1 udp 659136 172.18.4.251 19630 typ host generation 0
a=ptime:20


good 200 ok:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.4.78;branch=z9hG4bKef6a.6cc24c86d42011b141c967c04c23e077.0
Via: SIP/2.0/UDP 10.9.138.26:5060;rport=5060;branch=z9hG4bKa2v9mvku4854npk24c4tn51vv2
Record-Route: <sip:172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes>
Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes>
From: <sip:8448224432 at 10.151.35.123>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33990014
To: <sip:07940845322 at 10.9.138.9>;tag=Fy3a1Zv7Kt6Xm
Call-ID: f59cb480-4f014116-1f782-7b23970a at 10.151.35.123
CSeq: 101 INVITE
Contact: <sip:mod_sofia at 172.18.4.254:5060>
User-Agent: Aeriandi Tel Server
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: 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: 224
x-nt-location: -1

v=0
o=FreeSWITCH 1425014601 1425014602 IN IP4 172.18.4.254
s=FreeSWITCH
c=IN IP4 172.18.4.254
t=0 0
m=audio 16858 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/06bb5d53/attachment-0001.html 

From brian at freeswitch.org  Mon Mar  2 21:15:01 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 2 Mar 2015 13:15:01 -0500
Subject: [Freeswitch-users] ICE SDP Params
In-Reply-To: <DB3PR05MB028E9043CA144DFE146793283100@DB3PR05MB028.eurprd05.prod.outlook.com>
References: <DB3PR05MB028E9043CA144DFE146793283100@DB3PR05MB028.eurprd05.prod.outlook.com>
Message-ID: <CAEJMVkAvFowQXiDE5KJrWsc+HEBzXmuVoPS5RjzPNf6Gw5FWyQ@mail.gmail.com>

This is all related to WebRTC, how are you creating the invite?  Logs would
be helpful./b

On Monday, March 2, 2015, Rob Moore <Rob.Moore at aeriandi.com> wrote:

>  Hi All,
>
>
>
> After a recent upgrade of Freeswitch I?ve been seeing some unusual
> behaviour in codec choice being made by our clients hardware during late
> negotiation calls.
>
> No other changes have  taken place that could have caused this unusual
> behaviour so we assume that it must be something the newer version of
>  freeswitch is doing in this situation.
>
>
>
> Comparing traces before and after the upgrade, the SDP in the final 200
> OK?s from our freeswitch now contains more parameters relating to ICE and
> other source specific attributes. I expect these additional params are what
> is upsetting our clients hardware.
>
>
>
> I?ve since attempted to disable all forms of NAT management in Freeswitch
> in an attempt to get rid of these extra SDP attributes but none of them
> seem to have had any effect:
>
>
>
> Setting the following in the sip Profile to disable stun / NAT.
>
> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-auto-disable
>
> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#stun-enabled
>
>
>
> ensuring the following are set to the local ip of the server
>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=*localIP* "/>
>
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=*localIP *"/>
>
>
>
>
>
> Does anyone have any suggestions on how I can remove these additional SDP
> params?
>
>
>
>
>
> I?ve included an example good and bad  200 ok in case I?ve missed anything
> else that?s obvious.
>
>
>
> Many thanks
>
>
>
> Rob
>
>
>
> *Bad 200 ok: *
>
>
>
> *SIP/2.0 200 OK*
>
> *Via: SIP/2.0/UDP
> 172.18.4.78;branch=z9hG4bK0ba4.7a1f3a47e3eb3a2a4e8e13fd7dda8ab8.0*
>
> *Via: SIP/2.0/UDP
> 10.9.138.26:5060;rport=5060;branch=z9hG4bK9ivhm3hvpf5c7vrsu2gh7ou4o2*
>
> *Record-Route: <sip: 172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes>*
>
> *Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes>*
>
> *From: <sip:8448224432 at 10.150.35.124
> <javascript:_e(%7B%7D,'cvml','sip:8448224432 at 10.150.35.124');>>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33986537*
>
> *To: <sip:00353876721075 at 10.9.138.9
> <javascript:_e(%7B%7D,'cvml','sip:00353876721075 at 10.9.138.9');>>;tag=r3ZZc86S529mD*
>
> *Call-ID: b87ef500-4f012e3b-1f738-7c23960a at 10.150.35.124
> <javascript:_e(%7B%7D,'cvml','b87ef500-4f012e3b-1f738-7c23960a at 10.150.35.124');>*
>
> *CSeq: 101 INVITE*
>
> *Contact: <sip:mod_sofia at 172.18.4.251:5060
> <http://sip:mod_sofia at 172.18.4.251:5060>>*
>
> *User-Agent: Aeriandi Tel Server*
>
> *Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE*
>
> *Supported: path, replaces*
>
> *Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer*
>
> *Content-Type: application/sdp*
>
> *Content-Disposition: session*
>
> *Content-Length: 841*
>
> *x-nt-location: -1*
>
>
>
> *v=0*
>
> *o=FreeSWITCH 1425007009 1425007010 IN IP4 172.18.4.251*
>
> *s=FreeSWITCH*
>
> *c=IN IP4 172.18.4.251*
>
> *t=0 0*
>
> *a=msid-semantic: WMS hGYoorbKxEnXtlapBPoffcf3QQg7ijgq*
>
> *m=audio 19630 RTP/SAVPF 18 0 8 101*
>
> *a=rtpmap:18 G729/8000*
>
> *a=rtpmap:0 PCMU/8000*
>
> *a=rtpmap:8 PCMA/8000*
>
> *a=rtpmap:101 telephone-event/8000*
>
> *a=fingerprint:sha-256
> 18:6A:21:5F:BF:02:8B:52:29:96:85:6B:05:99:B2:9D:C7:B3:26:DB:F9:32:5A:90:51:62:E8:21:3E:C2:23:C3*
>
> *a=rtcp-mux*
>
> *a=rtcp:19630 IN IP4 172.18.4.251*
>
> *a=ssrc:3641993291 cname:Q4E4mGFKbSxu4yJv*
>
> *a=ssrc:3641993291 msid:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq a0*
>
> *a=ssrc:3641993291 mslabel:hGYoorbKxEnXtlapBPoffcf3QQg7ijgq*
>
> *a=ssrc:3641993291 label:hGYoorbKxEnXtlapBPoffcf3QQg7ijgqa0*
>
> *a=ice-ufrag:eYAj0GcHniFvsLL8*
>
> *a=ice-pwd:Gg3SMysNgP8bdIwhwXqnttUH*
>
> *a=candidate:1204810811 1 udp 659136 172.18.4.251 19630 typ host
> generation 0*
>
> *a=ptime:20*
>
>
>
>
>
> *good 200 ok:*
>
>
>
> *SIP/2.0 200 OK*
>
> *Via: SIP/2.0/UDP
> 172.18.4.78;branch=z9hG4bKef6a.6cc24c86d42011b141c967c04c23e077.0*
>
> *Via: SIP/2.0/UDP
> 10.9.138.26:5060;rport=5060;branch=z9hG4bKa2v9mvku4854npk24c4tn51vv2*
>
> *Record-Route: <sip:172.18.4.78;r2=on;lr=on;rtpi=1;nat=yes>*
>
> *Record-Route: <sip:10.16.0.3;r2=on;lr=on;rtpi=1;nat=yes>*
>
> *From: <sip:8448224432 at 10.151.35.123
> <javascript:_e(%7B%7D,'cvml','sip:8448224432 at 10.151.35.123');>>;tag=130e56d9-dcc0-4483-9c8e-edf93ab1fb9a-33990014*
>
> *To: <sip:07940845322 at 10.9.138.9
> <javascript:_e(%7B%7D,'cvml','sip:07940845322 at 10.9.138.9');>>;tag=Fy3a1Zv7Kt6Xm*
>
> *Call-ID: f59cb480-4f014116-1f782-7b23970a at 10.151.35.123
> <javascript:_e(%7B%7D,'cvml','f59cb480-4f014116-1f782-7b23970a at 10.151.35.123');>*
>
> *CSeq: 101 INVITE*
>
> *Contact: <sip:mod_sofia at 172.18.4.254:5060
> <http://sip:mod_sofia at 172.18.4.254:5060>>*
>
> *User-Agent: Aeriandi Tel Server*
>
> *Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE*
>
> *Supported: 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: 224*
>
> *x-nt-location: -1*
>
>
>
> *v=0*
>
> *o=FreeSWITCH 1425014601 1425014602 IN IP4 172.18.4.254*
>
> *s=FreeSWITCH*
>
> *c=IN IP4 172.18.4.254*
>
> *t=0 0*
>
> *m=audio 16858 RTP/AVP 18 0 8 101*
>
> *a=fmtp:18 annexb=no*
>
> *a=rtpmap:101 telephone-event/8000*
>
> *a=fmtp:101 0-16*
>
> *a=ptime:20*
>
>
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/991e65e0/attachment.html 

From msc at freeswitch.org  Mon Mar  2 21:42:46 2015
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Mar 2015 10:42:46 -0800
Subject: [Freeswitch-users] How can i create dynamic dialplan in
	freeswitch.
In-Reply-To: <CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>
References: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>
	<CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>
Message-ID: <CAKzWOxXa9M+Q+1wA-JVjrmd5_xXPUd3YggOMp+KOdF8BExFLTg@mail.gmail.com>

See also chapter 9 of the FreeSWITCH 1.2 book, appropriately entitled,
"Moving Beyond the Static XML Configuration."

-MC

On Mon, Mar 2, 2015 at 9:24 AM, Vik Killa <vipkilla at gmail.com> wrote:

> Hi,
> Look at mod_xml_curl to do a 'dynamic' dialplan.
> Thanks.
>
> On Mon, Mar 2, 2015 at 12:21 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi, user.
>>
>> After working for more than 3 months while writing my dialplan in static
>> xml file,but now wants to know how can i effectively create dynamic
>> dialplan in freeswitch.
>>
>> Thanks.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/24d2d7da/attachment-0001.html 

From bordmi at rarus.ru  Tue Mar  3 00:09:48 2015
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=)
Date: Tue, 3 Mar 2015 00:09:48 +0300
Subject: [Freeswitch-users] Fwd: How it works? Lua & MySQL throug ODBC
In-Reply-To: <CAL2h29i683d+Cbh5t2UrfSeKwKgWmrRXK0+xo43Pe37T8GppMQ@mail.gmail.com>
References: <CAL2h29i683d+Cbh5t2UrfSeKwKgWmrRXK0+xo43Pe37T8GppMQ@mail.gmail.com>
Message-ID: <CAL2h29jfwPVHH73DeKeV4dqFgiXF4hgGiY4NoZUsqA1FJ4vbcQ@mail.gmail.com>

Hi, All!

I have experienced periodicaly problems with FreeSWITCH running LUA scripts.

This scripts are event hooks. In some unknown reasons some times FreeSWITCH
crashes without any records in log. I`ve some qustions:

1. How can I enable more detailed debug?
2. How to store freeswitch.core in some explained previously place?
3. May be thread blocking while doing transaction to MySQL through ODBC the
source of my problems? If yes, how to solve it?

-- 
with best regards,
Dmitriy Borisov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/fce4b144/attachment.html 

From dylan at onsip.com  Tue Mar  3 00:28:19 2015
From: dylan at onsip.com (Dylan Mikus)
Date: Mon, 2 Mar 2015 16:28:19 -0500
Subject: [Freeswitch-users] Determining if Freeswitch channel is using a
	video codec
Message-ID: <CAOQuP_25HNSoP0BPAhCVSAKzyADCOwoGnNoty34170dDxRT9CA@mail.gmail.com>

I?m trying to determine if a given channel over Freeswitch is using a video
codec. In my config/vars.xml file, I?ve set the codecs line to:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264"/>

Logs

My SDP negotiation appears to be correct. The INVITE:

INVITE sip:queuecard at cyberdyne.onsip.com SIP/2.0
Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
Max-Forwards: 70
To: <sip:queuecard at cyberdyne.onsip.com>
From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
Call-ID: n3o4g4i724sq7qkekp07
CSeq: 8622 INVITE
Proxy-Authorization: Digest algorithm=MD5,
username="cyberdyne_bender", realm="jnctn.net",
nonce="54f4ce2e000013e4888519dec3ca2ee1ef9023f82d4d8922",
uri="sip:queuecard at cyberdyne.onsip.com",
response="f842951ecc11c3510d1e1b7abcdeb51f", qop=auth,
cnonce="d153n6udlh74", nc=00000001
Contact: <sip:1fjqm8ps at o8iatftbl1mn.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
Content-Type: application/sdp
Supported: 100rel,outbound
User-Agent: SIP.js/0.6.3 InstaPhone
Content-Length: 1649

v=0
o=Mozilla-SIPUA-35.0.1 10886 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:a256418b
a=ice-pwd:62e2ae7154b57f00ed0b1a2003ccf7af
a=fingerprint:sha-256
EA:C4:92:D4:94:62:18:41:39:2E:42:B4:4E:B7:32:9E:66:FE:7C:01:57:AC:2C:4C:E4:66:4F:3B:B6:91:FA:DC
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2130379007 192.168.1.38 58531 typ host
a=candidate:0 2 UDP 2130379006 192.168.1.38 64677 typ host
a=candidate:1 1 UDP 1694236671 38.104.167.182 49209 typ srflx raddr
192.168.1.38 rport 58531
a=candidate:1 2 UDP 1694236670 38.104.167.182 51209 typ srflx raddr
192.168.1.38 rport 64677
m=video 9 UDP/TLS/RTP/SAVPF 120 126 97
c=IN IP4 0.0.0.0
a=rtpmap:120 VP8/90000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01f
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:126 nack
a=rtcp-fb:126 nack pli
a=rtcp-fb:126 ccm fir
a=rtcp-fb:97 nack
a=rtcp-fb:97 nack pli
a=rtcp-fb:97 ccm fir
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2130379007 192.168.1.38 59562 typ host
a=candidate:0 2 UDP 2130379006 192.168.1.38 61464 typ host
a=candidate:1 1 UDP 1694236671 38.104.167.182 59357 typ srflx raddr
192.168.1.38 rport 59562
a=candidate:1 2 UDP 1694236670 38.104.167.182 21168 typ srflx raddr
192.168.1.38 rport 61464

The 200 OK response:

SIP/2.0 200 OK
Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
Record-Route: <sip:38.109.82.167;lr;ftag=fneppn1lhh;did=24.0b0068f6>
Record-Route: <sip:38.109.82.192;lr;ftag=fneppn1lhh;did=24.4c4e8343>
Record-Route: <sip:38.109.82.193:5060;transport=udp;lr;ovid=cac1e79d>
Record-Route: <sip:2b14c410c1 at 38.109.82.193:80;transport=ws;lr;ovid=cac1e79d>
From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
To: <sip:queuecard at cyberdyne.onsip.com>;tag=30yQvF62DQyyg
Call-ID: n3o4g4i724sq7qkekp07
CSeq: 8622 INVITE
Contact: <sip:dtlsgw0 at app2-1.55-broad-1.app.jnctn.net>
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1738

v=0
o=FreeSWITCH 1425307798 1425307799 IN IP4 38.109.82.228
s=FreeSWITCH
c=IN IP4 38.109.82.228
t=0 0
a=msid-semantic: WMS 61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
m=audio 21882 UDP/TLS/RTP/SAVPF 109 101
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1;usedtx=1;maxaveragebitrate=30000
a=rtpmap:101 telephone-event/8000
a=recvonly
a=silenceSupp:off - - - -
a=ptime:20
a=fingerprint:sha-256
D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
a=rtcp:21883 IN IP4 38.109.82.228
a=ssrc:2365215248 cname:CkzZ9cdxFymMTiha
a=ssrc:2365215248 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a0
a=ssrc:2365215248 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
a=ssrc:2365215248 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwa0
a=ice-ufrag:zqC4ZTWsD5d4Hyqa
a=ice-pwd:IQ0IT35osh0bq7bPoDKenwwR
a=candidate:9358589392 1 udp 659136 38.109.82.228 21882 typ host generation 0
a=candidate:9358589392 2 udp 659134 38.109.82.228 21883 typ host generation 0
m=video 23680 UDP/TLS/RTP/SAVPF 126
b=AS:256
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
a=recvonly
a=fingerprint:sha-256
D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
a=rtcp:23681 IN IP4 38.109.82.228
a=rtcp-fb:* fir pli
a=ssrc:1652571152 cname:CkzZ9cdxFymMTiha
a=ssrc:1652571152 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw v0
a=ssrc:1652571152 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
a=ssrc:1652571152 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwv0
a=ice-ufrag:kirEYQCPyInSpi7Y
a=ice-pwd:fbIhKWJB3fFuGVyQ4QlSwNxU
a=candidate:9055446981 1 udp 659136 38.109.82.228 23680 typ host generation 0
a=candidate:9055446981 2 udp 659134 38.109.82.228 23681 typ host generation 0

We offer:

a=rtpmap:120 VP8/90000
a=rtpmap:126 H264/90000

and we accept:

a=rtpmap:126 H264/90000

Note that this is on Firefox 35.0.1.
Response

While this call is up, I run show channels in fs_cli and get the following:

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
b41772a4-c11b-11e4-a78f-7585ed98a76c,inbound,2015-03-02
20:35:45,1425328545,sofia/sip0/bender at cyberdyne.onsip.com,CS_SOFT_EXECUTE,Bender
Rodriguez,bender,38.109.82.167,queuecard,uuid_bridge,bc7c6d64-c11b-11e4-a798-7585ed98a76c,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound
Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Outbound
Call,terabithia
bc7c6d64-c11b-11e4-a798-7585ed98a76c,outbound,2015-03-02
20:35:59,1425328559,sofia/app/c3po at cyberdyne.onsip.com,CS_EXCHANGE_MEDIA,Bender
Rodriguez,bender,38.109.82.167,1000,uuid_bridge,b41772a4-c11b-11e4-a78f-7585ed98a76c,XML,generic-app,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound
Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Bender
Rodriguez,bender
bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,inbound,2015-03-02
20:35:59,1425328559,sofia/sip0/bender at cyberdyne.onsip.com,CS_EXECUTE,Bender
Rodriguez,bender,38.109.82.167,gl8k15o7,bridge,{force_transfer_context=refer}sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid;transport=ws;aor=c3po%40cyberdyne.onsip.com,XML,default,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound
Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Outbound
Call,gl8k15o7
bc82805a-c11b-11e4-a7ae-7585ed98a76c,outbound,2015-03-02
20:35:59,1425328559,sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid,CS_EXCHANGE_MEDIA,Bender
Rodriguez,bender,38.109.82.167,gl8k15o7,,,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound
Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Bender
Rodriguez,bender

The read codecs and write codecs are OPUS, except for a websocket transport
that lists an XML codec, I think. Is something up with my setup, or do we
only show the audio codec being used when we run the show channels command?
Any other idea for how to determine whether a Freeswitch channel is using
video? I?m trying to stay away from sending custom headers and I want to be
able to figure this out within Freeswitch. In other words, I don?t want a
receiving application to try to figure out whether it is in video or not. I
just want to query my Freeswitch service to find out.

Thanks, guys! I appreciate any help.
?

-- 
Dylan Mikus
Software Engineer
OnSIP
www.onsip.com
p. 212.933.9190 x7060
SIP/Email: dylan at onsip.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/cd74c081/attachment-0001.html 

From dylan at onsip.com  Tue Mar  3 00:35:10 2015
From: dylan at onsip.com (Dylan Mikus)
Date: Mon, 2 Mar 2015 16:35:10 -0500
Subject: [Freeswitch-users] Determining if Freeswitch channel is using a
	video codec
In-Reply-To: <CAOQuP_25HNSoP0BPAhCVSAKzyADCOwoGnNoty34170dDxRT9CA@mail.gmail.com>
References: <CAOQuP_25HNSoP0BPAhCVSAKzyADCOwoGnNoty34170dDxRT9CA@mail.gmail.com>
Message-ID: <CAOQuP_3XocnTPOcMyZ-4ueu5NJy8d8=NQ3Snmgi+Y8ATW8e43Q@mail.gmail.com>

Actually, I do not necessarily need the codec. I only need to determine if
a call is using video or not. The codec is useful additional information,
but not necessary.

On Mon, Mar 2, 2015 at 4:28 PM, Dylan Mikus <dylan at onsip.com> wrote:

> I?m trying to determine if a given channel over Freeswitch is using a
> video codec. In my config/vars.xml file, I?ve set the codecs line to:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264"/>
>
> Logs
>
> My SDP negotiation appears to be correct. The INVITE:
>
> INVITE sip:queuecard at cyberdyne.onsip.com SIP/2.0
> Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
> Max-Forwards: 70
> To: <sip:queuecard at cyberdyne.onsip.com>
> From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
> Call-ID: n3o4g4i724sq7qkekp07
> CSeq: 8622 INVITE
> Proxy-Authorization: Digest algorithm=MD5, username="cyberdyne_bender", realm="jnctn.net", nonce="54f4ce2e000013e4888519dec3ca2ee1ef9023f82d4d8922", uri="sip:queuecard at cyberdyne.onsip.com", response="f842951ecc11c3510d1e1b7abcdeb51f", qop=auth, cnonce="d153n6udlh74", nc=00000001
> Contact: <sip:1fjqm8ps at o8iatftbl1mn.invalid;transport=ws;ob>
> Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
> Content-Type: application/sdp
> Supported: 100rel,outbound
> User-Agent: SIP.js/0.6.3 InstaPhone
> Content-Length: 1649
>
> v=0
> o=Mozilla-SIPUA-35.0.1 10886 0 IN IP4 0.0.0.0
> s=SIP Call
> t=0 0
> a=ice-ufrag:a256418b
> a=ice-pwd:62e2ae7154b57f00ed0b1a2003ccf7af
> a=fingerprint:sha-256 EA:C4:92:D4:94:62:18:41:39:2E:42:B4:4E:B7:32:9E:66:FE:7C:01:57:AC:2C:4C:E4:66:4F:3B:B6:91:FA:DC
> m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
> c=IN IP4 0.0.0.0
> a=rtpmap:109 opus/48000/2
> a=ptime:20
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=setup:actpass
> a=rtcp-mux
> a=candidate:0 1 UDP 2130379007 192.168.1.38 58531 typ host
> a=candidate:0 2 UDP 2130379006 192.168.1.38 64677 typ host
> a=candidate:1 1 UDP 1694236671 38.104.167.182 49209 typ srflx raddr 192.168.1.38 rport 58531
> a=candidate:1 2 UDP 1694236670 38.104.167.182 51209 typ srflx raddr 192.168.1.38 rport 64677
> m=video 9 UDP/TLS/RTP/SAVPF 120 126 97
> c=IN IP4 0.0.0.0
> a=rtpmap:120 VP8/90000
> a=rtpmap:126 H264/90000
> a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
> a=rtpmap:97 H264/90000
> a=fmtp:97 profile-level-id=42e01f
> a=sendrecv
> a=rtcp-fb:120 nack
> a=rtcp-fb:120 nack pli
> a=rtcp-fb:120 ccm fir
> a=rtcp-fb:126 nack
> a=rtcp-fb:126 nack pli
> a=rtcp-fb:126 ccm fir
> a=rtcp-fb:97 nack
> a=rtcp-fb:97 nack pli
> a=rtcp-fb:97 ccm fir
> a=setup:actpass
> a=rtcp-mux
> a=candidate:0 1 UDP 2130379007 192.168.1.38 59562 typ host
> a=candidate:0 2 UDP 2130379006 192.168.1.38 61464 typ host
> a=candidate:1 1 UDP 1694236671 38.104.167.182 59357 typ srflx raddr 192.168.1.38 rport 59562
> a=candidate:1 2 UDP 1694236670 38.104.167.182 21168 typ srflx raddr 192.168.1.38 rport 61464
>
> The 200 OK response:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
> Record-Route: <sip:38.109.82.167;lr;ftag=fneppn1lhh;did=24.0b0068f6>
> Record-Route: <sip:38.109.82.192;lr;ftag=fneppn1lhh;did=24.4c4e8343>
> Record-Route: <sip:38.109.82.193:5060;transport=udp;lr;ovid=cac1e79d>
> Record-Route: <sip:2b14c410c1 at 38.109.82.193:80;transport=ws;lr;ovid=cac1e79d>
> From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
> To: <sip:queuecard at cyberdyne.onsip.com>;tag=30yQvF62DQyyg
> Call-ID: n3o4g4i724sq7qkekp07
> CSeq: 8622 INVITE
> Contact: <sip:dtlsgw0 at app2-1.55-broad-1.app.jnctn.net>
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
> Supported: precondition, 100rel, timer, path, replaces
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 1738
>
> v=0
> o=FreeSWITCH 1425307798 1425307799 IN IP4 38.109.82.228
> s=FreeSWITCH
> c=IN IP4 38.109.82.228
> t=0 0
> a=msid-semantic: WMS 61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
> m=audio 21882 UDP/TLS/RTP/SAVPF 109 101
> a=rtpmap:109 opus/48000/2
> a=fmtp:109 useinbandfec=1;usedtx=1;maxaveragebitrate=30000
> a=rtpmap:101 telephone-event/8000
> a=recvonly
> a=silenceSupp:off - - - -
> a=ptime:20
> a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
> a=rtcp:21883 IN IP4 38.109.82.228
> a=ssrc:2365215248 cname:CkzZ9cdxFymMTiha
> a=ssrc:2365215248 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a0
> a=ssrc:2365215248 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
> a=ssrc:2365215248 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwa0
> a=ice-ufrag:zqC4ZTWsD5d4Hyqa
> a=ice-pwd:IQ0IT35osh0bq7bPoDKenwwR
> a=candidate:9358589392 1 udp 659136 38.109.82.228 21882 typ host generation 0
> a=candidate:9358589392 2 udp 659134 38.109.82.228 21883 typ host generation 0
> m=video 23680 UDP/TLS/RTP/SAVPF 126
> b=AS:256
> a=rtpmap:126 H264/90000
> a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
> a=recvonly
> a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
> a=rtcp:23681 IN IP4 38.109.82.228
> a=rtcp-fb:* fir pli
> a=ssrc:1652571152 cname:CkzZ9cdxFymMTiha
> a=ssrc:1652571152 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw v0
> a=ssrc:1652571152 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
> a=ssrc:1652571152 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwv0
> a=ice-ufrag:kirEYQCPyInSpi7Y
> a=ice-pwd:fbIhKWJB3fFuGVyQ4QlSwNxU
> a=candidate:9055446981 1 udp 659136 38.109.82.228 23680 typ host generation 0
> a=candidate:9055446981 2 udp 659134 38.109.82.228 23681 typ host generation 0
>
> We offer:
>
> a=rtpmap:120 VP8/90000
> a=rtpmap:126 H264/90000
>
> and we accept:
>
> a=rtpmap:126 H264/90000
>
> Note that this is on Firefox 35.0.1.
> Response
>
> While this call is up, I run show channels in fs_cli and get the
> following:
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> b41772a4-c11b-11e4-a78f-7585ed98a76c,inbound,2015-03-02 20:35:45,1425328545,sofia/sip0/bender at cyberdyne.onsip.com,CS_SOFT_EXECUTE,Bender Rodriguez,bender,38.109.82.167,queuecard,uuid_bridge,bc7c6d64-c11b-11e4-a798-7585ed98a76c,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Outbound Call,terabithia
> bc7c6d64-c11b-11e4-a798-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/app/c3po at cyberdyne.onsip.com,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,1000,uuid_bridge,b41772a4-c11b-11e4-a78f-7585ed98a76c,XML,generic-app,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Bender Rodriguez,bender
> bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,inbound,2015-03-02 20:35:59,1425328559,sofia/sip0/bender at cyberdyne.onsip.com,CS_EXECUTE,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,bridge,{force_transfer_context=refer}sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid;transport=ws;aor=c3po%40cyberdyne.onsip.com,XML,default,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Outbound Call,gl8k15o7
> bc82805a-c11b-11e4-a7ae-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,,,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Bender Rodriguez,bender
>
> The read codecs and write codecs are OPUS, except for a websocket
> transport that lists an XML codec, I think. Is something up with my setup,
> or do we only show the audio codec being used when we run the show
> channels command? Any other idea for how to determine whether a
> Freeswitch channel is using video? I?m trying to stay away from sending
> custom headers and I want to be able to figure this out within Freeswitch.
> In other words, I don?t want a receiving application to try to figure out
> whether it is in video or not. I just want to query my Freeswitch service
> to find out.
>
> Thanks, guys! I appreciate any help.
> ?
>
> --
> Dylan Mikus
> Software Engineer
> OnSIP
> www.onsip.com
> p. 212.933.9190 x7060
> SIP/Email: dylan at onsip.com
>



-- 
Dylan Mikus
Software Engineer
OnSIP
www.onsip.com
p. 212.933.9190 x7060
SIP/Email: dylan at onsip.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/300ac906/attachment-0001.html 

From mike at jerris.com  Tue Mar  3 00:59:58 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 2 Mar 2015 16:59:58 -0500
Subject: [Freeswitch-users] freeswitch say application currency in
	multipal language
In-Reply-To: <CAFmZFiPcjW2fMDJGk-gxUMfrjkhOAwAGiN1bWBaXT0WHQwCutQ@mail.gmail.com>
References: <CAG6TCM-uBQC-Ddh9ynLF1MZnCPT=ctszJgALrkGmqyWKNv4ZBw@mail.gmail.com>
	<CAKzWOxXA81JxyPaapvEdofGTUsRxYjFzK9yXgLVMC6mP2xECRQ@mail.gmail.com>
	<CAFmZFiPcjW2fMDJGk-gxUMfrjkhOAwAGiN1bWBaXT0WHQwCutQ@mail.gmail.com>
Message-ID: <AD82ABCA-4BF0-4620-A022-98C737D3DB00@jerris.com>

We never added it but would be open to others doing the work to add it and getting us a pull request.  We will need to get some additional sound prompts as well and come up with some sane way to handle different currency names.  If you have a proposal of a sane flexible way to do this, make a proposal and a pul request and we can look at getting the needed prompts.


> On Mar 1, 2015, at 8:50 PM, Abaci B <abaci64 at gmail.com> wrote:
> 
> Just wondering if support for multiple currencies was ever added, if not is there any plans?
> 
> On Thu, Apr 4, 2013 at 2:14 PM, Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
> I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome. 
> 
> Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas:
> 
> euro, euros
> franc, francs
> Canadian, Australian, US dollar/dollars
> pound, pounds
> 
> Send me some more ideas and I will get them added to the to-be-recorded list.
> 
> -MC
> 
> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388 at gmail.com <mailto:bhavikpatel14388 at gmail.com>> wrote:
> Hi all, 
> I use free switch and i want to play sounds file like if user has credit in USD then doller.wav file play and EUR then another file will be play. 
> 
> Currently it play doller.wav by default in /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR so how can this possible. 
> 
> Is that any easy way to do this thing in multi language currency play in say application. 
> 
> i use this syntax in my free-switch dial plan 
> $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>"; 
> 
> Thanks In advance...
> 
> -- 
> Thanks,
> Bhavik Patel
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/11f79112/attachment.html 

From krice at freeswitch.org  Tue Mar  3 01:08:50 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 02 Mar 2015 22:08:50 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
	February 21st-27th
Message-ID: <54f4df721b0a5_65fc1732086645@resque-worker-ip-10-168-230-218.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1DLKdYZ
FreeSWITCH Week in Review (Master Branch) February 21st-27th
Hello, again. This passed week in the FreeSWITCH master branch we had 11 commits. The features for this week are: updating mod_verto to proxy additional variables and the ability to force URL refresh in mod_http_cache.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7312 Update mod_verto to proxy additional variables

FS-7323 Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.

Improvements in cross platform build supports:

FS-6520 Fix for libv8 build issue using MSVC 2013

The following bugs were squashed:

FS-7307 Fixed buffering issue when recording calls in native format

FS-7126 Fixed coredump when calling the translate application

FS-7314 Fix for configure error caused by a broken openssl 1.0.2 includes

FS-7313 Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail

FS-7322 Fix for issues building on centos 5 and others distributions with older autotools

FS-6758 Fixed issue with hold dropping calls on Skinny Cisco 7961G

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/c25fd087/attachment.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 06:35:45 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 09:05:45 +0530
Subject: [Freeswitch-users] Freeswitch and sipML
Message-ID: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>

Hi,

I work on freeswitch to develop call centre solutions for customers. I am using browser based sipML client & facing the problem of excessive amount of sessions that do not close decently. This probably leads to hanging of my application. Has anyone faced such problem & can someone suggest an effective solution / safeguard. I would prefer using browser because it is effective for pop up applications in call centre environment.

Regards,

Naveen Khanna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/61184a44/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 06:38:03 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 09:08:03 +0530
Subject: [Freeswitch-users] Embedding Freeswitch
Message-ID: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>

Hi,

Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch.


Regards,

Naveen Khanna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/a8a7c6be/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 06:39:33 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 09:09:33 +0530
Subject: [Freeswitch-users] Freeswitch Video Conferencing required
Message-ID: <284F8FAC-A57A-43B3-9802-565B89E223E8@gmail.com>

HI,

Suggest video conference solution with Freeswitch that can be supported. Need minimum 16 party video conference sessions with streaming server & multicast capability.

Regards,

Naveen Khanna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/0eefb617/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 06:40:26 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 09:10:26 +0530
Subject: [Freeswitch-users] Help required on FXO FXS module
Message-ID: <EFAFF307-0612-4868-A22F-CA8636D8CC61@gmail.com>

Hi,

Can someone help source or design FXO FXS modules or boards at around $5 per port with command line interface.

Regards,

Naveen Khanna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/994e10ed/attachment-0001.html 

From tom at tomlynn.com  Tue Mar  3 07:16:19 2015
From: tom at tomlynn.com (Tom Lynn)
Date: Mon, 2 Mar 2015 20:16:19 -0800
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
Message-ID: <CAERdjiacacxxLgTJy_jZT2AOiNxUscZqwtUJSZ2g12Afh=q2tw@mail.gmail.com>

Naveen, I think you need a consultant that you can pay for solutions.

On Mon, Mar 2, 2015 at 7:38 PM, Naveen Khanna <naveen.khanna.bm at gmail.com>
wrote:

> Hi,
>
> Can someone suggest a low cost standard single board computer, around $
> 50, to run 25 concurrent sessions & 100 registrations of SIP clients with
> limited applications of Freeswitch.
>
>
> Regards,
>
> Naveen Khanna
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/268ca3ec/attachment.html 

From mike at jerris.com  Tue Mar  3 07:59:03 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 2 Mar 2015 23:59:03 -0500
Subject: [Freeswitch-users] Freeswitch and sipML
In-Reply-To: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>
References: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>
Message-ID: <C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>

I would reccomend using sip.js if it must be sip, or if sip is not a requirement take a look at our own custom client verto.


> On Mar 2, 2015, at 10:35 PM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> Hi,
> 
> I work on freeswitch to develop call centre solutions for customers. I am using browser based sipML client & facing the problem of excessive amount of sessions that do not close decently. This probably leads to hanging of my application. Has anyone faced such problem & can someone suggest an effective solution / safeguard. I would prefer using browser because it is effective for pop up applications in call centre environment.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150302/30eed8cd/attachment.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 07:28:56 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 09:58:56 +0530
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <CAERdjiacacxxLgTJy_jZT2AOiNxUscZqwtUJSZ2g12Afh=q2tw@mail.gmail.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
	<CAERdjiacacxxLgTJy_jZT2AOiNxUscZqwtUJSZ2g12Afh=q2tw@mail.gmail.com>
Message-ID: <B05F8379-BCFC-4DF1-B2E0-4FD12A43438E@gmail.com>

Thanks Tom,

Can you provide me some pointers on some information with contacts.  

Regards,

Naveen Khanna


> On 03-Mar-2015, at 9:46 am, Tom Lynn <tom at tomlynn.com> wrote:
> 
> Naveen, I think you need a consultant that you can pay for solutions.
> 
> On Mon, Mar 2, 2015 at 7:38 PM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
> Hi,
> 
> Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch.
> 
> 
> Regards,
> 
> Naveen Khanna
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/f1aab545/attachment.html 

From mike at jerris.com  Tue Mar  3 08:00:05 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Mar 2015 00:00:05 -0500
Subject: [Freeswitch-users] Freeswitch Video Conferencing required
In-Reply-To: <284F8FAC-A57A-43B3-9802-565B89E223E8@gmail.com>
References: <284F8FAC-A57A-43B3-9802-565B89E223E8@gmail.com>
Message-ID: <866CE18D-4524-4B49-8C14-5105A4832670@jerris.com>

Freeswitch 1.6 will include this functionality.  We are working hard to get this completed.  If you are interested in contributing to this work you can contact consulting at freeswitch.org.

> On Mar 2, 2015, at 10:39 PM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> HI,
> 
> Suggest video conference solution with Freeswitch that can be supported. Need minimum 16 party video conference sessions with streaming server & multicast capability

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/55a11d26/attachment.html 

From mike at jerris.com  Tue Mar  3 08:07:50 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Mar 2015 00:07:50 -0500
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <B05F8379-BCFC-4DF1-B2E0-4FD12A43438E@gmail.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
	<CAERdjiacacxxLgTJy_jZT2AOiNxUscZqwtUJSZ2g12Afh=q2tw@mail.gmail.com>
	<B05F8379-BCFC-4DF1-B2E0-4FD12A43438E@gmail.com>
Message-ID: <5E58D12C-4303-48C7-AA54-E4644226E6A5@jerris.com>

Consulting services are provided via FreeSWITCH Solutions.  You can contact us at consulting at freeswitch.org.

Thanks
Mike

> On Mar 2, 2015, at 11:28 PM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> Thanks Tom,
> 
> Can you provide me some pointers on some information with contacts.  
> 
> Regards,
> 
> Naveen Khanna
> 
> 
>> On 03-Mar-2015, at 9:46 am, Tom Lynn <tom at tomlynn.com <mailto:tom at tomlynn.com>> wrote:
>> 
>> Naveen, I think you need a consultant that you can pay for solutions.
>> 
>> On Mon, Mar 2, 2015 at 7:38 PM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>> Hi,
>> 
>> Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/d3502de3/attachment-0001.html 

From max at nysolutions.com  Tue Mar  3 08:22:14 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 3 Mar 2015 05:22:14 +0000
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <5E58D12C-4303-48C7-AA54-E4644226E6A5@jerris.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
	<CAERdjiacacxxLgTJy_jZT2AOiNxUscZqwtUJSZ2g12Afh=q2tw@mail.gmail.com>
	<B05F8379-BCFC-4DF1-B2E0-4FD12A43438E@gmail.com>
	<5E58D12C-4303-48C7-AA54-E4644226E6A5@jerris.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315CD9423@ex2.connex.local>

Have a look at the odroid or RasPI

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, March 3, 2015 12:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Embedding Freeswitch

Consulting services are provided via FreeSWITCH Solutions.  You can contact us at consulting at freeswitch.org<mailto:consulting at freeswitch.org>.

Thanks
Mike

On Mar 2, 2015, at 11:28 PM, Naveen Khanna <naveen.khanna.bm at gmail.com<mailto:naveen.khanna.bm at gmail.com>> wrote:

Thanks Tom,

Can you provide me some pointers on some information with contacts.

Regards,

Naveen Khanna


On 03-Mar-2015, at 9:46 am, Tom Lynn <tom at tomlynn.com<mailto:tom at tomlynn.com>> wrote:

Naveen, I think you need a consultant that you can pay for solutions.

On Mon, Mar 2, 2015 at 7:38 PM, Naveen Khanna <naveen.khanna.bm at gmail.com<mailto:naveen.khanna.bm at gmail.com>> wrote:
Hi,

Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/360e517a/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/360e517a/attachment.jpg 

From jungleboogie0 at gmail.com  Tue Mar  3 08:29:24 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Mon, 2 Mar 2015 21:29:24 -0800
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
Message-ID: <CAKE2PDurg6VdKucdU2zYH=g0PJrVEiWonB3WoT+d4eweNCOQYw@mail.gmail.com>

Hi Naveen,
On 2 March 2015 at 19:38, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> Hi,
>
> Can someone suggest a low cost standard single board computer, around $ 50,
> to run 25 concurrent sessions & 100 registrations of SIP clients with
> limited applications of Freeswitch.
>

Specific to freeswitch, I don't know but this has caught my eye:
http://www.minnowboard.org/meet-minnowboard-max/

64bit atom with up to a gig of RAM.

The new raspberry pi is quite nice but its 32bit.

>
> Regards,
>
> Naveen Khanna




-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From naveen.khanna.bm at gmail.com  Tue Mar  3 08:38:36 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 11:08:36 +0530
Subject: [Freeswitch-users] Freeswitch and sipML
In-Reply-To: <C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>
References: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>
	<C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>
Message-ID: <2BEC9C40-3157-425E-B053-58D0372F17CA@gmail.com>

Thanks for the inputs.

Regards,

Naveen Khanna


> On 03-Mar-2015, at 10:29 am, Michael Jerris <mike at jerris.com> wrote:
> 
> I would reccomend using sip.js if it must be sip, or if sip is not a requirement take a look at our own custom client verto.
> 
> 
>> On Mar 2, 2015, at 10:35 PM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>> 
>> Hi,
>> 
>> I work on freeswitch to develop call centre solutions for customers. I am using browser based sipML client & facing the problem of excessive amount of sessions that do not close decently. This probably leads to hanging of my application. Has anyone faced such problem & can someone suggest an effective solution / safeguard. I would prefer using browser because it is effective for pop up applications in call centre environment.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/a7016a44/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Tue Mar  3 08:42:58 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 3 Mar 2015 11:12:58 +0530
Subject: [Freeswitch-users] Freeswitch Video Conferencing required
In-Reply-To: <866CE18D-4524-4B49-8C14-5105A4832670@jerris.com>
References: <284F8FAC-A57A-43B3-9802-565B89E223E8@gmail.com>
	<866CE18D-4524-4B49-8C14-5105A4832670@jerris.com>
Message-ID: <7E75055F-5D4A-4B79-A3A1-A7C108950928@gmail.com>

Thanks for the inputs. Yes sure, I will be happy to contribute. 

Regards,

Naveen Khanna.


> On 03-Mar-2015, at 10:30 am, Michael Jerris <mike at jerris.com> wrote:
> 
> Freeswitch 1.6 will include this functionality.  We are working hard to get this completed.  If you are interested in contributing to this work you can contact consulting at freeswitch.org <mailto:consulting at freeswitch.org>.
> 
>> On Mar 2, 2015, at 10:39 PM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>> 
>> HI,
>> 
>> Suggest video conference solution with Freeswitch that can be supported. Need minimum 16 party video conference sessions with streaming server & multicast capability
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/254ef0b0/attachment-0001.html 

From telishisheer at gmail.com  Tue Mar  3 09:03:29 2015
From: telishisheer at gmail.com (Shisheer Teli)
Date: Tue, 3 Mar 2015 11:33:29 +0530
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
Message-ID: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>

Hi Team,

My freeswitch server is on IPv6, and now i am able register extension with
IPv6 in freeswitch.

but  i am unable to call from IPv6 extensions..

can help ..?

Regards,
shisheer T
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/d34738a4/attachment.html 

From manish.talwar at nexxuspg.com  Tue Mar  3 09:21:40 2015
From: manish.talwar at nexxuspg.com (Manish Talwar)
Date: Tue, 3 Mar 2015 06:21:40 +0000
Subject: [Freeswitch-users]
 =?utf-8?q?Implementing_telecom=E2=80=8B_module?=
 =?utf-8?q?_with_FreeSwitch?=
In-Reply-To: <1424911092386.69803@nexxuspg.com>
References: <1424911092386.69803@nexxuspg.com>
Message-ID: <1425412349467.9202@nexxuspg.com>

Hello,


Please suggest me about my below mentioned email.


Thanks,


Regards,?

Manish Talwar

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Manish Talwar <manish.talwar at nexxuspg.com>
Sent: 25 February 2015 16:37
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Implementing telecom? module with FreeSwitch


Hi,


We have successfully implemented FreeSwitch with our IVR application using httapi module and its running fine now. After implementing IVR application, we are looking for implementing telecom? module with FreeSwitch now.


We have a plan to activating a set of 20 number, for this we have a "sangoma wanpipe driver" on the server and some kernel modules loaded that will communicate with a Sangoma A104 card installed there. Incoming Qatari +974 phone calls will arrive and will be translated on our server to SIP traffic for our IVR system to process.  One or more of the numbers will be reserved as office numbers.


I have looked into telecom service of FreeSwitch and found freeTDM module for implementing telecom with FreeSwitch.


Can we achieve this telecom implementation by freeTDM module of FreeSwitch? If yes, then please help me for implementing ?telecom? module ?with FreeSwitch with freeTDM and let me know all details about it. Also, please let me know any other useful information regarding this module.


Thanks,


Regards,

Manish Talwar
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/497ce46c/attachment.html 

From jayesh1017 at gmail.com  Tue Mar  3 10:12:58 2015
From: jayesh1017 at gmail.com (Jayesh Nambiar)
Date: Tue, 03 Mar 2015 07:12:58 +0000
Subject: [Freeswitch-users] mod_callcenter not updating
References: <CAP9PqkQkq_-LCvy+eJ5aoXzhhzR0dLo_wyHNvsh8CtfD5LhD3Q@mail.gmail.com>
Message-ID: <CANAYozsYSDUN+ke6d57twyqvwsarBBaksQXRPpgKpoaQKmc-pQ@mail.gmail.com>

I believe you need to do a reloadxml before the reload mod_callcenter. Also
there are callcenter_config related API commands for agents and queues
which you can use to add/remove agents without editing the conf file !!

On Sat, Feb 28, 2015 at 10:58 AM Ali Jibran <ali.jibran44 at gmail.com> wrote:

> I am using FreeSWITCH (Version 1.5.15b git 556cb5c 2015-02-05 00:55:29Z
> 64bit). For some reason "reload mod_callcenter" doesn't update when I edit
> callcenter.conf.xml.
> It only updates agents being added but the ones that have been removed do
> not go away. Is it some bug?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/b2c1d4ad/attachment.html 

From lists at ione.ch  Tue Mar  3 10:16:17 2015
From: lists at ione.ch (Roman_)
Date: Tue, 3 Mar 2015 00:16:17 -0700 (MST)
Subject: [Freeswitch-users] Javascript,
	session_in_hangup_hook and api_hangup_hook
Message-ID: <1425366977120-7596149.post@n2.nabble.com>

Hi,

I have been trying to run a javascript on hangup in my dialplan, where I
need access to the session in the script, but I cannot get it to work.
Searching the mailing list also has not revealed anything pertinent (except
that it *should* work for at least lua and javascript). My (partial)
dialplan looks as follows:

<action application="set" data="session_in_hangup_hook=true"/>
<action application="set" data="api_hangup_hook=jsrun
/etc/freeswitch/scripts/script.js"/>
<action application="bridge" data="..."/>

The javascript gets called reliably, however, the session object is not
available. If I try to access it, I get an error:

2015-03-02 22:19:22.260371 [ERR] script.js:7 Exception: TypeError: Object
#<error> has no method 'getVariable' (near: "var total_billed =
session.getVariable("nibble_total_billed");")

So it seems the session object isn't actually a session, but some error?
Just printing the session object to the console will output "false".

Is there something I am missing here? Has anybody managed to use the session
object in a javascript script in the api hangup hook?

I am using FreeSWITCH Version 1.4.15-1~64bit (-1 64bit) (the debian
packages).

Any help would be greatly appreciated.

Thanks and best regards,

Roman




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Javascript-session-in-hangup-hook-and-api-hangup-hook-tp7596149.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ssinyagin at gmail.com  Tue Mar  3 12:04:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 10:04:51 +0100
Subject: [Freeswitch-users] Embedding Freeswitch
In-Reply-To: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
References: <4C0E960D-A8E3-4E9C-AEEC-E9D69A8E138C@gmail.com>
Message-ID: <CACOYK=i00hoS5VS0JE4zKBRKxOKr8Y6Z+dyqtY8sTY1jhOiqEg@mail.gmail.com>

PC Engines APU platform is a good choice:
http://pcengines.ch/apu.htm

Here's my Debian installer for it:
https://github.com/ssinyagin/pcengines-apu-debian-cd

Here are some of my blog posts about the device:
https://txlab.wordpress.com/tag/pcengines/

I'm not affiliated with PC Engines, but their office is around the
corner, so I get the board with next-day delivery :-)



On Tue, Mar 3, 2015 at 4:38 AM, Naveen Khanna
<naveen.khanna.bm at gmail.com> wrote:
> Hi,
>
> Can someone suggest a low cost standard single board computer, around $ 50,
> to run 25 concurrent sessions & 100 registrations of SIP clients with
> limited applications of Freeswitch.
>
>
> Regards,
>
> Naveen Khanna
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Tue Mar  3 12:08:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 10:08:51 +0100
Subject: [Freeswitch-users] Freeswitch Video Conferencing required
In-Reply-To: <866CE18D-4524-4B49-8C14-5105A4832670@jerris.com>
References: <284F8FAC-A57A-43B3-9802-565B89E223E8@gmail.com>
	<866CE18D-4524-4B49-8C14-5105A4832670@jerris.com>
Message-ID: <CACOYK=gnmtkVPV=Rj0Z2o86ZQXhvhGp3W2xb1FN6UHdPRbBEWA@mail.gmail.com>

That will be great to test. The current video conferencing that is
available with mod_conference is not very useful, especially with
multi-vendor or multi-platform clients.



On Tue, Mar 3, 2015 at 6:00 AM, Michael Jerris <mike at jerris.com> wrote:
> Freeswitch 1.6 will include this functionality.  We are working hard to get
> this completed.  If you are interested in contributing to this work you can
> contact consulting at freeswitch.org.
>
> On Mar 2, 2015, at 10:39 PM, Naveen Khanna <naveen.khanna.bm at gmail.com>
> wrote:
>
> HI,
>
> Suggest video conference solution with Freeswitch that can be supported.
> Need minimum 16 party video conference sessions with streaming server &
> multicast capability
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Tue Mar  3 12:09:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 10:09:51 +0100
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
Message-ID: <CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>

but you didn't provide any information, so it's difficult to help.

On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com> wrote:
> Hi Team,
>
> My freeswitch server is on IPv6, and now i am able register extension with
> IPv6 in freeswitch.
>
> but  i am unable to call from IPv6 extensions..
>
> can help ..?
>
> Regards,
> shisheer T
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From telishisheer at gmail.com  Tue Mar  3 12:23:48 2015
From: telishisheer at gmail.com (Shisheer Teli)
Date: Tue, 3 Mar 2015 14:53:48 +0530
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
	<CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
Message-ID: <CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>

Hi Team,

My freeswitch server is on IPv6, and now i am able register extension with
IPv6 in freeswitch.

but  i am unable to call from IPv6 extensions..

Error:
015-03-03 14:49:21.568064 [NOTICE] switch_channel.c:1055 New Channel
sofia/internal-ipv6/1102@[clientipv6address]:5060
[60707716-c186-11e4-88f0-adeca182559b]
2015-03-03 14:49:21.588040 [WARNING] sofia_reg.c:2827 Can't find user
[1102@[serveripv6address]]
from clientipv6address
You must define a domain called '[serveripv6address]' in your directory and
add a user with the id="1102" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2015-03-03 14:49:21.588040 [NOTICE] sofia.c:2063 Hangup
sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_NEW] [CALL_REJECTED]
2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1641 Session 1
(sofia/internal-ipv6/1102@[clientipv6address]:5060) Ended
2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_DESTROY]

Regards,
Shisheer


On Tue, Mar 3, 2015 at 2:39 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> but you didn't provide any information, so it's difficult to help.
>
> On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com>
> wrote:
> > Hi Team,
> >
> > My freeswitch server is on IPv6, and now i am able register extension
> with
> > IPv6 in freeswitch.
> >
> > but  i am unable to call from IPv6 extensions..
> >
> > can help ..?
> >
> > Regards,
> > shisheer T
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Shisheer Teli
Phone: +91-022 2278 2519 / 2121
shisheer at tifr.res.in
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/1d4adf96/attachment.html 

From ssinyagin at gmail.com  Tue Mar  3 12:34:25 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 10:34:25 +0100
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
	<CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
	<CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
Message-ID: <CACOYK=jNN-Lfqz9HWQAuYY29GeZ8+e5VYrfryTJ5PCWZum4FFw@mail.gmail.com>

what is "show registrations" telling in regards to users and realms?

Also it probably makes sense to use a real domain name with SRV and
NAPTR DNS entries, instead of plain IPv6 address.



On Tue, Mar 3, 2015 at 10:23 AM, Shisheer Teli <telishisheer at gmail.com> wrote:
> Hi Team,
>
> My freeswitch server is on IPv6, and now i am able register extension with
> IPv6 in freeswitch.
>
> but  i am unable to call from IPv6 extensions..
>
> Error:
> 015-03-03 14:49:21.568064 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal-ipv6/1102@[clientipv6address]:5060
> [60707716-c186-11e4-88f0-adeca182559b]
> 2015-03-03 14:49:21.588040 [WARNING] sofia_reg.c:2827 Can't find user
> [1102@[serveripv6address]] from clientipv6address
> You must define a domain called '[serveripv6address]' in your directory and
> add a user with the id="1102" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2015-03-03 14:49:21.588040 [NOTICE] sofia.c:2063 Hangup
> sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_NEW] [CALL_REJECTED]
> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1641 Session 1
> (sofia/internal-ipv6/1102@[clientipv6address]:5060) Ended
> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1645 Close Channel
> sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_DESTROY]
>
> Regards,
> Shisheer
>
>
> On Tue, Mar 3, 2015 at 2:39 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>>
>> but you didn't provide any information, so it's difficult to help.
>>
>> On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com>
>> wrote:
>> > Hi Team,
>> >
>> > My freeswitch server is on IPv6, and now i am able register extension
>> > with
>> > IPv6 in freeswitch.
>> >
>> > but  i am unable to call from IPv6 extensions..
>> >
>> > can help ..?
>> >
>> > Regards,
>> > shisheer T
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> Regards,
> Shisheer Teli
> Phone: +91-022 2278 2519 / 2121
> shisheer at tifr.res.in
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From Rob.Moore at aeriandi.com  Tue Mar  3 13:27:20 2015
From: Rob.Moore at aeriandi.com (Rob Moore)
Date: Tue, 3 Mar 2015 10:27:20 +0000
Subject: [Freeswitch-users] ICE SDP Params
Message-ID: <DB3PR05MB0286C6B3C5BC977C6BFB48383110@DB3PR05MB028.eurprd05.prod.outlook.com>

Hi Brian,

I thought as much, WebRTC isn?t something we are trying to use at the moment (although im sure we?ll find a use for it in the not too distant future.)

Invites are created using the bridge application in XML dialplan.

I have SIP pcaps but I don?t have and Freeswitch traces at the moment as the issue is only appearing once in say 500 calls on our production system so it can be a little awkward to pin down detailed tracing.
I?m working on getting an example today and will post back as soon as possible.

Is there any way to disable WebRTC entirely? That could be worth a try whilst I get a test setup for this scenario.


From: Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>>
Date: Mon, Mar 2, 2015 at 6:15 PM
Subject: Re: [Freeswitch-users] ICE SDP Params
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>


This is all related to WebRTC, how are you creating the invite?  Logs would be helpful./b



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/c2dca0b6/attachment-0001.html 

From idokan at gmail.com  Tue Mar  3 13:27:40 2015
From: idokan at gmail.com (ik)
Date: Tue, 3 Mar 2015 12:27:40 +0200
Subject: [Freeswitch-users] cherry pick calls
In-Reply-To: <CACOYK=g5XzFN7s228rypSu2G1XkMpH9yRf=HxcEycRRVVfNT3w@mail.gmail.com>
References: <CAOccsKCNn5m7+SOK9PtaGqqnAfYBMegXYbsWjxBhNiukEDi1pA@mail.gmail.com>
	<CACOYK=g5XzFN7s228rypSu2G1XkMpH9yRf=HxcEycRRVVfNT3w@mail.gmail.com>
Message-ID: <CAOccsKBrOdA9U6v_1TAAeB=9pov0jNV6XaXjvy+RO3JvVrwQVQ@mail.gmail.com>

On Mon, Mar 2, 2015 at 4:16 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> you can let the inbound calls play MOH, and use ESL to uuid_break and
> uuid_bridge the ones you need.
>


Thank you for the answer, but I'm unsure how to hold the calls with MOH,
unless it's some sort of queue or conference.


>
> On Mon, Mar 2, 2015 at 2:07 PM, ik <idokan at gmail.com> wrote:
> > Hello,
> >
> > I'm looking for a way to have some sort of queue that I can cherry pick a
> > specific caller that I wish to bridge with a specific member.
> >
> > The only way I can think of, is by using valet parking, is there another
> way
> > to do it, that is simpler?
> >
> > Thanks,
> > Ido
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/63a2faf9/attachment.html 

From ssinyagin at gmail.com  Tue Mar  3 13:54:35 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 11:54:35 +0100
Subject: [Freeswitch-users] cherry pick calls
In-Reply-To: <CAOccsKBrOdA9U6v_1TAAeB=9pov0jNV6XaXjvy+RO3JvVrwQVQ@mail.gmail.com>
References: <CAOccsKCNn5m7+SOK9PtaGqqnAfYBMegXYbsWjxBhNiukEDi1pA@mail.gmail.com>
	<CACOYK=g5XzFN7s228rypSu2G1XkMpH9yRf=HxcEycRRVVfNT3w@mail.gmail.com>
	<CAOccsKBrOdA9U6v_1TAAeB=9pov0jNV6XaXjvy+RO3JvVrwQVQ@mail.gmail.com>
Message-ID: <CACOYK=j0onaLnMjNPwx=WOki9UhVx1n_kmY6ERYTHUpwC6WjxQ@mail.gmail.com>

just call the application "playback" with MOH stream

On Tue, Mar 3, 2015 at 11:27 AM, ik <idokan at gmail.com> wrote:
>
>
> On Mon, Mar 2, 2015 at 4:16 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>>
>> you can let the inbound calls play MOH, and use ESL to uuid_break and
>> uuid_bridge the ones you need.
>
>
>
> Thank you for the answer, but I'm unsure how to hold the calls with MOH,
> unless it's some sort of queue or conference.
>
>>
>>
>> On Mon, Mar 2, 2015 at 2:07 PM, ik <idokan at gmail.com> wrote:
>> > Hello,
>> >
>> > I'm looking for a way to have some sort of queue that I can cherry pick
>> > a
>> > specific caller that I wish to bridge with a specific member.
>> >
>> > The only way I can think of, is by using valet parking, is there another
>> > way
>> > to do it, that is simpler?
>> >
>> > Thanks,
>> > Ido
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lists at kavun.ch  Tue Mar  3 15:54:23 2015
From: lists at kavun.ch (Emrah)
Date: Tue, 3 Mar 2015 13:54:23 +0100
Subject: [Freeswitch-users] Random calls failing with WRONG_CALL_STATe
	when using TLS
In-Reply-To: <CAEJMVkDJWfGufv6=VBcQKzX1vAnfrjYYQtAruu1Y3pqkVXe+LA@mail.gmail.com>
References: <45FAC76E-D2B7-483A-88AB-9FB98600C42B@kavun.ch>
	<2B416F4C-561E-48E1-A31D-BB82854AB84E@kavun.ch>
	<A77C5D22-4305-412C-B4F2-77DDA3D620ED@kavun.ch>
	<7AB693FD-921B-43F0-81B4-41610CC5A4C3@freeswitch.org>
	<F8F22522-3F29-4479-8315-EA2189E622BF@kavun.ch>
	<CAEJMVkDJWfGufv6=VBcQKzX1vAnfrjYYQtAruu1Y3pqkVXe+LA@mail.gmail.com>
Message-ID: <001E3109-E3ED-481E-A456-64B9603A8A44@kavun.ch>

Hey Brian, just saw this message.
There is no other UA in between FS and the endpoint. There is a regular NAT, that's all.
What seems to happen is:
endpoint -> FS: invite = ok
FS -> endpoint: 407 = OK
Endpoint -> FS: invite = Fails with SSL error.

What are the components I should capture to open up a Jira? FS Logs, FS Siptrace, anything else?

Thanks!
> On Feb 16, 2015, at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
> 
> Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD
> 
> Your issue is the contact has no port 443 or transport=tls right?  What sits between FS and the endpoint?
> 
> On Sun, Feb 15, 2015 at 5:38 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
> Thanks Ken. Is there a way to filter the SIP trace? It's a busy box.
> 
>> On Feb 14, 2015, at 3:35 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>> 
>> Open a jire with a full debug login including sip tracing on
>> 
>> Sent from my iPhone
>> 
>> On Feb 13, 2015, at 7:57 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>> 
>>> Hi,
>>> The issue is persistent. I am curious to know if anyone else on the list is experiencing this. It doesn't seem to have been reported before.
>>> Should I dedicate a profile to TLS use only?
>>> I also posted a message on the list about receiving options packet with the wrong transport. Are these 2 issues connected? Here is a copy paste of my message:
>>> 
>>> My experience with FS and TLS has been rather mixed so far. It's been a little inconsistent in keeping NAT sessions up and users discoverable.
>>> One thing I've noticed is that FS advertises the wrong information in option packets. The following is what I receive over my TLS session which is working on port 443.
>>> 1.2.3.4:443 <http://1.2.3.4:443/> -(SIP over TLS)-> 10.0.0.99:51132 <http://10.0.0.99:51132/>
>>> OPTIONS sip:53178246 at 10.0.0.99:56494;transport=tls;received=5.6.7.8:51132 <> SIP/2.0
>>> Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD
>>> Route: <sip:53178246 at 5.6.7.8:51132 <>>;transport=tls
>>> Max-Forwards: 70
>>> From: <sip:mod_sofia at 1.2.3.4:5060 <>>;tag=Q6XDFHeUUrcHD
>>> To: <sip:user at domain.com <>>
>>> Call-ID: 0a052f23-34a8-4158-8c88-fd2a70ffb561_c2RhaSoOYBR6jfJe4ndLoTTKJMrO2gMv
>>> CSeq: 71498568 OPTIONS
>>> Contact: <sip:mod_sofia at 1.2.3.4:5060 <>>
>>> User-Agent: FreeSWITCH
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, path, replaces
>>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>>> Content-Length: 0
>>> 
>>> As you can see FS stamps the packet with a port 5060... No reference to port 443 with a transport=tls.
>>> 
>>> What shall be done?
>>> 
>>>> On Feb 5, 2015, at 3:18 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>>>> 
>>>> Hi there,
>>>> This issue is happening all around with devices using TLS. It's not very frequent with softphones, but not inexistant.
>>>> Any pointers would be greatly appreciated. Do you have  best practice configs you'd like to share?
>>>> 
>>>> Thanks
>>>>> On Jan 30, 2015, at 6:10 PM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>> wrote:
>>>>> 
>>>>> Hi all,
>>>>> I am facing a very frustrating issue. I often have to dial twice when using my Yealink phone with TLS because the first attempt times out.
>>>>> The logs on the Yealink indicate that the first invite is successfully received, to which my FS sends a 100 trying and 407 proxy auth required. It is subsequently when my phone sends back the invite that the connection crashes with the following error:
>>>>> SSL ERROR SYSCALL
>>>>> 
>>>>> Is this something common? Why does the SSL connection crashes when the phone attempts to send the second invite? My phone is behind NAT.
>>>>> 
>>>>> It is going to be a crazy expedition to collect the logs and Pastebin them, so I am tempting my luck on the list first to see if you have any pointers.
>>>>> 
>>>>> As a last piece, my Bria on my iPHone, among other clients, never had this issue. I did experience it from time to time with Blink on Mac OS X.
>>>>> 
>>>>> Any help appreciated.
>>>>> 
>>>>> Emrah
>>>> 
>>> 
>>> _________________________________________________________________________
>>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/b712dd7d/attachment-0001.html 

From brian at freeswitch.org  Tue Mar  3 16:38:02 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 08:38:02 -0500
Subject: [Freeswitch-users] Random calls failing with WRONG_CALL_STATe
 when using TLS
In-Reply-To: <001E3109-E3ED-481E-A456-64B9603A8A44@kavun.ch>
References: <45FAC76E-D2B7-483A-88AB-9FB98600C42B@kavun.ch>
	<2B416F4C-561E-48E1-A31D-BB82854AB84E@kavun.ch>
	<A77C5D22-4305-412C-B4F2-77DDA3D620ED@kavun.ch>
	<7AB693FD-921B-43F0-81B4-41610CC5A4C3@freeswitch.org>
	<F8F22522-3F29-4479-8315-EA2189E622BF@kavun.ch>
	<CAEJMVkDJWfGufv6=VBcQKzX1vAnfrjYYQtAruu1Y3pqkVXe+LA@mail.gmail.com>
	<001E3109-E3ED-481E-A456-64B9603A8A44@kavun.ch>
Message-ID: <CAEJMVkB=GZ2K-zAN5vK91aj0HBbN-0AeuV3tJTZg0nWpGb+3fg@mail.gmail.com>

sofia global siptrace on
sofia loglevel all 9

Then outline the scenario and config on the JIRA.

On Tue, Mar 3, 2015 at 7:54 AM, Emrah <lists at kavun.ch> wrote:

> Hey Brian, just saw this message.
> There is no other UA in between FS and the endpoint. There is a regular
> NAT, that's all.
> What seems to happen is:
> endpoint -> FS: invite = ok
> FS -> endpoint: 407 = OK
> Endpoint -> FS: invite = Fails with SSL error.
>
> What are the components I should capture to open up a Jira? FS Logs, FS
> Siptrace, anything else?
>
> Thanks!
>
> On Feb 16, 2015, at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
>
> Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD
>
> Your issue is the contact has no port 443 or transport=tls right?  What
> sits between FS and the endpoint?
>
> On Sun, Feb 15, 2015 at 5:38 AM, Emrah <lists at kavun.ch> wrote:
>
>> Thanks Ken. Is there a way to filter the SIP trace? It's a busy box.
>>
>> On Feb 14, 2015, at 3:35 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> Open a jire with a full debug login including sip tracing on
>>
>> Sent from my iPhone
>>
>> On Feb 13, 2015, at 7:57 PM, Emrah <lists at kavun.ch> wrote:
>>
>> Hi,
>> The issue is persistent. I am curious to know if anyone else on the list
>> is experiencing this. It doesn't seem to have been reported before.
>> Should I dedicate a profile to TLS use only?
>> I also posted a message on the list about receiving options packet with
>> the wrong transport. Are these 2 issues connected? Here is a copy paste of
>> my message:
>>
>> My experience with FS and TLS has been rather mixed so far. It's been a
>> little inconsistent in keeping NAT sessions up and users discoverable.
>> One thing I've noticed is that FS advertises the wrong information in
>> option packets. The following is what I receive over my TLS session which
>> is working on port 443.
>> 1.2.3.4:443 -(SIP over TLS)-> 10.0.0.99:51132
>> OPTIONS sip:53178246 at 10.0.0.99:56494;transport=tls;received=5.6.7.8:51132
>>  SIP/2.0
>> Via: SIP/2.0/TLS 1.2.3.4:443;branch=z9hG4bK6Kv171Q3U5rrD
>> Route: <sip:53178246 at 5.6.7.8:51132>;transport=tls
>> Max-Forwards: 70
>> From: <sip:mod_sofia at 1.2.3.4:5060>;tag=Q6XDFHeUUrcHD
>> To: <sip:user at domain.com>
>> Call-ID:
>> 0a052f23-34a8-4158-8c88-fd2a70ffb561_c2RhaSoOYBR6jfJe4ndLoTTKJMrO2gMv
>> CSeq: 71498568 OPTIONS
>> Contact: <sip:mod_sofia at 1.2.3.4:5060>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
>> line-seize, call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Content-Length: 0
>>
>> As you can see FS stamps the packet with a port 5060... No reference to
>> port 443 with a transport=tls.
>>
>> What shall be done?
>>
>> On Feb 5, 2015, at 3:18 PM, Emrah <lists at kavun.ch> wrote:
>>
>> Hi there,
>> This issue is happening all around with devices using TLS. It's not very
>> frequent with softphones, but not inexistant.
>> Any pointers would be greatly appreciated. Do you have  best practice
>> configs you'd like to share?
>>
>> Thanks
>>
>> On Jan 30, 2015, at 6:10 PM, Emrah <lists at kavun.ch> wrote:
>>
>> Hi all,
>> I am facing a very frustrating issue. I often have to dial twice when
>> using my Yealink phone with TLS because the first attempt times out.
>> The logs on the Yealink indicate that the first invite is successfully
>> received, to which my FS sends a 100 trying and 407 proxy auth required. It
>> is subsequently when my phone sends back the invite that the connection
>> crashes with the following error:
>> SSL ERROR SYSCALL
>>
>> Is this something common? Why does the SSL connection crashes when the
>> phone attempts to send the second invite? My phone is behind NAT.
>>
>> It is going to be a crazy expedition to collect the logs and Pastebin
>> them, so I am tempting my luck on the list first to see if you have any
>> pointers.
>>
>> As a last piece, my Bria on my iPHone, among other clients, never had
>> this issue. I did experience it from time to time with Blink on Mac OS X.
>>
>> Any help appreciated.
>>
>> Emrah
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/cfb3a3bd/attachment.html 

From brian at freeswitch.org  Tue Mar  3 16:39:18 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 08:39:18 -0500
Subject: [Freeswitch-users] ICE SDP Params
In-Reply-To: <DB3PR05MB0286C6B3C5BC977C6BFB48383110@DB3PR05MB028.eurprd05.prod.outlook.com>
References: <DB3PR05MB0286C6B3C5BC977C6BFB48383110@DB3PR05MB028.eurprd05.prod.outlook.com>
Message-ID: <CAEJMVkAG+Jn07upn3JQLyhs7si7BfCb8fn0DybOv9Ce28cTUHQ@mail.gmail.com>

Thats odd, do you happen to know if the inbound call had an SAVPF?  It
shouldn't enable that unless it smells webrtc in the SDP.  Have you ever
enabled XML CDR's?  Those would help narrow this down probably.

On Tue, Mar 3, 2015 at 5:27 AM, Rob Moore <Rob.Moore at aeriandi.com> wrote:

>  Hi Brian,
>
>
>
> I thought as much, WebRTC isn?t something we are trying to use at the
> moment (although im sure we?ll find a use for it in the not too distant
> future.)
>
>
>
> Invites are created using the bridge application in XML dialplan.
>
>
>
> I have SIP pcaps but I don?t have and Freeswitch traces at the moment as
> the issue is only appearing once in say 500 calls on our production system
> so it can be a little awkward to pin down detailed tracing.
>
> I?m working on getting an example today and will post back as soon as
> possible.
>
>
>
> Is there any way to disable WebRTC entirely? That could be worth a try
> whilst I get a test setup for this scenario.
>
>
>
>
> From: *Brian West* <brian at freeswitch.org>
> Date: Mon, Mar 2, 2015 at 6:15 PM
> Subject: Re: [Freeswitch-users] ICE SDP Params
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> This is all related to WebRTC, how are you creating the invite?  Logs
> would be helpful./b
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/b4c793fc/attachment-0001.html 

From brian at freeswitch.org  Tue Mar  3 16:42:18 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 08:42:18 -0500
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
	<CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
	<CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
Message-ID: <CAEJMVkDExnoQdKgbpNq1xaRUgkzzdYr+WAYrZ0-iKzuk4rbshA@mail.gmail.com>

You'll need to use a domain name or use force-register-domain and
force-register-db-domain to force the auth into a specific domain, the
vanilla configs do this already so you've made extra steps to undo that.

In addition I don't think we've ever added ipv6 ACL support either, so
thats one that needs to be done at some point.

On Tue, Mar 3, 2015 at 4:23 AM, Shisheer Teli <telishisheer at gmail.com>
wrote:

> Hi Team,
>
> My freeswitch server is on IPv6, and now i am able register extension with
> IPv6 in freeswitch.
>
> but  i am unable to call from IPv6 extensions..
>
> Error:
> 015-03-03 14:49:21.568064 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal-ipv6/1102@[clientipv6address]:5060
> [60707716-c186-11e4-88f0-adeca182559b]
> 2015-03-03 14:49:21.588040 [WARNING] sofia_reg.c:2827 Can't find user
> [1102@[serveripv6address]] from clientipv6address
> You must define a domain called '[serveripv6address]' in your directory
> and add a user with the id="1102" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2015-03-03 14:49:21.588040 [NOTICE] sofia.c:2063 Hangup
> sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_NEW] [CALL_REJECTED]
> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1641 Session 1
> (sofia/internal-ipv6/1102@[clientipv6address]:5060) Ended
> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_DESTROY]
>
> Regards,
> Shisheer
>
>
> On Tue, Mar 3, 2015 at 2:39 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> but you didn't provide any information, so it's difficult to help.
>>
>> On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com>
>> wrote:
>> > Hi Team,
>> >
>> > My freeswitch server is on IPv6, and now i am able register extension
>> with
>> > IPv6 in freeswitch.
>> >
>> > but  i am unable to call from IPv6 extensions..
>> >
>> > can help ..?
>> >
>> > Regards,
>> > shisheer T
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards,
> Shisheer Teli
> Phone: +91-022 2278 2519 / 2121
> shisheer at tifr.res.in
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/dc7aa7b5/attachment.html 

From brian at freeswitch.org  Tue Mar  3 16:46:14 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 08:46:14 -0500
Subject: [Freeswitch-users]
	=?utf-8?q?Implementing_telecom=E2=80=8B_module?=
	=?utf-8?q?_with_FreeSwitch?=
In-Reply-To: <1425412349467.9202@nexxuspg.com>
References: <1424911092386.69803@nexxuspg.com>
	<1425412349467.9202@nexxuspg.com>
Message-ID: <CAEJMVkB=dkcampLOKer9r5QC6FeANr+v+f9SXe_+aa_arw=F1w@mail.gmail.com>

Most of what you want is either on our wiki/confluence or on the sangoma
wiki, maybe you can narrow down your request once you attempt to deploy /
implement the solution?

On Tue, Mar 3, 2015 at 1:21 AM, Manish Talwar <manish.talwar at nexxuspg.com>
wrote:

>  Hello,
>
>
>  Please suggest me about my below mentioned email.
>
>
>  Thanks,
>
>
>  Regards,?
>
> Manish Talwar
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Manish Talwar
> <manish.talwar at nexxuspg.com>
> *Sent:* 25 February 2015 16:37
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Implementing telecom? module with FreeSwitch
>
>
> Hi,
>
>
>  We have successfully implemented FreeSwitch with our IVR application
> using httapi module and its running fine now. After implementing IVR
> application, we are looking for implementing telecom? module with
> FreeSwitch now.
>
>
>  We have a plan to activating a set of 20 number, for this we have a
> "sangoma wanpipe driver" on the server and some kernel modules loaded that
> will communicate with a Sangoma A104 card installed there. Incoming Qatari
> +974 phone calls will arrive and will be translated on our server to SIP
> traffic for our IVR system to process.  One or more of the numbers will be
> reserved as office numbers.
>
>
>  I have looked into telecom service of FreeSwitch and found freeTDM
> module for implementing telecom with FreeSwitch.
>
>
>  Can we achieve this telecom implementation by freeTDM module of
> FreeSwitch? If yes, then please help me for implementing ?telecom? module ?with
> FreeSwitch with freeTDM and let me know all details about it. Also, please
> let me know any other useful information regarding this module.
>
>
>  Thanks,
>
>
>  Regards,
>
> Manish Talwar
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/34403382/attachment-0001.html 

From brian at freeswitch.org  Tue Mar  3 16:51:04 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 08:51:04 -0500
Subject: [Freeswitch-users] Help required on FXO FXS module
In-Reply-To: <EFAFF307-0612-4868-A22F-CA8636D8CC61@gmail.com>
References: <EFAFF307-0612-4868-A22F-CA8636D8CC61@gmail.com>
Message-ID: <CAEJMVkB8ecgso82HP6sTAEQamwdgLUa47wS-8q_wOtNVAokBhg@mail.gmail.com>

This is not really the purpose of this mailing list.  Have you looked at
whats out there already?

On Mon, Mar 2, 2015 at 10:40 PM, Naveen Khanna <naveen.khanna.bm at gmail.com>
wrote:

> Hi,
>
> Can someone help source or design FXO FXS modules or boards at around $5
> per port with command line interface.
>
> Regards,
>
> Naveen Khanna
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/447bd638/attachment.html 

From dylan at onsip.com  Tue Mar  3 18:35:47 2015
From: dylan at onsip.com (Dylan Mikus)
Date: Tue, 3 Mar 2015 10:35:47 -0500
Subject: [Freeswitch-users] Determining if Freeswitch channel is using a
	video codec
In-Reply-To: <CAOQuP_3XocnTPOcMyZ-4ueu5NJy8d8=NQ3Snmgi+Y8ATW8e43Q@mail.gmail.com>
References: <CAOQuP_25HNSoP0BPAhCVSAKzyADCOwoGnNoty34170dDxRT9CA@mail.gmail.com>
	<CAOQuP_3XocnTPOcMyZ-4ueu5NJy8d8=NQ3Snmgi+Y8ATW8e43Q@mail.gmail.com>
Message-ID: <CAOQuP_04bm9gWXEi1pmOLvYnrLkvdhAjiRh2d7s-k-hMFeM_QQ@mail.gmail.com>

So, poking around, I might have found a solution:

eval uuid:<THE_CHANNEL_UUID> ${variable_video_read_codec}

Other possible variables to check:

variable_video_possible
variable_video_read_codec
variable_video_write_codec
variable_rtp_last_video_codec_string
variable_rtp_use_video_codec_name
variable_rtp_use_video_codec_fmtp

?

On Mon, Mar 2, 2015 at 4:35 PM, Dylan Mikus <dylan at onsip.com> wrote:

> Actually, I do not necessarily need the codec. I only need to determine if
> a call is using video or not. The codec is useful additional information,
> but not necessary.
>
> On Mon, Mar 2, 2015 at 4:28 PM, Dylan Mikus <dylan at onsip.com> wrote:
>
>> I?m trying to determine if a given channel over Freeswitch is using a
>> video codec. In my config/vars.xml file, I?ve set the codecs line to:
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264"/>
>>
>> Logs
>>
>> My SDP negotiation appears to be correct. The INVITE:
>>
>> INVITE sip:queuecard at cyberdyne.onsip.com SIP/2.0
>> Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
>> Max-Forwards: 70
>> To: <sip:queuecard at cyberdyne.onsip.com>
>> From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
>> Call-ID: n3o4g4i724sq7qkekp07
>> CSeq: 8622 INVITE
>> Proxy-Authorization: Digest algorithm=MD5, username="cyberdyne_bender", realm="jnctn.net", nonce="54f4ce2e000013e4888519dec3ca2ee1ef9023f82d4d8922", uri="sip:queuecard at cyberdyne.onsip.com", response="f842951ecc11c3510d1e1b7abcdeb51f", qop=auth, cnonce="d153n6udlh74", nc=00000001
>> Contact: <sip:1fjqm8ps at o8iatftbl1mn.invalid;transport=ws;ob>
>> Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE
>> Content-Type: application/sdp
>> Supported: 100rel,outbound
>> User-Agent: SIP.js/0.6.3 InstaPhone
>> Content-Length: 1649
>>
>> v=0
>> o=Mozilla-SIPUA-35.0.1 10886 0 IN IP4 0.0.0.0
>> s=SIP Call
>> t=0 0
>> a=ice-ufrag:a256418b
>> a=ice-pwd:62e2ae7154b57f00ed0b1a2003ccf7af
>> a=fingerprint:sha-256 EA:C4:92:D4:94:62:18:41:39:2E:42:B4:4E:B7:32:9E:66:FE:7C:01:57:AC:2C:4C:E4:66:4F:3B:B6:91:FA:DC
>> m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
>> c=IN IP4 0.0.0.0
>> a=rtpmap:109 opus/48000/2
>> a=ptime:20
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=setup:actpass
>> a=rtcp-mux
>> a=candidate:0 1 UDP 2130379007 192.168.1.38 58531 typ host
>> a=candidate:0 2 UDP 2130379006 192.168.1.38 64677 typ host
>> a=candidate:1 1 UDP 1694236671 38.104.167.182 49209 typ srflx raddr 192.168.1.38 rport 58531
>> a=candidate:1 2 UDP 1694236670 38.104.167.182 51209 typ srflx raddr 192.168.1.38 rport 64677
>> m=video 9 UDP/TLS/RTP/SAVPF 120 126 97
>> c=IN IP4 0.0.0.0
>> a=rtpmap:120 VP8/90000
>> a=rtpmap:126 H264/90000
>> a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 profile-level-id=42e01f
>> a=sendrecv
>> a=rtcp-fb:120 nack
>> a=rtcp-fb:120 nack pli
>> a=rtcp-fb:120 ccm fir
>> a=rtcp-fb:126 nack
>> a=rtcp-fb:126 nack pli
>> a=rtcp-fb:126 ccm fir
>> a=rtcp-fb:97 nack
>> a=rtcp-fb:97 nack pli
>> a=rtcp-fb:97 ccm fir
>> a=setup:actpass
>> a=rtcp-mux
>> a=candidate:0 1 UDP 2130379007 192.168.1.38 59562 typ host
>> a=candidate:0 2 UDP 2130379006 192.168.1.38 61464 typ host
>> a=candidate:1 1 UDP 1694236671 38.104.167.182 59357 typ srflx raddr 192.168.1.38 rport 59562
>> a=candidate:1 2 UDP 1694236670 38.104.167.182 21168 typ srflx raddr 192.168.1.38 rport 61464
>>
>> The 200 OK response:
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099
>> Record-Route: <sip:38.109.82.167;lr;ftag=fneppn1lhh;did=24.0b0068f6>
>> Record-Route: <sip:38.109.82.192;lr;ftag=fneppn1lhh;did=24.4c4e8343>
>> Record-Route: <sip:38.109.82.193:5060;transport=udp;lr;ovid=cac1e79d>
>> Record-Route: <sip:2b14c410c1 at 38.109.82.193:80;transport=ws;lr;ovid=cac1e79d>
>> From: "Bender Rodriguez" <sip:bender at cyberdyne.onsip.com>;tag=fneppn1lhh
>> To: <sip:queuecard at cyberdyne.onsip.com>;tag=30yQvF62DQyyg
>> Call-ID: n3o4g4i724sq7qkekp07
>> CSeq: 8622 INVITE
>> Contact: <sip:dtlsgw0 at app2-1.55-broad-1.app.jnctn.net>
>> Accept: application/sdp
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY
>> Supported: precondition, 100rel, timer, path, replaces
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 1738
>>
>> v=0
>> o=FreeSWITCH 1425307798 1425307799 IN IP4 38.109.82.228
>> s=FreeSWITCH
>> c=IN IP4 38.109.82.228
>> t=0 0
>> a=msid-semantic: WMS 61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
>> m=audio 21882 UDP/TLS/RTP/SAVPF 109 101
>> a=rtpmap:109 opus/48000/2
>> a=fmtp:109 useinbandfec=1;usedtx=1;maxaveragebitrate=30000
>> a=rtpmap:101 telephone-event/8000
>> a=recvonly
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
>> a=rtcp:21883 IN IP4 38.109.82.228
>> a=ssrc:2365215248 cname:CkzZ9cdxFymMTiha
>> a=ssrc:2365215248 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a0
>> a=ssrc:2365215248 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
>> a=ssrc:2365215248 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwa0
>> a=ice-ufrag:zqC4ZTWsD5d4Hyqa
>> a=ice-pwd:IQ0IT35osh0bq7bPoDKenwwR
>> a=candidate:9358589392 1 udp 659136 38.109.82.228 21882 typ host generation 0
>> a=candidate:9358589392 2 udp 659134 38.109.82.228 21883 typ host generation 0
>> m=video 23680 UDP/TLS/RTP/SAVPF 126
>> b=AS:256
>> a=rtpmap:126 H264/90000
>> a=fmtp:126 profile-level-id=42e01f;packetization-mode=1
>> a=recvonly
>> a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2
>> a=rtcp:23681 IN IP4 38.109.82.228
>> a=rtcp-fb:* fir pli
>> a=ssrc:1652571152 cname:CkzZ9cdxFymMTiha
>> a=ssrc:1652571152 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw v0
>> a=ssrc:1652571152 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw
>> a=ssrc:1652571152 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwv0
>> a=ice-ufrag:kirEYQCPyInSpi7Y
>> a=ice-pwd:fbIhKWJB3fFuGVyQ4QlSwNxU
>> a=candidate:9055446981 1 udp 659136 38.109.82.228 23680 typ host generation 0
>> a=candidate:9055446981 2 udp 659134 38.109.82.228 23681 typ host generation 0
>>
>> We offer:
>>
>> a=rtpmap:120 VP8/90000
>> a=rtpmap:126 H264/90000
>>
>> and we accept:
>>
>> a=rtpmap:126 H264/90000
>>
>> Note that this is on Firefox 35.0.1.
>> Response
>>
>> While this call is up, I run show channels in fs_cli and get the
>> following:
>>
>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
>> b41772a4-c11b-11e4-a78f-7585ed98a76c,inbound,2015-03-02 20:35:45,1425328545,sofia/sip0/bender at cyberdyne.onsip.com,CS_SOFT_EXECUTE,Bender Rodriguez,bender,38.109.82.167,queuecard,uuid_bridge,bc7c6d64-c11b-11e4-a798-7585ed98a76c,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Outbound Call,terabithia
>> bc7c6d64-c11b-11e4-a798-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/app/c3po at cyberdyne.onsip.com,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,1000,uuid_bridge,b41772a4-c11b-11e4-a78f-7585ed98a76c,XML,generic-app,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Bender Rodriguez,bender
>> bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,inbound,2015-03-02 20:35:59,1425328559,sofia/sip0/bender at cyberdyne.onsip.com,CS_EXECUTE,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,bridge,{force_transfer_context=refer}sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid;transport=ws;aor=c3po%40cyberdyne.onsip.com,XML,default,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Outbound Call,gl8k15o7
>> bc82805a-c11b-11e4-a7ae-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/sip0/sip:gl8k15o7 at e6kin9qicasi.invalid,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,,,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Bender Rodriguez,bender
>>
>> The read codecs and write codecs are OPUS, except for a websocket
>> transport that lists an XML codec, I think. Is something up with my setup,
>> or do we only show the audio codec being used when we run the show
>> channels command? Any other idea for how to determine whether a
>> Freeswitch channel is using video? I?m trying to stay away from sending
>> custom headers and I want to be able to figure this out within Freeswitch.
>> In other words, I don?t want a receiving application to try to figure out
>> whether it is in video or not. I just want to query my Freeswitch service
>> to find out.
>>
>> Thanks, guys! I appreciate any help.
>> ?
>>
>> --
>> Dylan Mikus
>> Software Engineer
>> OnSIP
>> www.onsip.com
>> p. 212.933.9190 x7060
>> SIP/Email: dylan at onsip.com
>>
>
>
>
> --
> Dylan Mikus
> Software Engineer
> OnSIP
> www.onsip.com
> p. 212.933.9190 x7060
> SIP/Email: dylan at onsip.com
>



-- 
Dylan Mikus
Software Engineer
OnSIP
www.onsip.com
p. 212.933.9190 x7060
SIP/Email: dylan at onsip.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/39322460/attachment-0001.html 

From victor.chukalovskiy at gmail.com  Tue Mar  3 18:36:17 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Tue, 03 Mar 2015 10:36:17 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.4.15 Released
In-Reply-To: <54a2fc7d8af73_738fdd33302001a@ip-10-146-191-228.mail>
References: <54a2fc7d8af73_738fdd33302001a@ip-10-146-191-228.mail>
Message-ID: <54F5D4F1.9080200@gmail.com>

Good day,

Am I on a wrong branch, or FS stable release did not have any commits 
since the New Year?
When doing "make current" I'm still on this version:

Version 1.4.15 git 507a0f2 2014-12-29

Thanks!
-Victor

On 14-12-30 02:26 PM, Ken Rice wrote:
> New Post on freeswitch.org from krice387
> check it out at http://freeswitch.org/freeswitch-1-4-15-released/
> FreeSWITCH 1.4.15 Released
>
> FreeSWITCH 1.4.14 has been released!
>
> This is routine maintenance release.
>
> Source Tarball available at 
> http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2
>
> Debian and Yum Repos have been updated as well.
>
> See the release notes below for a list of notable changes.
>
> Happy New Years From the FreeSWITCH Team!
>
> Release Notes:
>
> New features that were added:
>
>   * e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira:
>     https://jira.freeswitch.org/browse/FS-7025]
>   * a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh}
>   * 17574a8 Add bert stats to mod_bert::lost_sync event
>   * a26e29c vs2010 support for recent unimrcp changes
>   * cee8b30 Set rtp_has_crypto for dtls calls
>   * 5fcff50 FS-7093 Create uuid_drop_dtmf [Jira:
>     https://jira.freeswitch.org/browse/FS-7093]
>   * f024ea3 FS-7047 Arbitrary MRCP headers can now be sent to unimrcp
>     input components in mod_rayo [Jira:
>     https://jira.freeswitch.org/browse/FS-7047]
>   * e783999 Some changes to webrtc to make it work with iDoubs in
>     rtcweb profile mode
>   * d189e98 Allow 10ms jb
>  *
>       o 750b1dd FS-7114 Allow streaming binary data from mod_memcache
>
> Improvements in performance:
>
>   * 4bcf1d8 Use cached time to save cpu
>
> Improvements in cross platform build supports:
>
>   * 32c27b3 Added a Debian dependency to the CentOS6 makefile
>   * f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build
>     when relative path is used in configure.gnu ?with-sofia-sip
>   * [Jira: https://jira.freeswitch.org/browse/FS-7031]
>   * 061f3cb FS-7031 #resolve #comment [unimrcp] update library again
>     to pull in upstream fix for ?with-sofia-sip=../sofia-sip
>   * [Jira: https://jira.freeswitch.org/browse/FS-7031]
>   * 382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order
>     to not break x86 builds.
>   * dc9e904 FS-7025 fix compiler warning introduced from e55aee14
>     [Jira: https://jira.freeswitch.org/browse/FS-7025]
>   * b69c93e FS-7030 More work toward fixing FS build on Windows Visual
>     Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030]
>   * db66cdb Fix mrcp libraries to build correctly
>   * c327455 FS-7030 More work toward getting FS to build on Windows
>     Visual Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030]
>   * b341ff7 FS-7046: fix data types and casting on some vars to
>     silence windows build warnings in mod_verto [Jira:
>     https://jira.freeswitch.org/browse/FS-7046]
>   * 7ce5171 FS-7046 follow up on type change in mod_verto [Jira:
>     https://jira.freeswitch.org/browse/FS-7046]
>   * 357ffad Fix windows build error
>   * 0b414a8 vs2010 unimrcp working build
>   * 0c1e698 Update build deps for debian list
>   * 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used
>     error in mod_commands
>   * af6b23a FS-7046 Fix some additional Windows build warnings for
>     mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046]
>
> Additional documentation:
>
>   * f63f868 FS-7049 ? Documentation for state optional paramenter in
>     callcenter_config queue list and count [Jira:
>     https://jira.freeswitch.org/browse/FS-7049]
>
> In terms of stability these were the use cases that were fixed:
>
>   * 392c687 FS-7055 Fix for a stability race condition in FS [Jira:
>     https://jira.freeswitch.org/browse/FS-7055]
>   * d5119a7 FS-7091 Removed unnecessary mutex lock inside input
>     component?s cleanup function since the input component won?t be
>     cleaned up unless all references have been released, in mod_rayo
>     [Jira: https://jira.freeswitch.org/browse/FS-7091]
>
> These were the packaging improvements:
>
>   * 3c8dd3e Handle missing `lsb_release`
>   * 505cd29 Refactor distro detection and handling
>   * 430433a Improve error message
>   * d88bae1 Support optional debian parallel builds
>
> The following bugs were squashed:
>
>   * 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia
>     [Jira: https://jira.freeswitch.org/browse/FS-7015]
>   * 99a5b50 FS-7063 Fix for media delay issue [Jira:
>     https://jira.freeswitch.org/browse/FS-7063]
>   * 21458f8 FS-7062 On redirect, when uri are passed in without <>
>     with multiple uris, automatically add the q= header param in
>     decending order in mod_sofia. [Jira:
>     https://jira.freeswitch.org/browse/FS-7062]
>   * 5376e82 FS-6688 This will fix the normal case of record route from
>     a proxy without breaking normal changing of a contact in mod_sofia
>     [Jira: https://jira.freeswitch.org/browse/FS-6688]
>   * 06c241a FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close
>     #comment All of these bugs are invalidated due to a botched revert
>     [Jira: https://jira.freeswitch.org/browse/FS-6891]
>   * 922fd81 FS-7015 The code was not properly catching the 0.0.0.0
>     after changing it to work with ICE SDPs because it was looking in
>     the wrong place for the 0.0.0.0 [Jira:
>     https://jira.freeswitch.org/browse/FS-7015]
>   * 3d515cf Re-mark cur_payload as negotiated when detected as such by
>     parser or the rtp could stop working on session re-invite
>   * 19272dc FS-7078 Fix sip_header_as_string to properly
>     null_terminate on larger header strings [Jira:
>     https://jira.freeswitch.org/browse/FS-7078]
>   * e268a72 FS-6994 Fix for Codec OPUS decoder error in mod_opus
>     [Jira: https://jira.freeswitch.org/browse/FS-6994]
>   * 6dbb416 FS-7086 FS-6798 Fix for invalid codec tearing down the
>     call request [Jira: https://jira.freeswitch.org/browse/FS-7086]
>   * 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012
>     project files added by FS project [Jira:
>     https://jira.freeswitch.org/browse/FS-7030]
>   * bad5dc3 FS-7037 Fix for T38 fax break started by commit
>     5bbef7f1e50 [Jira: https://jira.freeswitch.org/browse/FS-7037]
>   * 72c3df5 FS-6891 FS-6713 #comment revert
>     1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira:
>     https://jira.freeswitch.org/browse/FS-6891]
>   * 2a7b022 FS-6980 #resolve don?t crash when using native recording
>     on recordstop the redo [Jira:
>     https://jira.freeswitch.org/browse/FS-6980]
>   * 35ba6a3 FS-6766 Fix verto caller ringback missing on conference
>     bridge in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-6766]
>   * e8cf9c7 FS-7045 Guarantee that dialed call can be joined when
>     answered event is sent in mod_rayo [Jira:
>     https://jira.freeswitch.org/browse/FS-7045]
>   * 4be6290 FS-7052 Moving jb queue swap operation out of the debug
>     block. [Jira: https://jira.freeswitch.org/browse/FS-7052]
>   * 843e495 FS-7051 Preserve the annexb=no/yes status in
>     mod_sangoma_codec [Jira: https://jira.freeswitch.org/browse/FS-7051]
>   * 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent
>     ptimes present and record session starts on bleg [Jira:
>     https://jira.freeswitch.org/browse/FS-7002]
>   * 4ce2ce3 FS-7092 Fixed bug with Comrex OPUS [Jira:
>     https://jira.freeswitch.org/browse/FS-7092]
>   * d786490 Fix timestamps in mod_bert broken by the cpu improvements
>     refactoring
>   * ba016c2 FS-7095 Fix for FS sending DTLS HELLO (and STUN binding
>     request) to wrong port [Jira:
>     https://jira.freeswitch.org/browse/FS-7095]
>   * e0dcd17 FS-7083 #comment patch to change mod_shout to use
>     lame_encode_buffer_interleaved on stereo channels so we don?t have
>     to mess with the input data [Jira:
>     https://jira.freeswitch.org/browse/FS-7083]
>   * 326289c FS-7083 This patch adds a dedicated thread for writing to
>     the file and the channel_variable RECORD_USE_THREAD=false will
>     disable it and sync may still be maintained at the cost of
>     dropping more data from the audio signal. [Jira:
>     https://jira.freeswitch.org/browse/FS-7083]
>   * 9fabbab Disable hard-mute when a session has a media bug attached
>   * 0200bc1 FS-7083 Fix divide by zero [Jira:
>     https://jira.freeswitch.org/browse/FS-7083]
>   * 067cb0f FS-7100 Make buffer for sub contact big enough in
>     mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7100]
>   * 7798b2f FS-6984 Set default video rates [Jira:
>     https://jira.freeswitch.org/browse/FS-6984]
>   * 763e6aa FS-7046 Fix warning introduced from b341ff7 [Jira:
>     https://jira.freeswitch.org/browse/FS-7046]
>   * 65e678b FS-7070 Fix mod_expr `clamp` function typo
>   * 0a66db6 FS-7111 Fix for bridge_early_media crash [Jira:
>     https://jira.freeswitch.org/browse/FS-7111]
>
> Miscellaneous commits:
>
>   * 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t
>     work when using non-source build dir.
>   * [Jira: https://jira.freeswitch.org/browse/FS-7031]
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/67f76ec7/attachment.html 

From s.safarov at gmail.com  Tue Mar  3 18:41:28 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 3 Mar 2015 15:41:28 +0000
Subject: [Freeswitch-users] module dependency
Message-ID: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>

Please help me declare module dependency
I has extended module radius_cdr by timezone support and from time to time
is getting following error

freeswitch at internal> reload mod_radius_cdr
+OK Reloading XML
+OK module unloaded
+OK module loaded

2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935
Stopping: mod_radius_cdr
2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955
mod_radius_cdr unloaded.
2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone 'Asia/Tokyo'
not found!
2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find timezone
Asia/Tokyo
, Setting timezone to GMT
2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465
Successfully Loaded [mod_radius_cdr]
2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded 1781
definitions


Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE
but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.

How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded
 before mod_radius_cdr?

Sergey
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/b18802cd/attachment-0001.html 

From gmaruzz at gmail.com  Tue Mar  3 18:43:23 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 3 Mar 2015 16:43:23 +0100
Subject: [Freeswitch-users] FreeSWITCH 1.4.15 Released
In-Reply-To: <54F5D4F1.9080200@gmail.com>
References: <54a2fc7d8af73_738fdd33302001a@ip-10-146-191-228.mail>
	<54F5D4F1.9080200@gmail.com>
Message-ID: <CALXCt0p4dFd7n4OBMYVwN1RdcnQhot5zDmhV2O0FPtnPxXcZ4g@mail.gmail.com>

All hail our Release Overlord!




On Tue, Mar 3, 2015 at 4:36 PM, Victor Chukalovskiy
<victor.chukalovskiy at gmail.com> wrote:
> Good day,
>
> Am I on a wrong branch, or FS stable release did not have any commits since
> the New Year?
> When doing "make current" I'm still on this version:
>
> Version 1.4.15 git 507a0f2 2014-12-29
>
> Thanks!
> -Victor
>
> On 14-12-30 02:26 PM, Ken Rice wrote:
>
> New Post on freeswitch.org from krice387
> check it out at http://freeswitch.org/freeswitch-1-4-15-released/
> FreeSWITCH 1.4.15 Released
>
> FreeSWITCH 1.4.14 has been released!
>
> This is routine maintenance release.
>
> Source Tarball available at
> http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2
>
> Debian and Yum Repos have been updated as well.
>
> See the release notes below for a list of notable changes.
>
> Happy New Years From the FreeSWITCH Team!
>
> Release Notes:
>
> New features that were added:
>
> e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira:
> https://jira.freeswitch.org/browse/FS-7025]
> a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh}
> 17574a8 Add bert stats to mod_bert::lost_sync event
> a26e29c vs2010 support for recent unimrcp changes
> cee8b30 Set rtp_has_crypto for dtls calls
> 5fcff50 FS-7093 Create uuid_drop_dtmf [Jira:
> https://jira.freeswitch.org/browse/FS-7093]
> f024ea3 FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input
> components in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7047]
> e783999 Some changes to webrtc to make it work with iDoubs in rtcweb profile
> mode
> d189e98 Allow 10ms jb
>
> 750b1dd FS-7114 Allow streaming binary data from mod_memcache
>
> Improvements in performance:
>
> 4bcf1d8 Use cached time to save cpu
>
> Improvements in cross platform build supports:
>
> 32c27b3 Added a Debian dependency to the CentOS6 makefile
> f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build when
> relative path is used in configure.gnu ?with-sofia-sip
> [Jira: https://jira.freeswitch.org/browse/FS-7031]
> 061f3cb FS-7031 #resolve #comment [unimrcp] update library again to pull in
> upstream fix for ?with-sofia-sip=../sofia-sip
> [Jira: https://jira.freeswitch.org/browse/FS-7031]
> 382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order to not
> break x86 builds.
> dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira:
> https://jira.freeswitch.org/browse/FS-7025]
> b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio
> 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030]
> db66cdb Fix mrcp libraries to build correctly
> c327455 FS-7030 More work toward getting FS to build on Windows Visual
> Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030]
> b341ff7 FS-7046: fix data types and casting on some vars to silence windows
> build warnings in mod_verto [Jira:
> https://jira.freeswitch.org/browse/FS-7046]
> 7ce5171 FS-7046 follow up on type change in mod_verto [Jira:
> https://jira.freeswitch.org/browse/FS-7046]
> 357ffad Fix windows build error
> 0b414a8 vs2010 unimrcp working build
> 0c1e698 Update build deps for debian list
> 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used error in
> mod_commands
> af6b23a FS-7046 Fix some additional Windows build warnings for mod_verto
> [Jira: https://jira.freeswitch.org/browse/FS-7046]
>
> Additional documentation:
>
> f63f868 FS-7049 ? Documentation for state optional paramenter in
> callcenter_config queue list and count [Jira:
> https://jira.freeswitch.org/browse/FS-7049]
>
> In terms of stability these were the use cases that were fixed:
>
> 392c687 FS-7055 Fix for a stability race condition in FS [Jira:
> https://jira.freeswitch.org/browse/FS-7055]
> d5119a7 FS-7091 Removed unnecessary mutex lock inside input component?s
> cleanup function since the input component won?t be cleaned up unless all
> references have been released, in mod_rayo [Jira:
> https://jira.freeswitch.org/browse/FS-7091]
>
> These were the packaging improvements:
>
> 3c8dd3e Handle missing `lsb_release`
> 505cd29 Refactor distro detection and handling
> 430433a Improve error message
> d88bae1 Support optional debian parallel builds
>
> The following bugs were squashed:
>
> 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia [Jira:
> https://jira.freeswitch.org/browse/FS-7015]
> 99a5b50 FS-7063 Fix for media delay issue [Jira:
> https://jira.freeswitch.org/browse/FS-7063]
> 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple
> uris, automatically add the q= header param in decending order in mod_sofia.
> [Jira: https://jira.freeswitch.org/browse/FS-7062]
> 5376e82 FS-6688 This will fix the normal case of record route from a proxy
> without breaking normal changing of a contact in mod_sofia [Jira:
> https://jira.freeswitch.org/browse/FS-6688]
> 06c241a FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close #comment All
> of these bugs are invalidated due to a botched revert [Jira:
> https://jira.freeswitch.org/browse/FS-6891]
> 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after
> changing it to work with ICE SDPs because it was looking in the wrong place
> for the 0.0.0.0 [Jira: https://jira.freeswitch.org/browse/FS-7015]
> 3d515cf Re-mark cur_payload as negotiated when detected as such by parser or
> the rtp could stop working on session re-invite
> 19272dc FS-7078 Fix sip_header_as_string to properly null_terminate on
> larger header strings [Jira: https://jira.freeswitch.org/browse/FS-7078]
> e268a72 FS-6994 Fix for Codec OPUS decoder error in mod_opus [Jira:
> https://jira.freeswitch.org/browse/FS-6994]
> 6dbb416 FS-7086 FS-6798 Fix for invalid codec tearing down the call request
> [Jira: https://jira.freeswitch.org/browse/FS-7086]
> 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project
> files added by FS project [Jira: https://jira.freeswitch.org/browse/FS-7030]
> bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira:
> https://jira.freeswitch.org/browse/FS-7037]
> 72c3df5 FS-6891 FS-6713 #comment revert
> 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira:
> https://jira.freeswitch.org/browse/FS-6891]
> 2a7b022 FS-6980 #resolve don?t crash when using native recording on
> recordstop the redo [Jira: https://jira.freeswitch.org/browse/FS-6980]
> 35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in
> mod_verto [Jira: https://jira.freeswitch.org/browse/FS-6766]
> e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event
> is sent in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7045]
> 4be6290 FS-7052 Moving jb queue swap operation out of the debug block.
> [Jira: https://jira.freeswitch.org/browse/FS-7052]
> 843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec
> [Jira: https://jira.freeswitch.org/browse/FS-7051]
> 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes
> present and record session starts on bleg [Jira:
> https://jira.freeswitch.org/browse/FS-7002]
> 4ce2ce3 FS-7092 Fixed bug with Comrex OPUS [Jira:
> https://jira.freeswitch.org/browse/FS-7092]
> d786490 Fix timestamps in mod_bert broken by the cpu improvements
> refactoring
> ba016c2 FS-7095 Fix for FS sending DTLS HELLO (and STUN binding request) to
> wrong port [Jira: https://jira.freeswitch.org/browse/FS-7095]
> e0dcd17 FS-7083 #comment patch to change mod_shout to use
> lame_encode_buffer_interleaved on stereo channels so we don?t have to mess
> with the input data [Jira: https://jira.freeswitch.org/browse/FS-7083]
> 326289c FS-7083 This patch adds a dedicated thread for writing to the file
> and the channel_variable RECORD_USE_THREAD=false will disable it and sync
> may still be maintained at the cost of dropping more data from the audio
> signal. [Jira: https://jira.freeswitch.org/browse/FS-7083]
> 9fabbab Disable hard-mute when a session has a media bug attached
> 0200bc1 FS-7083 Fix divide by zero [Jira:
> https://jira.freeswitch.org/browse/FS-7083]
> 067cb0f FS-7100 Make buffer for sub contact big enough in mod_sofia [Jira:
> https://jira.freeswitch.org/browse/FS-7100]
> 7798b2f FS-6984 Set default video rates [Jira:
> https://jira.freeswitch.org/browse/FS-6984]
> 763e6aa FS-7046 Fix warning introduced from b341ff7 [Jira:
> https://jira.freeswitch.org/browse/FS-7046]
> 65e678b FS-7070 Fix mod_expr `clamp` function typo
> 0a66db6 FS-7111 Fix for bridge_early_media crash [Jira:
> https://jira.freeswitch.org/browse/FS-7111]
>
> Miscellaneous commits:
>
> 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t work when
> using non-source build dir.
> [Jira: https://jira.freeswitch.org/browse/FS-7031]
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Tue Mar  3 18:48:31 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 03 Mar 2015 09:48:31 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.4.15 Released
In-Reply-To: <CALXCt0p4dFd7n4OBMYVwN1RdcnQhot5zDmhV2O0FPtnPxXcZ4g@mail.gmail.com>
Message-ID: <D11B33EF.95DA7%krice@freeswitch.org>

We're In the process of re-arranging things... Watch for an announcement
soon


On 3/3/15, 9:43 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> All hail our Release Overlord!




On Tue, Mar 3, 2015 at 4:36 PM, Victor
> Chukalovskiy
<victor.chukalovskiy at gmail.com> wrote:
> Good day,
>
> Am I on a
> wrong branch, or FS stable release did not have any commits since
> the New
> Year?
> When doing "make current" I'm still on this version:
>
> Version
> 1.4.15 git 507a0f2 2014-12-29
>
> Thanks!
> -Victor
>
> On 14-12-30 02:26 PM,
> Ken Rice wrote:
>
> New Post on freeswitch.org from krice387
> check it out at
> http://freeswitch.org/freeswitch-1-4-15-released/
> FreeSWITCH 1.4.15
> Released
>
> FreeSWITCH 1.4.14 has been released!
>
> This is routine
> maintenance release.
>
> Source Tarball available at
>
> http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2
>
> Debian and Yum Repos
> have been updated as well.
>
> See the release notes below for a list of
> notable changes.
>
> Happy New Years From the FreeSWITCH Team!
>
> Release
> Notes:
>
> New features that were added:
>
> e55aee1 FS-7025 Add
> drop_dtmf_masking_tone channel_variable [Jira:
>
> https://jira.freeswitch.org/browse/FS-7025]
> a8c5a0c FS-7048 Add timezone
> support to mod_say_{de,es,ja,nl,th,zh}
> 17574a8 Add bert stats to
> mod_bert::lost_sync event
> a26e29c vs2010 support for recent unimrcp
> changes
> cee8b30 Set rtp_has_crypto for dtls calls
> 5fcff50 FS-7093 Create
> uuid_drop_dtmf [Jira:
> https://jira.freeswitch.org/browse/FS-7093]
> f024ea3
> FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input
> components
> in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7047]
> e783999 Some
> changes to webrtc to make it work with iDoubs in rtcweb profile
> mode
>
> d189e98 Allow 10ms jb
>
> 750b1dd FS-7114 Allow streaming binary data from
> mod_memcache
>
> Improvements in performance:
>
> 4bcf1d8 Use cached time to
> save cpu
>
> Improvements in cross platform build supports:
>
> 32c27b3 Added
> a Debian dependency to the CentOS6 makefile
> f4876d5 FS-7031 [unimrcp] update
> sofia-sip.m4 so that it can build when
> relative path is used in
> configure.gnu ?with-sofia-sip
> [Jira:
> https://jira.freeswitch.org/browse/FS-7031]
> 061f3cb FS-7031 #resolve
> #comment [unimrcp] update library again to pull in
> upstream fix for
> ?with-sofia-sip=../sofia-sip
> [Jira:
> https://jira.freeswitch.org/browse/FS-7031]
> 382e683 Use FTDM_UINT64_FMT
> macro to log uint64_t values, in order to not
> break x86 builds.
> dc9e904
> FS-7025 fix compiler warning introduced from e55aee14 [Jira:
>
> https://jira.freeswitch.org/browse/FS-7025]
> b69c93e FS-7030 More work toward
> fixing FS build on Windows Visual Studio
> 2012 [Jira:
> https://jira.freeswitch.org/browse/FS-7030]
> db66cdb Fix mrcp libraries to
> build correctly
> c327455 FS-7030 More work toward getting FS to build on
> Windows Visual
> Studio 2012 [Jira:
> https://jira.freeswitch.org/browse/FS-7030]
> b341ff7 FS-7046: fix data types
> and casting on some vars to silence windows
> build warnings in mod_verto
> [Jira:
> https://jira.freeswitch.org/browse/FS-7046]
> 7ce5171 FS-7046 follow
> up on type change in mod_verto [Jira:
>
> https://jira.freeswitch.org/browse/FS-7046]
> 357ffad Fix windows build
> error
> 0b414a8 vs2010 unimrcp working build
> 0c1e698 Update build deps for
> debian list
> 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used
> error in
> mod_commands
> af6b23a FS-7046 Fix some additional Windows build
> warnings for mod_verto
> [Jira:
> https://jira.freeswitch.org/browse/FS-7046]
>
> Additional documentation:
>
>
> f63f868 FS-7049 ? Documentation for state optional paramenter in
>
> callcenter_config queue list and count [Jira:
>
> https://jira.freeswitch.org/browse/FS-7049]
>
> In terms of stability these
> were the use cases that were fixed:
>
> 392c687 FS-7055 Fix for a stability
> race condition in FS [Jira:
> https://jira.freeswitch.org/browse/FS-7055]
>
> d5119a7 FS-7091 Removed unnecessary mutex lock inside input component?s
>
> cleanup function since the input component won?t be cleaned up unless all>
> references have been released, in mod_rayo [Jira:
>
> https://jira.freeswitch.org/browse/FS-7091]
>
> These were the packaging
> improvements:
>
> 3c8dd3e Handle missing `lsb_release`
> 505cd29 Refactor
> distro detection and handling
> 430433a Improve error message
> d88bae1
> Support optional debian parallel builds
>
> The following bugs were
> squashed:
>
> 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia
> [Jira:
> https://jira.freeswitch.org/browse/FS-7015]
> 99a5b50 FS-7063 Fix for
> media delay issue [Jira:
> https://jira.freeswitch.org/browse/FS-7063]
>
> 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple
>
> uris, automatically add the q= header param in decending order in mod_sofia.
>
> [Jira: https://jira.freeswitch.org/browse/FS-7062]
> 5376e82 FS-6688 This will
> fix the normal case of record route from a proxy
> without breaking normal
> changing of a contact in mod_sofia [Jira:
>
> https://jira.freeswitch.org/browse/FS-6688]
> 06c241a FS-6891 FS-7002 FS-7059
> FS-7072 FS-7073 FS-7076 #close #comment All
> of these bugs are invalidated
> due to a botched revert [Jira:
> https://jira.freeswitch.org/browse/FS-6891]
>
> 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after
>
> changing it to work with ICE SDPs because it was looking in the wrong place
>
> for the 0.0.0.0 [Jira: https://jira.freeswitch.org/browse/FS-7015]
> 3d515cf
> Re-mark cur_payload as negotiated when detected as such by parser or
> the rtp
> could stop working on session re-invite
> 19272dc FS-7078 Fix
> sip_header_as_string to properly null_terminate on
> larger header strings
> [Jira: https://jira.freeswitch.org/browse/FS-7078]
> e268a72 FS-6994 Fix for
> Codec OPUS decoder error in mod_opus [Jira:
>
> https://jira.freeswitch.org/browse/FS-6994]
> 6dbb416 FS-7086 FS-6798 Fix for
> invalid codec tearing down the call request
> [Jira:
> https://jira.freeswitch.org/browse/FS-7086]
> 46adbec FS-7030 #comment
> [unimrcp] restore visual studio 2010/2012 project
> files added by FS project
> [Jira: https://jira.freeswitch.org/browse/FS-7030]
> bad5dc3 FS-7037 Fix for
> T38 fax break started by commit 5bbef7f1e50 [Jira:
>
> https://jira.freeswitch.org/browse/FS-7037]
> 72c3df5 FS-6891 FS-6713 #comment
> revert
> 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira:
>
> https://jira.freeswitch.org/browse/FS-6891]
> 2a7b022 FS-6980 #resolve don?t
> crash when using native recording on
> recordstop the redo [Jira:
> https://jira.freeswitch.org/browse/FS-6980]
> 35ba6a3 FS-6766 Fix verto caller
> ringback missing on conference bridge in
> mod_verto [Jira:
> https://jira.freeswitch.org/browse/FS-6766]
> e8cf9c7 FS-7045 Guarantee that
> dialed call can be joined when answered event
> is sent in mod_rayo [Jira:
> https://jira.freeswitch.org/browse/FS-7045]
> 4be6290 FS-7052 Moving jb queue
> swap operation out of the debug block.
> [Jira:
> https://jira.freeswitch.org/browse/FS-7052]
> 843e495 FS-7051 Preserve the
> annexb=no/yes status in mod_sangoma_codec
> [Jira:
> https://jira.freeswitch.org/browse/FS-7051]
> 158c1f2 FS-7002 Fix for recorded
> audio being choppy when diferent ptimes
> present and record session starts on
> bleg [Jira:
> https://jira.freeswitch.org/browse/FS-7002]
> 4ce2ce3 FS-7092
> Fixed bug with Comrex OPUS [Jira:
>
> https://jira.freeswitch.org/browse/FS-7092]
> d786490 Fix timestamps in
> mod_bert broken by the cpu improvements
> refactoring
> ba016c2 FS-7095 Fix
> for FS sending DTLS HELLO (and STUN binding request) to
> wrong port [Jira:
> https://jira.freeswitch.org/browse/FS-7095]
> e0dcd17 FS-7083 #comment patch
> to change mod_shout to use
> lame_encode_buffer_interleaved on stereo channels
> so we don?t have to mess
> with the input data [Jira:
> https://jira.freeswitch.org/browse/FS-7083]
> 326289c FS-7083 This patch adds
> a dedicated thread for writing to the file
> and the channel_variable
> RECORD_USE_THREAD=false will disable it and sync
> may still be maintained at
> the cost of dropping more data from the audio
> signal. [Jira:
> https://jira.freeswitch.org/browse/FS-7083]
> 9fabbab Disable hard-mute when a
> session has a media bug attached
> 0200bc1 FS-7083 Fix divide by zero [Jira:
>
> https://jira.freeswitch.org/browse/FS-7083]
> 067cb0f FS-7100 Make buffer for
> sub contact big enough in mod_sofia [Jira:
>
> https://jira.freeswitch.org/browse/FS-7100]
> 7798b2f FS-6984 Set default
> video rates [Jira:
> https://jira.freeswitch.org/browse/FS-6984]
> 763e6aa
> FS-7046 Fix warning introduced from b341ff7 [Jira:
>
> https://jira.freeswitch.org/browse/FS-7046]
> 65e678b FS-7070 Fix mod_expr
> `clamp` function typo
> 0a66db6 FS-7111 Fix for bridge_early_media crash
> [Jira:
> https://jira.freeswitch.org/browse/FS-7111]
>
> Miscellaneous
> commits:
>
> 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t
> work when
> using non-source build dir.
> [Jira:
> https://jira.freeswitch.org/browse/FS-7031]
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at 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://confluence.freeswitch.org
http://www.cl
> uecon.com

FreeSWITCH-users mailing
> list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman
> /listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt
> ions/freeswitch-users
http://www.freeswitch.org

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





From mike at jerris.com  Tue Mar  3 19:21:34 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Mar 2015 11:21:34 -0500
Subject: [Freeswitch-users] module dependency
In-Reply-To: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
Message-ID: <EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>

That is ALWAYS loaded before any other modules, so that not being loaded after.  Whats happening here, is the reload signal triggers the timezones to reload asynchronously.  This will require a code change to swap those out in some way that doesn't leave them empty for a short period, properly protected against race conditions.  This code is in switch_time.c.


> On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Please help me declare module dependency
> I has extended module radius_cdr by timezone support and from time to time is getting following error
> 
> freeswitch at internal> reload mod_radius_cdr
> +OK Reloading XML
> +OK module unloaded
> +OK module loaded
> 
> 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935 Stopping: mod_radius_cdr
> 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955 mod_radius_cdr unloaded.
> 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
> 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone 'Asia/Tokyo' not found!
> 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find timezone Asia/Tokyo
> , Setting timezone to GMT
> 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465 Successfully Loaded [mod_radius_cdr]
> 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded 1781 definitions
> 
> 
> Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.
> 
> How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded  before mod_radius_cdr?



From vladget at gmail.com  Tue Mar  3 20:50:03 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Tue, 3 Mar 2015 19:50:03 +0200
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
Message-ID: <CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>

Same traffic balanced between these two FS boxes and all CDRs from 1.4.5
came with PROXY, but all CDRs from 1.4.5 came with real codec were in
streams.



On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org> wrote:

> ZRTP hash in the sdp will cause it to toggle on too!
>
>
> On Saturday, February 28, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
> wrote:
>
>> Bote,
>> When I said identical configuration I mean files at FS configuration
>> directory.
>> G.729 license? No, I use proxy-media mode with no transcoding.
>>
>> Brian,
>> Both FS boxes configured for proxing media:
>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>> /sip_profiles/internal.xml
>> <param name="inbound-proxy-media" value="true"/>
>>
>> I do not understand why FS version 1.4.15 trying to hide actual
>> read/write codecs and change it by "PROXY"?
>>
>> Thank you.
>>
>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>
>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com>
>>> wrote:
>>>
>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>
>>>> I would say that these are not "absolutely" identical. As the FreeSWITCH
>>>> development team never sleeps it is likely that there are differences
>>>> in the
>>>> code that you now see. The first thing is to bring both machines up to
>>>> the
>>>> same release before comparing behaviors.
>>>>
>>>> Another suggestion is to confirm your G.729 license and configuration,
>>>> if
>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>> file(s)
>>>> in the correct locations and the other machine does not?
>>>>
>>>> Hope this helps.
>>>>
>>>> Bote
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Vladimir Getmanshchuk
>>>> Sent: Friday, 27 February, 2015 07:37
>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>> different FS
>>>> versions...
>>>>
>>>> Hello Everyone!
>>>>
>>>> I have two installations of FS with absolutely identical configurations.
>>>> Both has SIP profiles with proxy-media enabled.
>>>>
>>>> But on
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>
>>>> I have values in read_codec/write_codec variables at CDRs:
>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>
>>>> but on another one
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>
>>>> I have:
>>>> "read_codec":"G729","write_codec":"G729",
>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>
>>>> So Why?
>>>>
>>>>
>>>> --
>>>> Yours sincerely,
>>>> Vladimir Getmanshchuk
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Yours sincerely,
>> Vladimir Getmanshchuk
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/7d0be0af/attachment-0001.html 

From vladget at gmail.com  Tue Mar  3 20:52:47 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Tue, 3 Mar 2015 19:52:47 +0200
Subject: [Freeswitch-users] inheritance parameters from sofia.conf.xml
	to sip_profiles
In-Reply-To: <CAFiqYu=BGKG2z7PCmEm_i0eyUoEyPyf7kOyEre=Tcn6K5-33WQ@mail.gmail.com>
References: <CAMxcU5+2s2TOyP1_qRuBJs2fi1m1j6V=ysDCR6mPFM_zZaA2Dg@mail.gmail.com>
	<CAFiqYu=BGKG2z7PCmEm_i0eyUoEyPyf7kOyEre=Tcn6K5-33WQ@mail.gmail.com>
Message-ID: <CAMxcU5JVMnWjsn510xkCBRmy2k36usUqr=DO9BUmx2ZemkpRhA@mail.gmail.com>

Oh!

Thanks!

On Sun, Mar 1, 2015 at 11:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Profiles do not inherit parameters from global_settings.
>
> The valid parameters for global_settings are:
> log-level
> tracelevel
> debug-presence
> debug-sla
> max-reg-threads
> auto-restart
> rewrite-multicasted-fs-path
> capture-server
>
> On 1 March 2015 at 20:37, Vladimir Getmanshchuk <vladget at gmail.com> wrote:
>
>> Hi Everyone!
>>
>> Looks like inheritance of some parameters from sofia.conf.xml
>> <global_settings> to sip_profiles does not work.
>>
>> I've faced to problem with next parameters which configured at
>> <global_settings> in sofia.conf.xml:
>> - rtp-autofix-timing
>> - user-agent-string
>> but has no effect.
>>
>> Please advice.
>>
>> --
>> Yours sincerely,
>> Vladimir Getmanshchuk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/30f8406f/attachment.html 

From brian at freeswitch.org  Tue Mar  3 20:54:41 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 12:54:41 -0500
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
	<CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
Message-ID: <CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>

If you enable proxy media it will say proxy too....

On Tuesday, March 3, 2015, Vladimir Getmanshchuk <vladget at gmail.com> wrote:

> Same traffic balanced between these two FS boxes and all CDRs from 1.4.5
> came with PROXY, but all CDRs from 1.4.5 came with real codec were in
> streams.
>
>
>
> On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>
>> ZRTP hash in the sdp will cause it to toggle on too!
>>
>>
>> On Saturday, February 28, 2015, Vladimir Getmanshchuk <vladget at gmail.com
>> <javascript:_e(%7B%7D,'cvml','vladget at gmail.com');>> wrote:
>>
>>> Bote,
>>> When I said identical configuration I mean files at FS configuration
>>> directory.
>>> G.729 license? No, I use proxy-media mode with no transcoding.
>>>
>>> Brian,
>>> Both FS boxes configured for proxing media:
>>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>>> /sip_profiles/internal.xml
>>> <param name="inbound-proxy-media" value="true"/>
>>>
>>> I do not understand why FS version 1.4.15 trying to hide actual
>>> read/write codecs and change it by "PROXY"?
>>>
>>> Thank you.
>>>
>>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>>
>>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com>
>>>> wrote:
>>>>
>>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>
>>>>> I would say that these are not "absolutely" identical. As the
>>>>> FreeSWITCH
>>>>> development team never sleeps it is likely that there are differences
>>>>> in the
>>>>> code that you now see. The first thing is to bring both machines up to
>>>>> the
>>>>> same release before comparing behaviors.
>>>>>
>>>>> Another suggestion is to confirm your G.729 license and configuration,
>>>>> if
>>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>>> file(s)
>>>>> in the correct locations and the other machine does not?
>>>>>
>>>>> Hope this helps.
>>>>>
>>>>> Bote
>>>>>
>>>>>
>>>>> -----Original Message-----
>>>>> From: Vladimir Getmanshchuk
>>>>> Sent: Friday, 27 February, 2015 07:37
>>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>>> different FS
>>>>> versions...
>>>>>
>>>>> Hello Everyone!
>>>>>
>>>>> I have two installations of FS with absolutely identical
>>>>> configurations.
>>>>> Both has SIP profiles with proxy-media enabled.
>>>>>
>>>>> But on
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>
>>>>> I have values in read_codec/write_codec variables at CDRs:
>>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>>
>>>>> but on another one
>>>>> freeswitch at internal> version
>>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>
>>>>> I have:
>>>>> "read_codec":"G729","write_codec":"G729",
>>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>>
>>>>> So Why?
>>>>>
>>>>>
>>>>> --
>>>>> Yours sincerely,
>>>>> Vladimir Getmanshchuk
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Yours sincerely,
>>> Vladimir Getmanshchuk
>>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/3d55d457/attachment.html 

From s.safarov at gmail.com  Tue Mar  3 21:10:22 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 3 Mar 2015 21:10:22 +0300
Subject: [Freeswitch-users] module dependency
In-Reply-To: <EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
Message-ID: <CAHtxdDf=XBd68GHX7+ndabMGofqz+5uhfpTsmMYHWUT+4hiAzA@mail.gmail.com>

Do I understand correctly that is required rewrite the function
switch_load_timezones?

On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com> wrote:

> That is ALWAYS loaded before any other modules, so that not being loaded
> after.  Whats happening here, is the reload signal triggers the timezones
> to reload asynchronously.  This will require a code change to swap those
> out in some way that doesn't leave them empty for a short period, properly
> protected against race conditions.  This code is in switch_time.c.
>
>
> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> >
> > Please help me declare module dependency
> > I has extended module radius_cdr by timezone support and from time to
> time is getting following error
> >
> > freeswitch at internal> reload mod_radius_cdr
> > +OK Reloading XML
> > +OK module unloaded
> > +OK module loaded
> >
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935
> Stopping: mod_radius_cdr
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955
> mod_radius_cdr unloaded.
> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone
> 'Asia/Tokyo' not found!
> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find
> timezone Asia/Tokyo
> > , Setting timezone to GMT
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465
> Successfully Loaded [mod_radius_cdr]
> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded
> 1781 definitions
> >
> >
> > Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE
> but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.
> >
> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded
> before mod_radius_cdr?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/4b02b783/attachment-0001.html 

From jpablolorenzetti at hotmail.com  Tue Mar  3 21:26:30 2015
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Tue, 3 Mar 2015 12:26:30 -0600
Subject: [Freeswitch-users] Channels in the same session
Message-ID: <DUB405-EAS369B417B439EF3CDCB44249D9110@phx.gbl>

Hi, i m writing a module that needs to check for certain information in a database for the caller and the destination number, for this the module is subscribing to the CS_INIT channel events, so everytime a channel is created the module callback is called and it checks the numbers, the problem is that the callback gets called twice, for the creation of the a-leg of the call and the creation of the b-leg. Is there any way to accomplish what i m trying to do ? I have already try getting testing for the flags in the channel but it did not work, i might be doing it wrong maybe ? Thanks!

From s.safarov at gmail.com  Tue Mar  3 21:35:55 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 3 Mar 2015 21:35:55 +0300
Subject: [Freeswitch-users] module dependency
In-Reply-To: <EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
Message-ID: <CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>

Will it help addition of the configuration update flag of module
CORE_SOFTTIMER_MODULE.
And to add idle loop 'into the function switch_lookup_timezone until
'update is complete?

On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com> wrote:

> That is ALWAYS loaded before any other modules, so that not being loaded
> after.  Whats happening here, is the reload signal triggers the timezones
> to reload asynchronously.  This will require a code change to swap those
> out in some way that doesn't leave them empty for a short period, properly
> protected against race conditions.  This code is in switch_time.c.
>
>
> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> >
> > Please help me declare module dependency
> > I has extended module radius_cdr by timezone support and from time to
> time is getting following error
> >
> > freeswitch at internal> reload mod_radius_cdr
> > +OK Reloading XML
> > +OK module unloaded
> > +OK module loaded
> >
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935
> Stopping: mod_radius_cdr
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955
> mod_radius_cdr unloaded.
> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone
> 'Asia/Tokyo' not found!
> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find
> timezone Asia/Tokyo
> > , Setting timezone to GMT
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465
> Successfully Loaded [mod_radius_cdr]
> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded
> 1781 definitions
> >
> >
> > Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE
> but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.
> >
> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded
> before mod_radius_cdr?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/e95a9e82/attachment.html 

From blasterjr at gmail.com  Tue Mar  3 21:38:04 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 3 Mar 2015 11:38:04 -0700
Subject: [Freeswitch-users] Fwd: How it works? Lua & MySQL throug ODBC
In-Reply-To: <CAL2h29jfwPVHH73DeKeV4dqFgiXF4hgGiY4NoZUsqA1FJ4vbcQ@mail.gmail.com>
References: <CAL2h29i683d+Cbh5t2UrfSeKwKgWmrRXK0+xo43Pe37T8GppMQ@mail.gmail.com>
	<CAL2h29jfwPVHH73DeKeV4dqFgiXF4hgGiY4NoZUsqA1FJ4vbcQ@mail.gmail.com>
Message-ID: <CAO3hyzG1gNXSD4igknRHbMsvsfx_yGBA=vCPdKjHBct+679=bw@mail.gmail.com>

Make sure you are using UnixODBC version 2.3.X when dealing with MySQL.

I have had lots of problems with CORE dumps caused by ODBC + MySQL (might
not just be MySQL).

If you could elaborate on your Operating System (and version) that might
allow someone to help you out further.

On Mon, Mar 2, 2015 at 2:09 PM, ???????, ??????? / Dmitriy Borisov <
bordmi at rarus.ru> wrote:

> Hi, All!
>
> I have experienced periodicaly problems with FreeSWITCH running LUA
> scripts.
>
> This scripts are event hooks. In some unknown reasons some times
> FreeSWITCH crashes without any records in log. I`ve some qustions:
>
> 1. How can I enable more detailed debug?
> 2. How to store freeswitch.core in some explained previously place?
> 3. May be thread blocking while doing transaction to MySQL through ODBC
> the source of my problems? If yes, how to solve it?
>
> --
> with best regards,
> Dmitriy Borisov
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/881601c1/attachment.html 

From victor.chukalovskiy at gmail.com  Tue Mar  3 21:41:11 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Tue, 03 Mar 2015 13:41:11 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.4.15 Released
In-Reply-To: <D11B33EF.95DA7%krice@freeswitch.org>
References: <D11B33EF.95DA7%krice@freeswitch.org>
Message-ID: <54F60047.7020307@gmail.com>

Great, thanks! Will watch for an announcement then



On 15-03-03 10:48 AM, Ken Rice wrote:
> We're In the process of re-arranging things... Watch for an announcement
> soon
>
>
> On 3/3/15, 9:43 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> All hail our Release Overlord!
>
>
>
> On Tue, Mar 3, 2015 at 4:36 PM, Victor
>> Chukalovskiy
> <victor.chukalovskiy at gmail.com> wrote:
>> Good day,
>>
>> Am I on a
>> wrong branch, or FS stable release did not have any commits since
>> the New
>> Year?
>> When doing "make current" I'm still on this version:
>>
>> Version
>> 1.4.15 git 507a0f2 2014-12-29
>>
>> Thanks!
>> -Victor
>>
>> On 14-12-30 02:26 PM,
>> Ken Rice wrote:
>>
>> New Post on freeswitch.org from krice387
>> check it out at
>> http://freeswitch.org/freeswitch-1-4-15-released/
>> FreeSWITCH 1.4.15
>> Released
>>
>> FreeSWITCH 1.4.14 has been released!
>>
>> This is routine
>> maintenance release.
>>
>> Source Tarball available at
>>
>> http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2
>>
>> Debian and Yum Repos
>> have been updated as well.
>>
>> See the release notes below for a list of
>> notable changes.
>>
>> Happy New Years From the FreeSWITCH Team!
>>
>> Release
>> Notes:
>>
>> New features that were added:
>>
>> e55aee1 FS-7025 Add
>> drop_dtmf_masking_tone channel_variable [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7025]
>> a8c5a0c FS-7048 Add timezone
>> support to mod_say_{de,es,ja,nl,th,zh}
>> 17574a8 Add bert stats to
>> mod_bert::lost_sync event
>> a26e29c vs2010 support for recent unimrcp
>> changes
>> cee8b30 Set rtp_has_crypto for dtls calls
>> 5fcff50 FS-7093 Create
>> uuid_drop_dtmf [Jira:
>> https://jira.freeswitch.org/browse/FS-7093]
>> f024ea3
>> FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input
>> components
>> in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7047]
>> e783999 Some
>> changes to webrtc to make it work with iDoubs in rtcweb profile
>> mode
>>
>> d189e98 Allow 10ms jb
>>
>> 750b1dd FS-7114 Allow streaming binary data from
>> mod_memcache
>>
>> Improvements in performance:
>>
>> 4bcf1d8 Use cached time to
>> save cpu
>>
>> Improvements in cross platform build supports:
>>
>> 32c27b3 Added
>> a Debian dependency to the CentOS6 makefile
>> f4876d5 FS-7031 [unimrcp] update
>> sofia-sip.m4 so that it can build when
>> relative path is used in
>> configure.gnu ?with-sofia-sip
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7031]
>> 061f3cb FS-7031 #resolve
>> #comment [unimrcp] update library again to pull in
>> upstream fix for
>> ?with-sofia-sip=../sofia-sip
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7031]
>> 382e683 Use FTDM_UINT64_FMT
>> macro to log uint64_t values, in order to not
>> break x86 builds.
>> dc9e904
>> FS-7025 fix compiler warning introduced from e55aee14 [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7025]
>> b69c93e FS-7030 More work toward
>> fixing FS build on Windows Visual Studio
>> 2012 [Jira:
>> https://jira.freeswitch.org/browse/FS-7030]
>> db66cdb Fix mrcp libraries to
>> build correctly
>> c327455 FS-7030 More work toward getting FS to build on
>> Windows Visual
>> Studio 2012 [Jira:
>> https://jira.freeswitch.org/browse/FS-7030]
>> b341ff7 FS-7046: fix data types
>> and casting on some vars to silence windows
>> build warnings in mod_verto
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7046]
>> 7ce5171 FS-7046 follow
>> up on type change in mod_verto [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7046]
>> 357ffad Fix windows build
>> error
>> 0b414a8 vs2010 unimrcp working build
>> 0c1e698 Update build deps for
>> debian list
>> 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used
>> error in
>> mod_commands
>> af6b23a FS-7046 Fix some additional Windows build
>> warnings for mod_verto
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7046]
>>
>> Additional documentation:
>>
>>
>> f63f868 FS-7049 ? Documentation for state optional paramenter in
>>
>> callcenter_config queue list and count [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7049]
>>
>> In terms of stability these
>> were the use cases that were fixed:
>>
>> 392c687 FS-7055 Fix for a stability
>> race condition in FS [Jira:
>> https://jira.freeswitch.org/browse/FS-7055]
>>
>> d5119a7 FS-7091 Removed unnecessary mutex lock inside input component?s
>>
>> cleanup function since the input component won?t be cleaned up unless all>
>> references have been released, in mod_rayo [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7091]
>>
>> These were the packaging
>> improvements:
>>
>> 3c8dd3e Handle missing `lsb_release`
>> 505cd29 Refactor
>> distro detection and handling
>> 430433a Improve error message
>> d88bae1
>> Support optional debian parallel builds
>>
>> The following bugs were
>> squashed:
>>
>> 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7015]
>> 99a5b50 FS-7063 Fix for
>> media delay issue [Jira:
>> https://jira.freeswitch.org/browse/FS-7063]
>>
>> 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple
>>
>> uris, automatically add the q= header param in decending order in mod_sofia.
>>
>> [Jira: https://jira.freeswitch.org/browse/FS-7062]
>> 5376e82 FS-6688 This will
>> fix the normal case of record route from a proxy
>> without breaking normal
>> changing of a contact in mod_sofia [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-6688]
>> 06c241a FS-6891 FS-7002 FS-7059
>> FS-7072 FS-7073 FS-7076 #close #comment All
>> of these bugs are invalidated
>> due to a botched revert [Jira:
>> https://jira.freeswitch.org/browse/FS-6891]
>>
>> 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after
>>
>> changing it to work with ICE SDPs because it was looking in the wrong place
>>
>> for the 0.0.0.0 [Jira: https://jira.freeswitch.org/browse/FS-7015]
>> 3d515cf
>> Re-mark cur_payload as negotiated when detected as such by parser or
>> the rtp
>> could stop working on session re-invite
>> 19272dc FS-7078 Fix
>> sip_header_as_string to properly null_terminate on
>> larger header strings
>> [Jira: https://jira.freeswitch.org/browse/FS-7078]
>> e268a72 FS-6994 Fix for
>> Codec OPUS decoder error in mod_opus [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-6994]
>> 6dbb416 FS-7086 FS-6798 Fix for
>> invalid codec tearing down the call request
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7086]
>> 46adbec FS-7030 #comment
>> [unimrcp] restore visual studio 2010/2012 project
>> files added by FS project
>> [Jira: https://jira.freeswitch.org/browse/FS-7030]
>> bad5dc3 FS-7037 Fix for
>> T38 fax break started by commit 5bbef7f1e50 [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7037]
>> 72c3df5 FS-6891 FS-6713 #comment
>> revert
>> 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-6891]
>> 2a7b022 FS-6980 #resolve don?t
>> crash when using native recording on
>> recordstop the redo [Jira:
>> https://jira.freeswitch.org/browse/FS-6980]
>> 35ba6a3 FS-6766 Fix verto caller
>> ringback missing on conference bridge in
>> mod_verto [Jira:
>> https://jira.freeswitch.org/browse/FS-6766]
>> e8cf9c7 FS-7045 Guarantee that
>> dialed call can be joined when answered event
>> is sent in mod_rayo [Jira:
>> https://jira.freeswitch.org/browse/FS-7045]
>> 4be6290 FS-7052 Moving jb queue
>> swap operation out of the debug block.
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7052]
>> 843e495 FS-7051 Preserve the
>> annexb=no/yes status in mod_sangoma_codec
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7051]
>> 158c1f2 FS-7002 Fix for recorded
>> audio being choppy when diferent ptimes
>> present and record session starts on
>> bleg [Jira:
>> https://jira.freeswitch.org/browse/FS-7002]
>> 4ce2ce3 FS-7092
>> Fixed bug with Comrex OPUS [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7092]
>> d786490 Fix timestamps in
>> mod_bert broken by the cpu improvements
>> refactoring
>> ba016c2 FS-7095 Fix
>> for FS sending DTLS HELLO (and STUN binding request) to
>> wrong port [Jira:
>> https://jira.freeswitch.org/browse/FS-7095]
>> e0dcd17 FS-7083 #comment patch
>> to change mod_shout to use
>> lame_encode_buffer_interleaved on stereo channels
>> so we don?t have to mess
>> with the input data [Jira:
>> https://jira.freeswitch.org/browse/FS-7083]
>> 326289c FS-7083 This patch adds
>> a dedicated thread for writing to the file
>> and the channel_variable
>> RECORD_USE_THREAD=false will disable it and sync
>> may still be maintained at
>> the cost of dropping more data from the audio
>> signal. [Jira:
>> https://jira.freeswitch.org/browse/FS-7083]
>> 9fabbab Disable hard-mute when a
>> session has a media bug attached
>> 0200bc1 FS-7083 Fix divide by zero [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7083]
>> 067cb0f FS-7100 Make buffer for
>> sub contact big enough in mod_sofia [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7100]
>> 7798b2f FS-6984 Set default
>> video rates [Jira:
>> https://jira.freeswitch.org/browse/FS-6984]
>> 763e6aa
>> FS-7046 Fix warning introduced from b341ff7 [Jira:
>>
>> https://jira.freeswitch.org/browse/FS-7046]
>> 65e678b FS-7070 Fix mod_expr
>> `clamp` function typo
>> 0a66db6 FS-7111 Fix for bridge_early_media crash
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7111]
>>
>> Miscellaneous
>> commits:
>>
>> 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t
>> work when
>> using non-source build dir.
>> [Jira:
>> https://jira.freeswitch.org/browse/FS-7031]
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________>
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>>
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>>
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________>
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>>
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>>
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>
>



From mike at jerris.com  Tue Mar  3 22:20:50 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Mar 2015 14:20:50 -0500
Subject: [Freeswitch-users] module dependency
In-Reply-To: <CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
	<CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>
Message-ID: <0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>

yes it will require code changes there.  I wouldn't make an idle loop tho.  I would do something to swap out the pointers with the new ones and protect it all with a mutex.  I think we do something similar with dialplan reload.


> On Mar 3, 2015, at 1:35 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Will it help addition of the configuration update flag of module CORE_SOFTTIMER_MODULE.
> And to add idle loop 'into the function switch_lookup_timezone until 'update is complete?
> 
> On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> That is ALWAYS loaded before any other modules, so that not being loaded after.  Whats happening here, is the reload signal triggers the timezones to reload asynchronously.  This will require a code change to swap those out in some way that doesn't leave them empty for a short period, properly protected against race conditions.  This code is in switch_time.c.
> 
> 
> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> >
> > Please help me declare module dependency
> > I has extended module radius_cdr by timezone support and from time to time is getting following error
> >
> > freeswitch at internal> reload mod_radius_cdr
> > +OK Reloading XML
> > +OK module unloaded
> > +OK module loaded
> >
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935 Stopping: mod_radius_cdr
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955 mod_radius_cdr unloaded.
> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone 'Asia/Tokyo' not found!
> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find timezone Asia/Tokyo
> > , Setting timezone to GMT
> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465 Successfully Loaded [mod_radius_cdr]
> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded 1781 definitions
> >
> >
> > Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.
> >
> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded  before mod_radius_cdr?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/2239655b/attachment.html 

From tfred31 at yahoo.com  Tue Mar  3 23:49:13 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 3 Mar 2015 12:49:13 -0800
Subject: [Freeswitch-users] Getting FS to place Outbound calls
Message-ID: <1425415753.87368.YahooMailBasic@web160205.mail.bf1.yahoo.com>

Your FreeSWITCH note came to me in 'digest' mode with other postings.
So I think that this Reply will likely start a new thread when I hoped that it would not.
So my apologies if this doesn't work as desired.

Regardless, it seems like whenever I find a problem and report, I subsequently find a more basic problem which likely results in that which I reported.

================================================

Anyway, I still cannot get my Outbound calls to work.
But at a more basic level, I find that when I do:   sofia status    I see that my gateway ( velocity-outbound ) is being reported as  NOREG   which I assume to be Not Registered and therefore Will Not Work.

                                   Name          Type                                       Data      State
================================================================================
                            external-ipv6       profile   sip:mod_sofia@[2002:6b01:26bd::6b01:26bd]:5080        RUNNING (0)
        external-ipv6::example.com     gateway                    sip:joeuser at example.com           NOREG
external-ipv6::velocity-outbound        gateway               sip:FreeSWITCH@<SIP IP Here>      NOREG
                         <FS IP Here>        alias                                   internal                             ALIASED
                                   external      profile            sip:mod_sofia@<FS IP Here>:5080           RUNNING (0)
             external::example.com       gateway                    sip:joeuser at example.com           NOREG
         external::velocity-outbound      gateway               sip:FreeSWITCH@<SIP IP Here>      NOREG
                             internal-ipv6       profile   sip:mod_sofia@[2002:6b01:26bd::6b01:26bd]:5060        RUNNING (0)
                                   internal       profile            sip:mod_sofia@<FS IP Here>:5060           RUNNING (0)
================================================================================

I contacted my SIP line vendor and ran a test.
It seems that in spite of my using   external-ipv6.xml   they saw the INVITE come in from my FreeSWITCH and they responded with a 200 OK
But they never saw anything else come back to them after that with which to complete the 'handshake'

Interesting enough within    external-ipv6.xml    I modified it so that it would  <gateways>  would include nothing
And within   external.xml   I added the  <gateways>  to include:
     <X-PRE-PROCESS cmd="include" data="external/outbound_via_velocity.xml"/>
After which I did a full Restart on the FreeSWITCH service

And in spite of that I still see in the freeswitch.log:
    Added gateway 'velocity-outbound' to profile 'external-ipv6'

This is a total guess, but I assume that I need to see the    velocity-outbound   gateway that I am trying to use show up as a  REG  before I can go any further.

If that is correct, how/where would I look to find out what is wrong?

BTW:   In the conf\dialplan\default    directory  I did set the other plan's extensions to  .txt  so that they would be ignored.

Thanks
TF

 
 
   


From mthakershi at gmail.com  Wed Mar  4 00:12:48 2015
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 3 Mar 2015 15:12:48 -0600
Subject: [Freeswitch-users] Load test new install
Message-ID: <CAKHK4ZMV+fKp3u9q9NAXO7cPYW4+Nt=R30rmJ8U5LbnBCYA9gA@mail.gmail.com>

I just installed 64-bit FreeSwitch and want to move my 32-bit base to the
new server.

How can I do basic load test? How can I make dummy calls between two
servers?

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/8f5710de/attachment.html 

From vladget at gmail.com  Wed Mar  4 00:30:43 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Tue, 3 Mar 2015 23:30:43 +0200
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
	<CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
	<CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>
Message-ID: <CAMxcU5Jk2=ROwCUNPzjmGHRRikCWdiX-=zQ--NrRhpTDj11erg@mail.gmail.com>

As I have said before configs on both FS boxes are same. So Proxy-Media
enabled on both FS boxes.

On Tue, Mar 3, 2015 at 7:54 PM, Brian West <brian at freeswitch.org> wrote:

> If you enable proxy media it will say proxy too....
>
>
> On Tuesday, March 3, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
> wrote:
>
>> Same traffic balanced between these two FS boxes and all CDRs from 1.4.5
>> came with PROXY, but all CDRs from 1.4.5 came with real codec were in
>> streams.
>>
>>
>>
>> On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> ZRTP hash in the sdp will cause it to toggle on too!
>>>
>>>
>>> On Saturday, February 28, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
>>> wrote:
>>>
>>>> Bote,
>>>> When I said identical configuration I mean files at FS configuration
>>>> directory.
>>>> G.729 license? No, I use proxy-media mode with no transcoding.
>>>>
>>>> Brian,
>>>> Both FS boxes configured for proxing media:
>>>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>>>> /sip_profiles/internal.xml
>>>> <param name="inbound-proxy-media" value="true"/>
>>>>
>>>> I do not understand why FS version 1.4.15 trying to hide actual
>>>> read/write codecs and change it by "PROXY"?
>>>>
>>>> Thank you.
>>>>
>>>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>>>
>>>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com>
>>>>> wrote:
>>>>>
>>>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>
>>>>>> I would say that these are not "absolutely" identical. As the
>>>>>> FreeSWITCH
>>>>>> development team never sleeps it is likely that there are differences
>>>>>> in the
>>>>>> code that you now see. The first thing is to bring both machines up
>>>>>> to the
>>>>>> same release before comparing behaviors.
>>>>>>
>>>>>> Another suggestion is to confirm your G.729 license and
>>>>>> configuration, if
>>>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>>>> file(s)
>>>>>> in the correct locations and the other machine does not?
>>>>>>
>>>>>> Hope this helps.
>>>>>>
>>>>>> Bote
>>>>>>
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Vladimir Getmanshchuk
>>>>>> Sent: Friday, 27 February, 2015 07:37
>>>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>>>> different FS
>>>>>> versions...
>>>>>>
>>>>>> Hello Everyone!
>>>>>>
>>>>>> I have two installations of FS with absolutely identical
>>>>>> configurations.
>>>>>> Both has SIP profiles with proxy-media enabled.
>>>>>>
>>>>>> But on
>>>>>> freeswitch at internal> version
>>>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>
>>>>>> I have values in read_codec/write_codec variables at CDRs:
>>>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>>>
>>>>>> but on another one
>>>>>> freeswitch at internal> version
>>>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>
>>>>>> I have:
>>>>>> "read_codec":"G729","write_codec":"G729",
>>>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>>>
>>>>>> So Why?
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Yours sincerely,
>>>>>> Vladimir Getmanshchuk
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Yours sincerely,
>>>> Vladimir Getmanshchuk
>>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Yours sincerely,
>> Vladimir Getmanshchuk
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/1255fb39/attachment-0001.html 

From s.safarov at gmail.com  Wed Mar  4 00:56:50 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 4 Mar 2015 00:56:50 +0300
Subject: [Freeswitch-users] module dependency
In-Reply-To: <0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
	<CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>
	<0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
Message-ID: <CAHtxdDfWT_A1JCVC3EuGnuCiMhzJkr31A82ODYN05mNsBdJQtw@mail.gmail.com>

Do I need to make a request in jira?

On Tue, Mar 3, 2015 at 10:20 PM, Michael Jerris <mike at jerris.com> wrote:

> yes it will require code changes there.  I wouldn't make an idle loop
> tho.  I would do something to swap out the pointers with the new ones and
> protect it all with a mutex.  I think we do something similar with dialplan
> reload.
>
>
> On Mar 3, 2015, at 1:35 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> Will it help addition of the configuration update flag of module
> CORE_SOFTTIMER_MODULE.
> And to add idle loop 'into the function switch_lookup_timezone until
> 'update is complete?
>
> On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> That is ALWAYS loaded before any other modules, so that not being loaded
>> after.  Whats happening here, is the reload signal triggers the timezones
>> to reload asynchronously.  This will require a code change to swap those
>> out in some way that doesn't leave them empty for a short period, properly
>> protected against race conditions.  This code is in switch_time.c.
>>
>>
>> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>> >
>> > Please help me declare module dependency
>> > I has extended module radius_cdr by timezone support and from time to
>> time is getting following error
>> >
>> > freeswitch at internal> reload mod_radius_cdr
>> > +OK Reloading XML
>> > +OK module unloaded
>> > +OK module loaded
>> >
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935
>> Stopping: mod_radius_cdr
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955
>> mod_radius_cdr unloaded.
>> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
>> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone
>> 'Asia/Tokyo' not found!
>> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find
>> timezone Asia/Tokyo
>> > , Setting timezone to GMT
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465
>> Successfully Loaded [mod_radius_cdr]
>> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded
>> 1781 definitions
>> >
>> >
>> > Module currently depend of loaded configuradion of
>> CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded
>> CORE_SOFTTIMER_MODULE configuration.
>> >
>> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded
>> before mod_radius_cdr?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/3c7d07e5/attachment.html 

From brian at freeswitch.org  Wed Mar  4 01:00:06 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2015 17:00:06 -0500
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAMxcU5Jk2=ROwCUNPzjmGHRRikCWdiX-=zQ--NrRhpTDj11erg@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
	<CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
	<CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>
	<CAMxcU5Jk2=ROwCUNPzjmGHRRikCWdiX-=zQ--NrRhpTDj11erg@mail.gmail.com>
Message-ID: <CAEJMVkD=oy1FRLB5mKYBGe4S3L5eAFSFEP-9_4eZmPj80q3wBQ@mail.gmail.com>

Without looking a the logs I can only guess that something is triggering
it...happen to have a debug log of this?

On Tue, Mar 3, 2015 at 4:30 PM, Vladimir Getmanshchuk <vladget at gmail.com>
wrote:

> As I have said before configs on both FS boxes are same. So Proxy-Media
> enabled on both FS boxes.
>
> On Tue, Mar 3, 2015 at 7:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> If you enable proxy media it will say proxy too....
>>
>>
>> On Tuesday, March 3, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
>> wrote:
>>
>>> Same traffic balanced between these two FS boxes and all CDRs from 1.4.5
>>> came with PROXY, but all CDRs from 1.4.5 came with real codec were in
>>> streams.
>>>
>>>
>>>
>>> On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> ZRTP hash in the sdp will cause it to toggle on too!
>>>>
>>>>
>>>> On Saturday, February 28, 2015, Vladimir Getmanshchuk <
>>>> vladget at gmail.com> wrote:
>>>>
>>>>> Bote,
>>>>> When I said identical configuration I mean files at FS configuration
>>>>> directory.
>>>>> G.729 license? No, I use proxy-media mode with no transcoding.
>>>>>
>>>>> Brian,
>>>>> Both FS boxes configured for proxing media:
>>>>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>>>>> /sip_profiles/internal.xml
>>>>> <param name="inbound-proxy-media" value="true"/>
>>>>>
>>>>> I do not understand why FS version 1.4.15 trying to hide actual
>>>>> read/write codecs and change it by "PROXY"?
>>>>>
>>>>> Thank you.
>>>>>
>>>>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>>>>
>>>>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com>
>>>>>> wrote:
>>>>>>
>>>>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>
>>>>>>> I would say that these are not "absolutely" identical. As the
>>>>>>> FreeSWITCH
>>>>>>> development team never sleeps it is likely that there are
>>>>>>> differences in the
>>>>>>> code that you now see. The first thing is to bring both machines up
>>>>>>> to the
>>>>>>> same release before comparing behaviors.
>>>>>>>
>>>>>>> Another suggestion is to confirm your G.729 license and
>>>>>>> configuration, if
>>>>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>>>>> file(s)
>>>>>>> in the correct locations and the other machine does not?
>>>>>>>
>>>>>>> Hope this helps.
>>>>>>>
>>>>>>> Bote
>>>>>>>
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Vladimir Getmanshchuk
>>>>>>> Sent: Friday, 27 February, 2015 07:37
>>>>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>>>>> different FS
>>>>>>> versions...
>>>>>>>
>>>>>>> Hello Everyone!
>>>>>>>
>>>>>>> I have two installations of FS with absolutely identical
>>>>>>> configurations.
>>>>>>> Both has SIP profiles with proxy-media enabled.
>>>>>>>
>>>>>>> But on
>>>>>>> freeswitch at internal> version
>>>>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>>
>>>>>>> I have values in read_codec/write_codec variables at CDRs:
>>>>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>>>>
>>>>>>> but on another one
>>>>>>> freeswitch at internal> version
>>>>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>
>>>>>>> I have:
>>>>>>> "read_codec":"G729","write_codec":"G729",
>>>>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>>>>
>>>>>>> So Why?
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Yours sincerely,
>>>>>>> Vladimir Getmanshchuk
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Brian West*
>>>>>> brian at freeswitch.org
>>>>>>
>>>>>>
>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>> http://www.freeswitchbook.com
>>>>>> http://www.freeswitchcookbook.com
>>>>>>
>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Yours sincerely,
>>>>> Vladimir Getmanshchuk
>>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Yours sincerely,
>>> Vladimir Getmanshchuk
>>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/99cbb37e/attachment-0001.html 

From ssinyagin at gmail.com  Wed Mar  4 01:17:01 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 3 Mar 2015 23:17:01 +0100
Subject: [Freeswitch-users] Load test new install
In-Reply-To: <CAKHK4ZMV+fKp3u9q9NAXO7cPYW4+Nt=R30rmJ8U5LbnBCYA9gA@mail.gmail.com>
References: <CAKHK4ZMV+fKp3u9q9NAXO7cPYW4+Nt=R30rmJ8U5LbnBCYA9gA@mail.gmail.com>
Message-ID: <CACOYK=gqD80b4C9rSB5nPczLYjkaNVT+J95oms3gXqnxcHDbkQ@mail.gmail.com>

here I described some load tests:
https://txlab.wordpress.com/2014/04/18/freeswitch-performance-test-on-pc-engines-apu/
https://txlab.wordpress.com/2014/05/07/simple-performance-test-for-freeswitch-conferencing/



On Tue, Mar 3, 2015 at 10:12 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I just installed 64-bit FreeSwitch and want to move my 32-bit base to the
> new server.
>
> How can I do basic load test? How can I make dummy calls between two
> servers?
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Mar  4 01:21:04 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 3 Mar 2015 22:21:04 +0000
Subject: [Freeswitch-users] $${sounds_dir} variable
Message-ID: <CAK+KT39=gngRa6D+JoRhOPPwq8+ce0X6jKqvQ5POZPDwKaoEOw@mail.gmail.com>

Hi all,
Could someone please let me know which file contains the defined location
for the variable $${sounds_dir}

Thanks

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150303/63b4aa2a/attachment.html 

From pkelly at gmail.com  Wed Mar  4 11:31:56 2015
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 4 Mar 2015 08:31:56 +0000
Subject: [Freeswitch-users] ICE SDP Params
In-Reply-To: <CAEJMVkAG+Jn07upn3JQLyhs7si7BfCb8fn0DybOv9Ce28cTUHQ@mail.gmail.com>
References: <DB3PR05MB0286C6B3C5BC977C6BFB48383110@DB3PR05MB028.eurprd05.prod.outlook.com>
	<CAEJMVkAG+Jn07upn3JQLyhs7si7BfCb8fn0DybOv9Ce28cTUHQ@mail.gmail.com>
Message-ID: <CAEWH9M9_s9x0_H9Jp5KP7yhwVReVmndkpMpkhctKFMmkS=CePQ@mail.gmail.com>

Hi Brian

I have been working with Rob on the trace in question - the initial INVITE
into FreeSWITCH has no SDP, so FreeSWITCH itself is initiating the SDP
negotiation with the SDP it provides within the 200OK.

On 3 March 2015 at 13:39, Brian West <brian at freeswitch.org> wrote:

> Thats odd, do you happen to know if the inbound call had an SAVPF?  It
> shouldn't enable that unless it smells webrtc in the SDP.  Have you ever
> enabled XML CDR's?  Those would help narrow this down probably.
>
> On Tue, Mar 3, 2015 at 5:27 AM, Rob Moore <Rob.Moore at aeriandi.com> wrote:
>
>>  Hi Brian,
>>
>>
>>
>> I thought as much, WebRTC isn?t something we are trying to use at the
>> moment (although im sure we?ll find a use for it in the not too distant
>> future.)
>>
>>
>>
>> Invites are created using the bridge application in XML dialplan.
>>
>>
>>
>> I have SIP pcaps but I don?t have and Freeswitch traces at the moment as
>> the issue is only appearing once in say 500 calls on our production system
>> so it can be a little awkward to pin down detailed tracing.
>>
>> I?m working on getting an example today and will post back as soon as
>> possible.
>>
>>
>>
>> Is there any way to disable WebRTC entirely? That could be worth a try
>> whilst I get a test setup for this scenario.
>>
>>
>>
>>
>> From: *Brian West* <brian at freeswitch.org>
>> Date: Mon, Mar 2, 2015 at 6:15 PM
>> Subject: Re: [Freeswitch-users] ICE SDP Params
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>> This is all related to WebRTC, how are you creating the invite?  Logs
>> would be helpful./b
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/05b7b58f/attachment.html 

From mogsy.uk at gmail.com  Wed Mar  4 17:17:42 2015
From: mogsy.uk at gmail.com (Rob Moore)
Date: Wed, 4 Mar 2015 14:17:42 +0000
Subject: [Freeswitch-users] ICE SDP Params
In-Reply-To: <CAEWH9M9_s9x0_H9Jp5KP7yhwVReVmndkpMpkhctKFMmkS=CePQ@mail.gmail.com>
References: <DB3PR05MB0286C6B3C5BC977C6BFB48383110@DB3PR05MB028.eurprd05.prod.outlook.com>
	<CAEJMVkAG+Jn07upn3JQLyhs7si7BfCb8fn0DybOv9Ce28cTUHQ@mail.gmail.com>
	<CAEWH9M9_s9x0_H9Jp5KP7yhwVReVmndkpMpkhctKFMmkS=CePQ@mail.gmail.com>
Message-ID: <CAJN0WQPy+P=z=JZgCR-kPLHGqQegh6gP-uA2qo7tjg56k92kvQ@mail.gmail.com>

Hi Brian,

 I've working today with our client to get a test instance of this issue
setup but to no avail. I have however been able to arrange some more
detailed tracing of a production instance of this issue which will provide
you a much better picture of whats going on.  I'm unable to post logs from
our production system to the user list without publishing sensitive
topology information so I've sent in a support ticket which contains the
following:

XML CDRs for a & b leg on an example call

PCAP trace of call flow for both legs.

Debug level Freeswitch logs for a leg and bleg.


Hopefully this will give you a much better picture of whats happening here.

Once we've got to the bottom of this. I'll post back here with any useful
information we find that other users might find beneficial in the future.

Thanks

Rob


On Wed, Mar 4, 2015 at 8:31 AM, Pete Kelly <pkelly at gmail.com> wrote:

> Hi Brian
>
> I have been working with Rob on the trace in question - the initial INVITE
> into FreeSWITCH has no SDP, so FreeSWITCH itself is initiating the SDP
> negotiation with the SDP it provides within the 200OK.
>
> On 3 March 2015 at 13:39, Brian West <brian at freeswitch.org> wrote:
>
>> Thats odd, do you happen to know if the inbound call had an SAVPF?  It
>> shouldn't enable that unless it smells webrtc in the SDP.  Have you ever
>> enabled XML CDR's?  Those would help narrow this down probably.
>>
>> On Tue, Mar 3, 2015 at 5:27 AM, Rob Moore <Rob.Moore at aeriandi.com> wrote:
>>
>>>  Hi Brian,
>>>
>>>
>>>
>>> I thought as much, WebRTC isn?t something we are trying to use at the
>>> moment (although im sure we?ll find a use for it in the not too distant
>>> future.)
>>>
>>>
>>>
>>> Invites are created using the bridge application in XML dialplan.
>>>
>>>
>>>
>>> I have SIP pcaps but I don?t have and Freeswitch traces at the moment as
>>> the issue is only appearing once in say 500 calls on our production system
>>> so it can be a little awkward to pin down detailed tracing.
>>>
>>> I?m working on getting an example today and will post back as soon as
>>> possible.
>>>
>>>
>>>
>>> Is there any way to disable WebRTC entirely? That could be worth a try
>>> whilst I get a test setup for this scenario.
>>>
>>>
>>>
>>>
>>> From: *Brian West* <brian at freeswitch.org>
>>> Date: Mon, Mar 2, 2015 at 6:15 PM
>>> Subject: Re: [Freeswitch-users] ICE SDP Params
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>
>>>
>>> This is all related to WebRTC, how are you creating the invite?  Logs
>>> would be helpful./b
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/c49425bf/attachment-0001.html 

From steveayre at gmail.com  Wed Mar  4 17:56:08 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 4 Mar 2015 14:56:08 +0000
Subject: [Freeswitch-users] $${sounds_dir} variable
In-Reply-To: <CAK+KT39=gngRa6D+JoRhOPPwq8+ce0X6jKqvQ5POZPDwKaoEOw@mail.gmail.com>
References: <CAK+KT39=gngRa6D+JoRhOPPwq8+ce0X6jKqvQ5POZPDwKaoEOw@mail.gmail.com>
Message-ID: <CAFiqYumTxFpauuxk0bK1NWiuKUN2TnKCWFbDRDRvff3JSmULSQ@mail.gmail.com>

The default is defined by the prefixes given when FreeSWITCH is compiled.

You can override it when freeswitch starts with the -sounds option.

On 3 March 2015 at 22:21, Richard Mace <richard.mace at gmail.com> wrote:

> Hi all,
> Could someone please let me know which file contains the defined location
> for the variable $${sounds_dir}
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/0aa1919a/attachment.html 

From bote_radio at botecomm.com  Wed Mar  4 19:35:32 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 4 Mar 2015 11:35:32 -0500
Subject: [Freeswitch-users] $${sounds_dir} variable
In-Reply-To: <CAFiqYumTxFpauuxk0bK1NWiuKUN2TnKCWFbDRDRvff3JSmULSQ@mail.gmail.com>
References: <CAK+KT39=gngRa6D+JoRhOPPwq8+ce0X6jKqvQ5POZPDwKaoEOw@mail.gmail.com>
	<CAFiqYumTxFpauuxk0bK1NWiuKUN2TnKCWFbDRDRvff3JSmULSQ@mail.gmail.com>
Message-ID: <067001d05699$3c6632a0$b53297e0$@botecomm.com>

I always start in conf/vars.xml and if I don?t find the variable defined there, I continue to look elsewhere.

 

In this case sounds_dir is defined as a pre-processor variable in vars.xml but can be overridden in a particular playback request if necessary. 

 

The command line switch mentioned by Steven provides another means to define it.

 

Bote

 

 

From: Steven Ayre
Sent: Wednesday, 04 March, 2015 09:56
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] $${sounds_dir} variable

 

The default is defined by the prefixes given when FreeSWITCH is compiled.

 

You can override it when freeswitch starts with the -sounds option.

 

On 3 March 2015 at 22:21, Richard Mace <richard.mace at gmail.com> wrote:

Hi all,

Could someone please let me know which file contains the defined location for the variable $${sounds_dir}

 

Thanks

 

Richard


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/c20bdccb/attachment.html 

From olegstolyar at gmail.com  Wed Mar  4 19:37:44 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 4 Mar 2015 08:37:44 -0800
Subject: [Freeswitch-users] Copying FreeSWITCH installation
Message-ID: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>

Hi guys,

I have a probably silly question.

If I have a fully functioning FS installation on (for instance) CentOS.
Will it work I simply copy the /usr/loca/freeswitch directory to another
CentOS machine with identical setup?

Or are there other places in the system (outside of the
/usr/local/freeswitch diectory)  that FS has files or settings in?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/922d77b1/attachment.html 

From raphael.lechner at gmail.com  Wed Mar  4 19:40:23 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Wed, 4 Mar 2015 17:40:23 +0100
Subject: [Freeswitch-users] Incoming Fax problems with bad rows
Message-ID: <53307FC0-B00D-4626-9757-6216FC4AF008@gmail.com>

Hi,

We have a problem that some incomings fax have many bad rows, from at least 2 different customer and therefore not all lines are readable.
We tried first with to remove the ATA Device and changed that, that we receive them by mod_spandsp but the problem still exists.

Attached the log file:
Sender 1:https://pastebin.freeswitch.org/23960 <https://pastebin.freeswitch.org/23960>
Sender 2: https://pastebin.freeswitch.org/23961 <https://pastebin.freeswitch.org/23961>

FreeSWITCH version:
1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)

We use a patton ISDN/VoIP Gateway.

Any hint how we can resolve/debug that?

Thank you,
Raphael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/aba1b97e/attachment.html 

From olegstolyar at gmail.com  Wed Mar  4 20:34:38 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 4 Mar 2015 09:34:38 -0800
Subject: [Freeswitch-users] Copying FreeSWITCH installation
In-Reply-To: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
References: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
Message-ID: <CAC8JMOJNA+UZRYFJ9F4d-kZbf=uZ=-99D-aEZUsCzWSAUVEpZg@mail.gmail.com>

Of course the target machine will have all the FS dependencies preinstalled.

On Wed, Mar 4, 2015 at 8:37 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Hi guys,
>
> I have a probably silly question.
>
> If I have a fully functioning FS installation on (for instance) CentOS.
> Will it work I simply copy the /usr/loca/freeswitch directory to another
> CentOS machine with identical setup?
>
> Or are there other places in the system (outside of the
> /usr/local/freeswitch diectory)  that FS has files or settings in?
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/3d7455a1/attachment.html 

From ing.antonyam at gmail.com  Wed Mar  4 20:40:09 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Wed, 4 Mar 2015 11:40:09 -0600
Subject: [Freeswitch-users] Help with mod xml_curl
Message-ID: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>

Hi


I have 1 freeswitch xml_curl directory configured and working properly
, now I want to add another freeswitch and see the same BD for the 2
fs .

The problem I have is that the same management extensions in the 2
freeswitch but they are 2 different ips , how can I do so that you can
see the same directory without the problem domain .

Anyone have an idea?

example

fs1 -> domain 1.1.1.1
fs2 -> domain 1.1.1.2

Have in common the extension 1000 but I can now register with a single
domain either that of the fs1 and fs2 .


regards.

r
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/76b29c58/attachment-0001.html 

From yehavi.bourvine at gmail.com  Wed Mar  4 20:43:30 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 4 Mar 2015 19:43:30 +0200
Subject: [Freeswitch-users] Copying FreeSWITCH installation
In-Reply-To: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
References: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
Message-ID: <CAAtWKeFRtsnzyiH820w7saEFNN=LUuchekvNBa=m0U1aRiaQbQ@mail.gmail.com>

I've done that several times (creating test machine, backup machine, etc.).
I copy that tree, delete all bin, mod, lib directories and build again the
binaries (just to be sure).

                        __Yehavi:

2015-03-04 18:37 GMT+02:00 Oleg Stolyar <olegstolyar at gmail.com>:

> Hi guys,
>
> I have a probably silly question.
>
> If I have a fully functioning FS installation on (for instance) CentOS.
> Will it work I simply copy the /usr/loca/freeswitch directory to another
> CentOS machine with identical setup?
>
> Or are there other places in the system (outside of the
> /usr/local/freeswitch diectory)  that FS has files or settings in?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/5b31e9d1/attachment.html 

From mike at jerris.com  Wed Mar  4 20:46:44 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 4 Mar 2015 12:46:44 -0500
Subject: [Freeswitch-users] Copying FreeSWITCH installation
In-Reply-To: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
References: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
Message-ID: <7AA3FA54-632F-47FB-BE90-D4C02434322F@jerris.com>

with default configure args, and as long as all dep libs are there on the new box.  It is possible to configure where this will not be the case.  For example, it is not for most of the packages.

> On Mar 4, 2015, at 11:37 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi guys,
> 
> I have a probably silly question.
> 
> If I have a fully functioning FS installation on (for instance) CentOS.  Will it work I simply copy the /usr/loca/freeswitch directory to another CentOS machine with identical setup?
> 
> Or are there other places in the system (outside of the /usr/local/freeswitch diectory)  that FS has files or settings in?=



From tony at intelecenter.com  Wed Mar  4 21:02:24 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 4 Mar 2015 10:02:24 -0800
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
Message-ID: <CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>

You can enable multiple domains on each Freeswitch instance following these
instructions:

https://wiki.freeswitch.org/wiki/Multiple_Companies

Then use domain names for your user directories rather than IP.

You can load balance between your Freeswitch servers using Opensips or
Kamailio or you can use DNS. You need a shared registration database with
this type of setup so each Freeswitch can route calls to registered
endpoints.

Thanks.

Tony Bourdeaux

On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <
ing.antonyam at gmail.com> wrote:

> Hi
>
>
> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
>
> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
>
> Anyone have an idea?
>
> example
>
> fs1 -> domain 1.1.1.1
> fs2 -> domain 1.1.1.2
>
> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
>
>
> regards.
>
> r
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/dfc6e629/attachment.html 

From olegstolyar at gmail.com  Wed Mar  4 21:37:05 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 4 Mar 2015 10:37:05 -0800
Subject: [Freeswitch-users] Copying FreeSWITCH installation
In-Reply-To: <7AA3FA54-632F-47FB-BE90-D4C02434322F@jerris.com>
References: <CAC8JMOL=X=xiLooO8z4yY-zvPzazJPVS560bEjLXFOwoVDHojA@mail.gmail.com>
	<7AA3FA54-632F-47FB-BE90-D4C02434322F@jerris.com>
Message-ID: <CAC8JMOL2XWb8dk-N0Ops4aKCZ-0QyLC9qHH0h4GCjsJq5bbF4Q@mail.gmail.com>

Thanks guys!

On Wed, Mar 4, 2015 at 9:46 AM, Michael Jerris <mike at jerris.com> wrote:

> with default configure args, and as long as all dep libs are there on the
> new box.  It is possible to configure where this will not be the case.  For
> example, it is not for most of the packages.
>
> > On Mar 4, 2015, at 11:37 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> >
> > Hi guys,
> >
> > I have a probably silly question.
> >
> > If I have a fully functioning FS installation on (for instance) CentOS.
> Will it work I simply copy the /usr/loca/freeswitch directory to another
> CentOS machine with identical setup?
> >
> > Or are there other places in the system (outside of the
> /usr/local/freeswitch diectory)  that FS has files or settings in?=
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/d07850ff/attachment.html 

From alhakeem at gmail.com  Wed Mar  4 23:04:53 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Wed, 4 Mar 2015 20:04:53 -0000
Subject: [Freeswitch-users] Freeswitch and sipML
In-Reply-To: <2BEC9C40-3157-425E-B053-58D0372F17CA@gmail.com>
References: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>	<C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>
	<2BEC9C40-3157-425E-B053-58D0372F17CA@gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAM1qZsSycuhJnvu2BUkZUdABAAAAAA==@gmail.com>

Hello Mike,
Are you referring to the mod_verto or a standalone custom lient verto ?
Cheers,
AH
 
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Naveen
Khanna
Sent: Tuesday, March 3, 2015 5:39 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch and sipML
 
Thanks for the inputs.
 
Regards,
 
Naveen Khanna
 
 
On 03-Mar-2015, at 10:29 am, Michael Jerris <mike at jerris.com> wrote:
 
I would reccomend using sip.js if it must be sip, or if sip is not a requirement
take a look at our own custom client verto.
 
 
On Mar 2, 2015, at 10:35 PM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
 
Hi,
 
I work on freeswitch to develop call centre solutions for customers. I am using
browser based sipML client & facing the problem of excessive amount of sessions
that do not close decently. This probably leads to hanging of my application.
Has anyone faced such problem & can someone suggest an effective solution /
safeguard. I would prefer using browser because it is effective for pop up
applications in call centre environment.
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/56c45bec/attachment-0001.html 

From mike at jerris.com  Wed Mar  4 23:09:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 4 Mar 2015 15:09:53 -0500
Subject: [Freeswitch-users] Freeswitch and sipML
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAM1qZsSycuhJnvu2BUkZUdABAAAAAA==@gmail.com>
References: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>
	<C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>
	<2BEC9C40-3157-425E-B053-58D0372F17CA@gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAM1qZsSycuhJnvu2BUkZUdABAAAAAA==@gmail.com>
Message-ID: <9C36AC0D-D532-4541-88CF-E32F87C12028@jerris.com>

I don't understand your question.

> On Mar 4, 2015, at 3:04 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello Mike,
> Are you referring to the mod_verto or a standalone custom lient verto ?
> Cheers,
> AH
> ? <>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Naveen Khanna
> Sent: Tuesday, March 3, 2015 5:39 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freeswitch and sipML
>  
> Thanks for the inputs.
>  
> Regards,
>  
> Naveen Khanna
>  
>> On 03-Mar-2015, at 10:29 am, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>  
>> I would reccomend using sip.js if it must be sip, or if sip is not a requirement take a look at our own custom client verto.
>>   
>>> On Mar 2, 2015, at 10:35 PM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>>>  
>>> Hi,
>>>  
>>> I work on freeswitch to develop call centre solutions for customers. I am using browser based sipML client & facing the problem of excessive amount of sessions that do not close decently. This probably leads to hanging of my application. Has anyone faced such problem & can someone suggest an effective solution / safeguard. I would prefer using browser because it is effective for pop up applications in call centre environment.
>> 
>>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/adc83320/attachment.html 

From vipkilla at gmail.com  Wed Mar  4 23:10:38 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 4 Mar 2015 15:10:38 -0500
Subject: [Freeswitch-users] Freeswitch and sipML
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAM1qZsSycuhJnvu2BUkZUdABAAAAAA==@gmail.com>
References: <9CD27BAA-61F1-4A6A-961A-E03FE6BF09F0@gmail.com>
	<C8DCE672-A263-4CD4-B2BC-2531AAB4BB11@jerris.com>
	<2BEC9C40-3157-425E-B053-58D0372F17CA@gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAM1qZsSycuhJnvu2BUkZUdABAAAAAA==@gmail.com>
Message-ID: <CAC-LwPPZXfJByGTcgwe9V3S+uYsBh3mdxX3CWOeR9+18w9DPmg@mail.gmail.com>

verto client and mod_verto are both required.
mod_verto is a FS endpoint module used to connect the verto client.

On Wed, Mar 4, 2015 at 3:04 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello Mike,
>
> Are you referring to the mod_verto or a standalone custom lient verto ?
>
> Cheers,
>
> AH
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Naveen
> Khanna
> *Sent:* Tuesday, March 3, 2015 5:39 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Freeswitch and sipML
>
>
>
> Thanks for the inputs.
>
>
>
> Regards,
>
>
>
> Naveen Khanna
>
>
>
>
>
> On 03-Mar-2015, at 10:29 am, Michael Jerris <mike at jerris.com> wrote:
>
>
>
> I would reccomend using sip.js if it must be sip, or if sip is not a
> requirement take a look at our own custom client verto.
>
>
>
>
>
> On Mar 2, 2015, at 10:35 PM, Naveen Khanna <naveen.khanna.bm at gmail.com>
> wrote:
>
>
>
> Hi,
>
>
>
> I work on freeswitch to develop call centre solutions for customers. I am
> using browser based sipML client & facing the problem of excessive amount
> of sessions that do not close decently. This probably leads to hanging of
> my application. Has anyone faced such problem & can someone suggest an
> effective solution / safeguard. I would prefer using browser because it is
> effective for pop up applications in call centre environment.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/31306619/attachment.html 

From s.safarov at gmail.com  Wed Mar  4 23:56:53 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 4 Mar 2015 23:56:53 +0300
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
Message-ID: <CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>

Tony is required configure "PostgreSQL in the core" for FS cluster in
Active-Active mode?
It is work?

On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com>
wrote:

> You can enable multiple domains on each Freeswitch instance following
> these instructions:
>
> https://wiki.freeswitch.org/wiki/Multiple_Companies
>
> Then use domain names for your user directories rather than IP.
>
> You can load balance between your Freeswitch servers using Opensips or
> Kamailio or you can use DNS. You need a shared registration database with
> this type of setup so each Freeswitch can route calls to registered
> endpoints.
>
> Thanks.
>
> Tony Bourdeaux
>
> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <
> ing.antonyam at gmail.com> wrote:
>
>> Hi
>>
>>
>> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
>>
>> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
>>
>> Anyone have an idea?
>>
>> example
>>
>> fs1 -> domain 1.1.1.1
>> fs2 -> domain 1.1.1.2
>>
>> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
>>
>>
>> regards.
>>
>> r
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Tony Bourdeaux
>
> *Intelecenter, LLC*
>
> ph: 805-428-3031
>
> Skype: tony.bourdeaux
>
>
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/56bad349/attachment-0001.html 

From tony at intelecenter.com  Thu Mar  5 00:32:00 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 4 Mar 2015 13:32:00 -0800
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
	<CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>
Message-ID: <CAANhKX++NvzdJqhcYuEGcDq-dhx8evG7WByt+Tn16SbyGZBL_g@mail.gmail.com>

yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect to
the same db server or cluster.

On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Tony is required configure "PostgreSQL in the core" for FS cluster in
> Active-Active mode?
> It is work?
>
> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com>
> wrote:
>
>> You can enable multiple domains on each Freeswitch instance following
>> these instructions:
>>
>> https://wiki.freeswitch.org/wiki/Multiple_Companies
>>
>> Then use domain names for your user directories rather than IP.
>>
>> You can load balance between your Freeswitch servers using Opensips or
>> Kamailio or you can use DNS. You need a shared registration database with
>> this type of setup so each Freeswitch can route calls to registered
>> endpoints.
>>
>> Thanks.
>>
>> Tony Bourdeaux
>>
>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <
>> ing.antonyam at gmail.com> wrote:
>>
>>> Hi
>>>
>>>
>>> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
>>>
>>> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
>>>
>>> Anyone have an idea?
>>>
>>> example
>>>
>>> fs1 -> domain 1.1.1.1
>>> fs2 -> domain 1.1.1.2
>>>
>>> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
>>>
>>>
>>> regards.
>>>
>>> r
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> Tony Bourdeaux
>>
>> *Intelecenter, LLC*
>>
>> ph: 805-428-3031
>>
>> Skype: tony.bourdeaux
>>
>>
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/6af33c5f/attachment.html 

From mike at jerris.com  Thu Mar  5 00:36:17 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 4 Mar 2015 16:36:17 -0500
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CAANhKX++NvzdJqhcYuEGcDq-dhx8evG7WByt+Tn16SbyGZBL_g@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
	<CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>
	<CAANhKX++NvzdJqhcYuEGcDq-dhx8evG7WByt+Tn16SbyGZBL_g@mail.gmail.com>
Message-ID: <3F682802-F46A-4387-9675-20C8DB019246@jerris.com>

You can't do mysql native, but you can over odbc.  That being said we have seen a lot of issues of the years with mysql in general so i wouldn't recommend that.



> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
> 
> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect to the same db server or cluster.
> 
> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Tony is required configure "PostgreSQL in the core" for FS cluster in Active-Active mode?
> It is work?
> 
> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com <mailto:tony at intelecenter.com>> wrote:
> You can enable multiple domains on each Freeswitch instance following these instructions:
> 
> https://wiki.freeswitch.org/wiki/Multiple_Companies <https://wiki.freeswitch.org/wiki/Multiple_Companies>
> 
> Then use domain names for your user directories rather than IP.
> 
> You can load balance between your Freeswitch servers using Opensips or Kamailio or you can use DNS. You need a shared registration database with this type of setup so each Freeswitch can route calls to registered endpoints.
> 
> Thanks.
> 
> Tony Bourdeaux
> 
> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote:
> Hi
> 
> 
> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
> 
> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
> 
> Anyone have an idea?
> 
> example
> 
> fs1 -> domain 1.1.1.1
> fs2 -> domain 1.1.1.2
> 
> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
> 
> regards.
> r
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Tony Bourdeaux <>
> Intelecenter, LLC <>
> ph: 805-428-3031 <tel:805-428-3031>
> Skype: tony.bourdeaux
>  
>  
> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>  
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Tony Bourdeaux <>
> Intelecenter, LLC <>
> ph: 805-428-3031
> Skype: tony.bourdeaux
>  
>  
> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/8c9a834b/attachment-0001.html 

From tony at intelecenter.com  Thu Mar  5 00:50:07 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 4 Mar 2015 13:50:07 -0800
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <3F682802-F46A-4387-9675-20C8DB019246@jerris.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
	<CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>
	<CAANhKX++NvzdJqhcYuEGcDq-dhx8evG7WByt+Tn16SbyGZBL_g@mail.gmail.com>
	<3F682802-F46A-4387-9675-20C8DB019246@jerris.com>
Message-ID: <CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>

Hi Michael-

have you had issues with MySql in general or in this type of configuration?
Just curious what types of issues?

On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com> wrote:

> You can't do mysql native, but you can over odbc.  That being said we have
> seen a lot of issues of the years with mysql in general so i wouldn't
> recommend that.
>
>
>
> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
>
> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect
> to the same db server or cluster.
>
> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Tony is required configure "PostgreSQL in the core" for FS cluster in
>> Active-Active mode?
>> It is work?
>>
>> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com>
>> wrote:
>>
>>> You can enable multiple domains on each Freeswitch instance following
>>> these instructions:
>>>
>>> https://wiki.freeswitch.org/wiki/Multiple_Companies
>>>
>>> Then use domain names for your user directories rather than IP.
>>>
>>> You can load balance between your Freeswitch servers using Opensips or
>>> Kamailio or you can use DNS. You need a shared registration database with
>>> this type of setup so each Freeswitch can route calls to registered
>>> endpoints.
>>>
>>> Thanks.
>>>
>>> Tony Bourdeaux
>>>
>>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <
>>> ing.antonyam at gmail.com> wrote:
>>>
>>>> Hi
>>>>
>>>>
>>>> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
>>>>
>>>> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
>>>>
>>>> Anyone have an idea?
>>>>
>>>> example
>>>>
>>>> fs1 -> domain 1.1.1.1
>>>> fs2 -> domain 1.1.1.2
>>>>
>>>> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
>>>>
>>>>
>>>> regards.
>>>>
>>>> r
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Tony Bourdeaux
>>> *Intelecenter, LLC*
>>> ph: 805-428-3031
>>> Skype: tony.bourdeaux
>>>
>>>
>>>
>>>
>>> "This message and any attachments are solely for the intended recipient
>>> and may contain confidential or privileged information. If you are not the
>>> intended recipient, any disclosure, copying, use, or distribution of the
>>> information included in this message and any attachments is prohibited. If
>>> you have received this communication in error, please notify me by reply
>>> e-mail and immediately and permanently delete this message and any
>>> attachments."
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Tony Bourdeaux
> *Intelecenter, LLC*
> ph: 805-428-3031
> Skype: tony.bourdeaux
>
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/bac1e164/attachment.html 

From jorgemariodlc at gmail.com  Thu Mar  5 01:17:47 2015
From: jorgemariodlc at gmail.com (jorgemariodlc)
Date: Wed, 4 Mar 2015 15:17:47 -0700 (MST)
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: <1425507467266-7596150.post@n2.nabble.com>

I actually working in it, I found it yesterday you need to add those lines
(/autoload_configs/lua.conf.xml):

<configuration name="lua.conf" description="LUA Configuration">
  <settings>
    
    <hook event="CHANNEL_ANSWER" script="answer.lua"/>
    <hook event="CHANNEL_HANGUP_COMPLETE" script="hangup.lua"/>

  </settings>
</configuration>

Check this link to know what information is in each event, because it's
important to handle the direction-call (Outbound, Inbound) 
https://wiki.freeswitch.org/wiki/Event_List 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Real-time-billing-application-for-the-FreeSWITCH-mod-lua-mod-perl-or-ESL-tp7593788p7596150.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mike at jerris.com  Thu Mar  5 01:26:09 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 4 Mar 2015 17:26:09 -0500
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKX+1hREBSoZJQBVJSGncx1Ug9crF0JPfHYKz3RdRvJ7Xrg@mail.gmail.com>
	<CAHtxdDd03D+mNSKKFUGoMXu=j8=-tqToHst8L2tVLryRG-7hDQ@mail.gmail.com>
	<CAANhKX++NvzdJqhcYuEGcDq-dhx8evG7WByt+Tn16SbyGZBL_g@mail.gmail.com>
	<3F682802-F46A-4387-9675-20C8DB019246@jerris.com>
	<CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
Message-ID: <4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>

In general.  We have seen tons of issues due to thread safety issues in the mysql odbc drivers.

> On Mar 4, 2015, at 4:50 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
> 
> Hi Michael-
> 
> have you had issues with MySql in general or in this type of configuration? Just curious what types of issues?
> 
> On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> You can't do mysql native, but you can over odbc.  That being said we have seen a lot of issues of the years with mysql in general so i wouldn't recommend that.
> 
> 
> 
>> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com <mailto:tony at intelecenter.com>> wrote:
>> 
>> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect to the same db server or cluster.
>> 
>> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> Tony is required configure "PostgreSQL in the core" for FS cluster in Active-Active mode?
>> It is work?
>> 
>> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com <mailto:tony at intelecenter.com>> wrote:
>> You can enable multiple domains on each Freeswitch instance following these instructions:
>> 
>> https://wiki.freeswitch.org/wiki/Multiple_Companies <https://wiki.freeswitch.org/wiki/Multiple_Companies>
>> 
>> Then use domain names for your user directories rather than IP.
>> 
>> You can load balance between your Freeswitch servers using Opensips or Kamailio or you can use DNS. You need a shared registration database with this type of setup so each Freeswitch can route calls to registered endpoints.
>> 
>> Thanks.
>> 
>> Tony Bourdeaux
>> 
>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote:
>> Hi
>> 
>> 
>> I have 1 freeswitch xml_curl directory configured and working properly , now I want to add another freeswitch and see the same BD for the 2 fs .
>> 
>> The problem I have is that the same management extensions in the 2 freeswitch but they are 2 different ips , how can I do so that you can see the same directory without the problem domain .
>> 
>> Anyone have an idea?
>> 
>> example
>> 
>> fs1 -> domain 1.1.1.1
>> fs2 -> domain 1.1.1.2
>> 
>> Have in common the extension 1000 but I can now register with a single domain either that of the fs1 and fs2 .
>> 
>> regards.
>> r
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> Tony Bourdeaux <>
>> Intelecenter, LLC <>
>> ph: 805-428-3031 <tel:805-428-3031>
>> Skype: tony.bourdeaux
>>  
>>  
>> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>>  
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> Tony Bourdeaux <>
>> Intelecenter, LLC <>
>> ph: 805-428-3031 <tel:805-428-3031>
>> Skype: tony.bourdeaux
>>  
>>  
>> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>>  
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Tony Bourdeaux <>
> Intelecenter, LLC <>
> ph: 805-428-3031
> Skype: tony.bourdeaux
>  
>  
> "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/ca1feccc/attachment-0001.html 

From sos at sokhapkin.dyndns.org  Thu Mar  5 01:38:30 2015
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 04 Mar 2015 17:38:30 -0500
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
	<4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>
Message-ID: <1709063.0cpCqfdO2y@sos>

Any pointers to mysql bug tracker? I don't see anything related there.

On Wednesday 04 March 2015 17:26:09 Michael Jerris wrote:
> In general.  We have seen tons of issues due to thread safety issues in the
> mysql odbc drivers.
> > On Mar 4, 2015, at 4:50 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
> > 
> > Hi Michael-
> > 
> > have you had issues with MySql in general or in this type of
> > configuration? Just curious what types of issues?
> > 
> > On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com
> > <mailto:mike at jerris.com>> wrote: You can't do mysql native, but you can
> > over odbc.  That being said we have seen a lot of issues of the years
> > with mysql in general so i wouldn't recommend that.> 
> >> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com
> >> <mailto:tony at intelecenter.com>> wrote:
> >> 
> >> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect
> >> to the same db server or cluster.
> >> 
> >> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com
> >> <mailto:s.safarov at gmail.com>> wrote: Tony is required configure
> >> "PostgreSQL in the core" for FS cluster in Active-Active mode? It is
> >> work?
> >> 
> >> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com
> >> <mailto:tony at intelecenter.com>> wrote: You can enable multiple domains
> >> on each Freeswitch instance following these instructions:
> >> 
> >> https://wiki.freeswitch.org/wiki/Multiple_Companies
> >> <https://wiki.freeswitch.org/wiki/Multiple_Companies>
> >> 
> >> Then use domain names for your user directories rather than IP.
> >> 
> >> You can load balance between your Freeswitch servers using Opensips or
> >> Kamailio or you can use DNS. You need a shared registration database
> >> with this type of setup so each Freeswitch can route calls to registered
> >> endpoints.
> >> 
> >> Thanks.
> >> 
> >> Tony Bourdeaux
> >> 
> >> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales
> >> <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote: Hi
> >> 
> >> 
> >> I have 1 freeswitch xml_curl directory configured and working properly ,
> >> now I want to add another freeswitch and see the same BD for the 2 fs .
> >> 
> >> The problem I have is that the same management extensions in the 2
> >> freeswitch but they are 2 different ips , how can I do so that you can
> >> see the same directory without the problem domain .
> >> 
> >> Anyone have an idea?
> >> 
> >> example
> >> 
> >> fs1 -> domain 1.1.1.1
> >> fs2 -> domain 1.1.1.2
> >> 
> >> Have in common the extension 1000 but I can now register with a single
> >> domain either that of the fs1 and fs2 .
> >> 
> >> regards.
> >> r
> >> 
> >> _________________________________________________________________________
> >> 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://confluence.freeswitch.org <http://confluence.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
> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> >> http://www.freeswitch.org <http://www.freeswitch.org/>
> > 
> > _________________________________________________________________________
> > 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://confluence.freeswitch.org <http://confluence.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
> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> > http://www.freeswitch.org <http://www.freeswitch.org/>


From ing.antonyam at gmail.com  Thu Mar  5 03:48:29 2015
From: ing.antonyam at gmail.com (Ing. Antonyam )
Date: Wed, 4 Mar 2015 18:48:29 -0600
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <1709063.0cpCqfdO2y@sos>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
	<4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>
	<1709063.0cpCqfdO2y@sos>
Message-ID: <39E22704-D509-4EA0-BCBD-A638F7E0B0FC@gmail.com>

Ok, tell them a little bit of architecture.

I have 2 fs, these contain this configuration:

- Core in DB
- Directory in DB [XML_CURL]
-dialplan [static from native fs]
- Configure [static from native fs]

I have set my opensips with the modules load balancer and dispatcher in are discharged 2 freeswitch and sends the requests according to the balancer.

In Part directory [XML_CURL] in the database have discharged 2 domains [ip] of freeswitch

+ ---- + ---------------- +
| Id | domain_name |
+ ---- + ---------------- +
| 1 | 1.1.1.1 |
| 2 | 1.1.1.2 |

and table is linked users

+ ---- + -------------- + ----------- + ------- +
| Id | username | domain_id | cache |
+ ---- + -------------- + ----------- + ------- +
| 1 | 1000ip | 1 | 0 |
+ ---- + -------------- + ----------- + ------- +

I currently connect to the ip of opensips [1.1.1.3] and which is responsible for sending the register to fs, but if the register reaches fs having ip 1.1.1.2 states that there is no extension to that domain, because that user is linked to in the above table 1 in the ip id 1.1.1.1.

my question is how can I make the extension or username created this not tied to a single domain if no reply to either of 2 fs.

Enviado desde mi iPhone

> El 04/03/2015, a las 16:38, Sergey Okhapkin <sos at sokhapkin.dyndns.org> escribi?:
> 
> Any pointers to mysql bug tracker? I don't see anything related there.
> 
>> On Wednesday 04 March 2015 17:26:09 Michael Jerris wrote:
>> In general.  We have seen tons of issues due to thread safety issues in the
>> mysql odbc drivers.
>>> On Mar 4, 2015, at 4:50 PM, Tony Bourdeaux <tony at intelecenter.com> wrote:
>>> 
>>> Hi Michael-
>>> 
>>> have you had issues with MySql in general or in this type of
>>> configuration? Just curious what types of issues?
>>> 
>>> On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com
>>> <mailto:mike at jerris.com>> wrote: You can't do mysql native, but you can
>>> over odbc.  That being said we have seen a lot of issues of the years
>>> with mysql in general so i wouldn't recommend that.> 
>>>> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com
>>>> <mailto:tony at intelecenter.com>> wrote:
>>>> 
>>>> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must connect
>>>> to the same db server or cluster.
>>>> 
>>>> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com
>>>> <mailto:s.safarov at gmail.com>> wrote: Tony is required configure
>>>> "PostgreSQL in the core" for FS cluster in Active-Active mode? It is
>>>> work?
>>>> 
>>>> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com
>>>> <mailto:tony at intelecenter.com>> wrote: You can enable multiple domains
>>>> on each Freeswitch instance following these instructions:
>>>> 
>>>> https://wiki.freeswitch.org/wiki/Multiple_Companies
>>>> <https://wiki.freeswitch.org/wiki/Multiple_Companies>
>>>> 
>>>> Then use domain names for your user directories rather than IP.
>>>> 
>>>> You can load balance between your Freeswitch servers using Opensips or
>>>> Kamailio or you can use DNS. You need a shared registration database
>>>> with this type of setup so each Freeswitch can route calls to registered
>>>> endpoints.
>>>> 
>>>> Thanks.
>>>> 
>>>> Tony Bourdeaux
>>>> 
>>>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales
>>>> <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote: Hi
>>>> 
>>>> 
>>>> I have 1 freeswitch xml_curl directory configured and working properly ,
>>>> now I want to add another freeswitch and see the same BD for the 2 fs .
>>>> 
>>>> The problem I have is that the same management extensions in the 2
>>>> freeswitch but they are 2 different ips , how can I do so that you can
>>>> see the same directory without the problem domain .
>>>> 
>>>> Anyone have an idea?
>>>> 
>>>> example
>>>> 
>>>> fs1 -> domain 1.1.1.1
>>>> fs2 -> domain 1.1.1.2
>>>> 
>>>> Have in common the extension 1000 but I can now register with a single
>>>> domain either that of the fs1 and fs2 .
>>>> 
>>>> regards.
>>>> r
>>>> 
>>>> _________________________________________________________________________
>>>> 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://confluence.freeswitch.org <http://confluence.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
>>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> 
>>> _________________________________________________________________________
>>> 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://confluence.freeswitch.org <http://confluence.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
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jpablolorenzetti at hotmail.com  Thu Mar  5 06:07:32 2015
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 5 Mar 2015 03:07:32 +0000
Subject: [Freeswitch-users] sessions and CS_INIT events
Message-ID: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>

Hi, i m writing a module in C that needs to check for certain information in a 
database for the caller and the destination number, 
for this the module is subscribing to the CS_INIT channel events, so everytime a channel is created 
the module callback is called and it checks the numbers, 
the problem is that the callback gets called twice, 
for the creation of the a-leg of the call and the creation of the b-leg. 
Is there any way to accomplish what i m trying to do ? 
Am i doing it the wrong way?
I have already try getting testing for the flags in the channel but it did not work,
testing of originator or originating does not yield anything ....
 
i might be doing it wrong maybe ? 

Thanks! 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/1b6fac64/attachment.html 

From tony at intelecenter.com  Thu Mar  5 06:57:29 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 4 Mar 2015 19:57:29 -0800
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <39E22704-D509-4EA0-BCBD-A638F7E0B0FC@gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
	<4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>
	<1709063.0cpCqfdO2y@sos>
	<39E22704-D509-4EA0-BCBD-A638F7E0B0FC@gmail.com>
Message-ID: <CAANhKXLe7cHf65AFFyvr+Vdq04vpz2CrLFvZbSwsHz2XchADCQ@mail.gmail.com>

use domain NAMES instead of IP's

On Wed, Mar 4, 2015 at 4:48 PM, Ing. Antonyam <ing.antonyam at gmail.com>
wrote:

> Ok, tell them a little bit of architecture.
>
> I have 2 fs, these contain this configuration:
>
> - Core in DB
> - Directory in DB [XML_CURL]
> -dialplan [static from native fs]
> - Configure [static from native fs]
>
> I have set my opensips with the modules load balancer and dispatcher in
> are discharged 2 freeswitch and sends the requests according to the
> balancer.
>
> In Part directory [XML_CURL] in the database have discharged 2 domains
> [ip] of freeswitch
>
> + ---- + ---------------- +
> | Id | domain_name |
> + ---- + ---------------- +
> | 1 | 1.1.1.1 |
> | 2 | 1.1.1.2 |
>
> and table is linked users
>
> + ---- + -------------- + ----------- + ------- +
> | Id | username | domain_id | cache |
> + ---- + -------------- + ----------- + ------- +
> | 1 | 1000ip | 1 | 0 |
> + ---- + -------------- + ----------- + ------- +
>
> I currently connect to the ip of opensips [1.1.1.3] and which is
> responsible for sending the register to fs, but if the register reaches fs
> having ip 1.1.1.2 states that there is no extension to that domain, because
> that user is linked to in the above table 1 in the ip id 1.1.1.1.
>
> my question is how can I make the extension or username created this not
> tied to a single domain if no reply to either of 2 fs.
>
> Enviado desde mi iPhone
>
> > El 04/03/2015, a las 16:38, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> escribi?:
> >
> > Any pointers to mysql bug tracker? I don't see anything related there.
> >
> >> On Wednesday 04 March 2015 17:26:09 Michael Jerris wrote:
> >> In general.  We have seen tons of issues due to thread safety issues in
> the
> >> mysql odbc drivers.
> >>> On Mar 4, 2015, at 4:50 PM, Tony Bourdeaux <tony at intelecenter.com>
> wrote:
> >>>
> >>> Hi Michael-
> >>>
> >>> have you had issues with MySql in general or in this type of
> >>> configuration? Just curious what types of issues?
> >>>
> >>> On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com
> >>> <mailto:mike at jerris.com>> wrote: You can't do mysql native, but you
> can
> >>> over odbc.  That being said we have seen a lot of issues of the years
> >>> with mysql in general so i wouldn't recommend that.>
> >>>> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com
> >>>> <mailto:tony at intelecenter.com>> wrote:
> >>>>
> >>>> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must
> connect
> >>>> to the same db server or cluster.
> >>>>
> >>>> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com
> >>>> <mailto:s.safarov at gmail.com>> wrote: Tony is required configure
> >>>> "PostgreSQL in the core" for FS cluster in Active-Active mode? It is
> >>>> work?
> >>>>
> >>>> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <tony at intelecenter.com
> >>>> <mailto:tony at intelecenter.com>> wrote: You can enable multiple
> domains
> >>>> on each Freeswitch instance following these instructions:
> >>>>
> >>>> https://wiki.freeswitch.org/wiki/Multiple_Companies
> >>>> <https://wiki.freeswitch.org/wiki/Multiple_Companies>
> >>>>
> >>>> Then use domain names for your user directories rather than IP.
> >>>>
> >>>> You can load balance between your Freeswitch servers using Opensips or
> >>>> Kamailio or you can use DNS. You need a shared registration database
> >>>> with this type of setup so each Freeswitch can route calls to
> registered
> >>>> endpoints.
> >>>>
> >>>> Thanks.
> >>>>
> >>>> Tony Bourdeaux
> >>>>
> >>>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales
> >>>> <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote: Hi
> >>>>
> >>>>
> >>>> I have 1 freeswitch xml_curl directory configured and working
> properly ,
> >>>> now I want to add another freeswitch and see the same BD for the 2 fs
> .
> >>>>
> >>>> The problem I have is that the same management extensions in the 2
> >>>> freeswitch but they are 2 different ips , how can I do so that you can
> >>>> see the same directory without the problem domain .
> >>>>
> >>>> Anyone have an idea?
> >>>>
> >>>> example
> >>>>
> >>>> fs1 -> domain 1.1.1.1
> >>>> fs2 -> domain 1.1.1.2
> >>>>
> >>>> Have in common the extension 1000 but I can now register with a single
> >>>> domain either that of the fs1 and fs2 .
> >>>>
> >>>> regards.
> >>>> r
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> 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://confluence.freeswitch.org <http://confluence.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
> >>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> >>>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>>
> >>>
> _________________________________________________________________________
> >>> 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://confluence.freeswitch.org <http://confluence.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
> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> >>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/c4b62474/attachment-0001.html 

From tony at intelecenter.com  Thu Mar  5 07:01:06 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Wed, 4 Mar 2015 20:01:06 -0800
Subject: [Freeswitch-users] Help with mod xml_curl
In-Reply-To: <CAANhKXLe7cHf65AFFyvr+Vdq04vpz2CrLFvZbSwsHz2XchADCQ@mail.gmail.com>
References: <CABFbAiWzCXkY3X1bsWTKgmv1hK9c4+eeGjsMmB9-M_OwtWRVyA@mail.gmail.com>
	<CAANhKXLuUFwHbp7nag0E4PGOnZW6p8RftOwmu9+N9n3qkMJ8gQ@mail.gmail.com>
	<4415AA6C-379A-49E7-9FEE-ED3C2881C71B@jerris.com>
	<1709063.0cpCqfdO2y@sos>
	<39E22704-D509-4EA0-BCBD-A638F7E0B0FC@gmail.com>
	<CAANhKXLe7cHf65AFFyvr+Vdq04vpz2CrLFvZbSwsHz2XchADCQ@mail.gmail.com>
Message-ID: <CAANhKX+m8sz=FNOvoQFQ5R537e2kjqX2=Xeuh3BxHjjkHSxKOg@mail.gmail.com>

take a look at this:

use the single IP  with multiple FS:

https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS

On Wed, Mar 4, 2015 at 7:57 PM, Tony Bourdeaux <tony at intelecenter.com>
wrote:

> use domain NAMES instead of IP's
>
> On Wed, Mar 4, 2015 at 4:48 PM, Ing. Antonyam <ing.antonyam at gmail.com>
> wrote:
>
>> Ok, tell them a little bit of architecture.
>>
>> I have 2 fs, these contain this configuration:
>>
>> - Core in DB
>> - Directory in DB [XML_CURL]
>> -dialplan [static from native fs]
>> - Configure [static from native fs]
>>
>> I have set my opensips with the modules load balancer and dispatcher in
>> are discharged 2 freeswitch and sends the requests according to the
>> balancer.
>>
>> In Part directory [XML_CURL] in the database have discharged 2 domains
>> [ip] of freeswitch
>>
>> + ---- + ---------------- +
>> | Id | domain_name |
>> + ---- + ---------------- +
>> | 1 | 1.1.1.1 |
>> | 2 | 1.1.1.2 |
>>
>> and table is linked users
>>
>> + ---- + -------------- + ----------- + ------- +
>> | Id | username | domain_id | cache |
>> + ---- + -------------- + ----------- + ------- +
>> | 1 | 1000ip | 1 | 0 |
>> + ---- + -------------- + ----------- + ------- +
>>
>> I currently connect to the ip of opensips [1.1.1.3] and which is
>> responsible for sending the register to fs, but if the register reaches fs
>> having ip 1.1.1.2 states that there is no extension to that domain, because
>> that user is linked to in the above table 1 in the ip id 1.1.1.1.
>>
>> my question is how can I make the extension or username created this not
>> tied to a single domain if no reply to either of 2 fs.
>>
>> Enviado desde mi iPhone
>>
>> > El 04/03/2015, a las 16:38, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>> escribi?:
>> >
>> > Any pointers to mysql bug tracker? I don't see anything related there.
>> >
>> >> On Wednesday 04 March 2015 17:26:09 Michael Jerris wrote:
>> >> In general.  We have seen tons of issues due to thread safety issues
>> in the
>> >> mysql odbc drivers.
>> >>> On Mar 4, 2015, at 4:50 PM, Tony Bourdeaux <tony at intelecenter.com>
>> wrote:
>> >>>
>> >>> Hi Michael-
>> >>>
>> >>> have you had issues with MySql in general or in this type of
>> >>> configuration? Just curious what types of issues?
>> >>>
>> >>> On Wed, Mar 4, 2015 at 1:36 PM, Michael Jerris <mike at jerris.com
>> >>> <mailto:mike at jerris.com>> wrote: You can't do mysql native, but you
>> can
>> >>> over odbc.  That being said we have seen a lot of issues of the years
>> >>> with mysql in general so i wouldn't recommend that.>
>> >>>> On Mar 4, 2015, at 4:32 PM, Tony Bourdeaux <tony at intelecenter.com
>> >>>> <mailto:tony at intelecenter.com>> wrote:
>> >>>>
>> >>>> yes- you can use PostgreSQL or ODBC for MySQL.  Each server must
>> connect
>> >>>> to the same db server or cluster.
>> >>>>
>> >>>> On Wed, Mar 4, 2015 at 12:56 PM, Sergey Safarov <s.safarov at gmail.com
>> >>>> <mailto:s.safarov at gmail.com>> wrote: Tony is required configure
>> >>>> "PostgreSQL in the core" for FS cluster in Active-Active mode? It is
>> >>>> work?
>> >>>>
>> >>>> On Wed, Mar 4, 2015 at 9:02 PM, Tony Bourdeaux <
>> tony at intelecenter.com
>> >>>> <mailto:tony at intelecenter.com>> wrote: You can enable multiple
>> domains
>> >>>> on each Freeswitch instance following these instructions:
>> >>>>
>> >>>> https://wiki.freeswitch.org/wiki/Multiple_Companies
>> >>>> <https://wiki.freeswitch.org/wiki/Multiple_Companies>
>> >>>>
>> >>>> Then use domain names for your user directories rather than IP.
>> >>>>
>> >>>> You can load balance between your Freeswitch servers using Opensips
>> or
>> >>>> Kamailio or you can use DNS. You need a shared registration database
>> >>>> with this type of setup so each Freeswitch can route calls to
>> registered
>> >>>> endpoints.
>> >>>>
>> >>>> Thanks.
>> >>>>
>> >>>> Tony Bourdeaux
>> >>>>
>> >>>> On Wed, Mar 4, 2015 at 9:40 AM, Antony Aguirre Morales
>> >>>> <ing.antonyam at gmail.com <mailto:ing.antonyam at gmail.com>> wrote: Hi
>> >>>>
>> >>>>
>> >>>> I have 1 freeswitch xml_curl directory configured and working
>> properly ,
>> >>>> now I want to add another freeswitch and see the same BD for the 2
>> fs .
>> >>>>
>> >>>> The problem I have is that the same management extensions in the 2
>> >>>> freeswitch but they are 2 different ips , how can I do so that you
>> can
>> >>>> see the same directory without the problem domain .
>> >>>>
>> >>>> Anyone have an idea?
>> >>>>
>> >>>> example
>> >>>>
>> >>>> fs1 -> domain 1.1.1.1
>> >>>> fs2 -> domain 1.1.1.2
>> >>>>
>> >>>> Have in common the extension 1000 but I can now register with a
>> single
>> >>>> domain either that of the fs1 and fs2 .
>> >>>>
>> >>>> regards.
>> >>>> r
>> >>>>
>> >>>>
>> _________________________________________________________________________
>> >>>> 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://confluence.freeswitch.org <http://confluence.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
>> >>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> >>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> 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://confluence.freeswitch.org <http://confluence.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
>> >>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> >>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
>
> Tony Bourdeaux
>
> *Intelecenter, LLC*
>
> ph: 805-428-3031
>
> Skype: tony.bourdeaux
>
>
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/e40d9125/attachment-0001.html 

From mishehu at freeswitch.org  Thu Mar  5 08:49:12 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 04 Mar 2015 23:49:12 -0600
Subject: [Freeswitch-users] module dependency
In-Reply-To: <CAHtxdDfWT_A1JCVC3EuGnuCiMhzJkr31A82ODYN05mNsBdJQtw@mail.gmail.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>	<CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>	<0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
	<CAHtxdDfWT_A1JCVC3EuGnuCiMhzJkr31A82ODYN05mNsBdJQtw@mail.gmail.com>
Message-ID: <54F7EE58.7060302@freeswitch.org>

It is highly recommended that you do a Jira and then create a branch for 
yourself to work on in our Stash system.  This will then allow you to 
create a branch in your own copy of the git repo, and then have the 
ability to submit a pull request back to the core dev team, at which 
time they can review your patches for inclusion to the main FreeSWITCH repo.

See 
https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Code 
and https://freeswitch.org/confluence/display/FREESWITCH/Pull+Requests 
for further information.

-- 
Yossi Neiman

On 03/03/2015 03:56 PM, Sergey Safarov wrote:

> Do I need to make a request in jira?
>
> On Tue, Mar 3, 2015 at 10:20 PM, Michael Jerris <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     yes it will require code changes there.  I wouldn't make an idle
>     loop tho.  I would do something to swap out the pointers with the
>     new ones and protect it all with a mutex.  I think we do something
>     similar with dialplan reload.
>
>
>>     On Mar 3, 2015, at 1:35 PM, Sergey Safarov <s.safarov at gmail.com
>>     <mailto:s.safarov at gmail.com>> wrote:
>>
>>     Will it help addition of the configuration update flag of module
>>     CORE_SOFTTIMER_MODULE.
>>     And to add idle loop 'into the function switch_lookup_timezone
>>     until 'update is complete?
>>
>>     On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com
>>     <mailto:mike at jerris.com>> wrote:
>>
>>         That is ALWAYS loaded before any other modules, so that not
>>         being loaded after.  Whats happening here, is the reload
>>         signal triggers the timezones to reload asynchronously.  This
>>         will require a code change to swap those out in some way that
>>         doesn't leave them empty for a short period, properly
>>         protected against race conditions.  This code is in
>>         switch_time.c.
>>
>>
>>         > On Mar 3, 2015, at 10:41 AM, Sergey Safarov
>>         <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>>         >
>>         > Please help me declare module dependency
>>         > I has extended module radius_cdr by timezone support and
>>         from time to time is getting following error
>>         >
>>         > freeswitch at internal> reload mod_radius_cdr
>>         > +OK Reloading XML
>>         > +OK module unloaded
>>         > +OK module loaded
>>         >
>>         > 2015-03-03 18:35:34.543407 [CONSOLE]
>>         switch_loadable_module.c:1935 Stopping: mod_radius_cdr
>>         > 2015-03-03 18:35:34.543407 [CONSOLE]
>>         switch_loadable_module.c:1955 mod_radius_cdr unloaded.
>>         > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
>>         > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324
>>         Timezone 'Asia/Tokyo' not found!
>>         > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992
>>         Cannot find timezone Asia/Tokyo
>>         > , Setting timezone to GMT
>>         > 2015-03-03 18:35:34.543407 [CONSOLE]
>>         switch_loadable_module.c:1465 Successfully Loaded
>>         [mod_radius_cdr]
>>         > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369
>>         Timezone reloaded 1781 definitions
>>         >
>>         >
>>         > Module currently depend of loaded configuradion of
>>         CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded
>>         CORE_SOFTTIMER_MODULE configuration.
>>         >
>>         > How can I make sure that CORE_SOFTTIMER_MODULE
>>         configuration is loaded  before mod_radius_cdr?
>>
>
>
>     _________________________________________________________________________
>     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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/a15e003d/attachment.html 

From mishehu at freeswitch.org  Thu Mar  5 09:07:21 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 05 Mar 2015 00:07:21 -0600
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
Message-ID: <54F7F299.2040909@freeswitch.org>

Is this a module that you are planning on releasing to the FreeSWITCH 
community?  If so, you'd have to provide a lot more information about 
what you're trying to do and *maybe* somebody would see a better way to 
do what you're trying to do.  My only guess is that you're not utilizing 
or consuming the data you receive in the event to properly determine 
whether or not to do your operations, but my guess is as good as any.

If this is, however, a closed module and you don't wish to share 
publicly the details, I think that you can contact 
consulting at freeswitch.org for information.  I think last I had heard 
that there is mutual NDA in the agreement, but you would have to contact 
them directly (and don't quote me :-)  )

-- 
Yossi Neiman

On 03/04/2015 09:07 PM, Juan Pablo L. wrote:
> Hi, i m writing a module in C that needs to check for certain information in a
> database for the caller and the destination number,
> for this the module is subscribing to the CS_INIT channel events, so everytime a channel is created
> the module callback is called and it checks the numbers,
> the problem is that the callback gets called twice,
> for the creation of the a-leg of the call and the creation of the b-leg.
> Is there any way to accomplish what i m trying to do ?
> Am i doing it the wrong way?
> I have already try getting testing for the flags in the channel but it did not work,
> testing of originator or originating does not yield anything ....
>   
> i might be doing it wrong maybe ?
>
> Thanks!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/0009263c/attachment.html 

From ssinyagin at gmail.com  Thu Mar  5 13:07:27 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 5 Mar 2015 11:07:27 +0100
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
Message-ID: <CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>

why at all do you need it to be a C module inside FreeSWITCH?

Why not writing an ESL program which would subscribe to events and
perform the needed actions?

How about the following scenario:

1. In the XML dialplan, you execute "park" application on the incoming call.

2. Your program is listening to events via ESL, and it recognizes that
a channel has been parked

3. Your program starts to playback the ringback tone into that channel

4. Your program performs all the needed lookups and sets needed
variables on the channel

5. Your program transfers or bridges the call where needed.

This is quite easy to implement in any programming language of your
choice, easy to debug, and it's easily scalable. It can be done in a
multi-threading fashion, like Go or Erlang, or even Java, and perform
as many parallel calls as required.

quite easy, and you don't have to mess with FreeSWITCH internals :)






On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
<jpablolorenzetti at hotmail.com> wrote:
> Hi, i m writing a module in C that needs to check for certain information in
> a
> database for the caller and the destination number,
> for this the module is subscribing to the CS_INIT channel events, so
> everytime a channel is created
> the module callback is called and it checks the numbers,
> the problem is that the callback gets called twice,
> for the creation of the a-leg of the call and the creation of the b-leg.
> Is there any way to accomplish what i m trying to do ?
> Am i doing it the wrong way?
> I have already try getting testing for the flags in the channel but it did
> not work,
> testing of originator or originating does not yield anything ....
>
> i might be doing it wrong maybe ?
>
> Thanks!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From telishisheer at gmail.com  Thu Mar  5 13:25:29 2015
From: telishisheer at gmail.com (Shisheer Teli)
Date: Thu, 5 Mar 2015 15:55:29 +0530
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CAEJMVkDExnoQdKgbpNq1xaRUgkzzdYr+WAYrZ0-iKzuk4rbshA@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
	<CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
	<CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
	<CAEJMVkDExnoQdKgbpNq1xaRUgkzzdYr+WAYrZ0-iKzuk4rbshA@mail.gmail.com>
Message-ID: <CACkRGLynnNswqmrd-5v3VybGQ2VTvhxLgP+UWJCSbO_SLqwhPQ@mail.gmail.com>

Hi Team,

I am able to  register extensions with IPv6 in freeswitch, but when i try
to call it says user not available.

Show registrations:
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata

1005,my_IPV4_serveraddress,ZjM3OTM4ZDg3NTU3MWI5NDE1YzFiZjkyNGM0YTZkY2U,sofia/internal/sip:1005 at ipv4_client_address
:54072;rinstance=2931f1f1dba24c55,1425551906,ipv6_clenet_address,54072,udp,RHEL62,

1003,my_IPV4_serveraddress,3da4821236b8221fd2636605fad3ec1c at 0
:0:0:0:0:0:0:0,sofia/internal-ipv6/sip:1003@
[ipv6_client_address]:5060;transport=udp;registering_acc=[ipv6server],1425550762,ipv6_client_address,5060,udp,RHEL62,

1002,my_IPV4_serveraddress,241dbd9396aff0d8f5c8b387e51fda7a at 0
:0:0:0:0:0:0:0,sofia/internal-ipv6/sip:1002@
[ipv6_client_address2]:5060;transport=udp;registering_acc=[ipv6server],1425550965,ipv6_client_address2,5060,udp,RHEL62,

1001,my_IPV4_serveraddress,b8230ebfb16fb8cb08246ba32b48f43a at 0
:0:0:0:0:0:0:0,sofia/internal-ipv6/sip:1001@
[ipv6_client_address3]:5060;transport=udp;registering_acc=[ipv6server],1425551003,ipv6_client_address3,5060,udp,RHEL62,




On Tue, Mar 3, 2015 at 7:12 PM, Brian West <brian at freeswitch.org> wrote:

> You'll need to use a domain name or use force-register-domain and
> force-register-db-domain to force the auth into a specific domain, the
> vanilla configs do this already so you've made extra steps to undo that.
>
> In addition I don't think we've ever added ipv6 ACL support either, so
> thats one that needs to be done at some point.
>
> On Tue, Mar 3, 2015 at 4:23 AM, Shisheer Teli <telishisheer at gmail.com>
> wrote:
>
>> Hi Team,
>>
>> My freeswitch server is on IPv6, and now i am able register extension
>> with IPv6 in freeswitch.
>>
>> but  i am unable to call from IPv6 extensions..
>>
>> Error:
>> 015-03-03 14:49:21.568064 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/internal-ipv6/1102@[clientipv6address]:5060
>> [60707716-c186-11e4-88f0-adeca182559b]
>> 2015-03-03 14:49:21.588040 [WARNING] sofia_reg.c:2827 Can't find user
>> [1102@[serveripv6address]] from clientipv6address
>> You must define a domain called '[serveripv6address]' in your directory
>> and add a user with the id="1102" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2015-03-03 14:49:21.588040 [NOTICE] sofia.c:2063 Hangup
>> sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_NEW]
>> [CALL_REJECTED]
>> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1641 Session 1
>> (sofia/internal-ipv6/1102@[clientipv6address]:5060) Ended
>> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1645 Close
>> Channel sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_DESTROY]
>>
>> Regards,
>> Shisheer
>>
>>
>> On Tue, Mar 3, 2015 at 2:39 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> but you didn't provide any information, so it's difficult to help.
>>>
>>> On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com>
>>> wrote:
>>> > Hi Team,
>>> >
>>> > My freeswitch server is on IPv6, and now i am able register extension
>>> with
>>> > IPv6 in freeswitch.
>>> >
>>> > but  i am unable to call from IPv6 extensions..
>>> >
>>> > can help ..?
>>> >
>>> > Regards,
>>> > shisheer T
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Regards,
>> Shisheer Teli
>> Phone: +91-022 2278 2519 / 2121
>> shisheer at tifr.res.in
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Shisheer Teli
Phone: +91-022 2278 2519 / 2121
shisheer at tifr.res.in
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/0ced83fd/attachment.html 

From steveayre at gmail.com  Thu Mar  5 14:46:03 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 5 Mar 2015 11:46:03 +0000
Subject: [Freeswitch-users] unable to call in freeswitch IPv6
In-Reply-To: <CAEJMVkDExnoQdKgbpNq1xaRUgkzzdYr+WAYrZ0-iKzuk4rbshA@mail.gmail.com>
References: <CACkRGLzpa7Ro8B7J_OqquSozs59LqwZKsEDvC7jw5=u-QiWr-A@mail.gmail.com>
	<CACOYK=hVOfzgiwPDJf2ZOQ3wNHGgV5d5Lkb+D5gghL3K=J11hA@mail.gmail.com>
	<CACkRGLzzJxhK1bb0bAjSys07fWrqqqi_KaFsyjZ6=AKPD_N-fg@mail.gmail.com>
	<CAEJMVkDExnoQdKgbpNq1xaRUgkzzdYr+WAYrZ0-iKzuk4rbshA@mail.gmail.com>
Message-ID: <CAFiqYunVhDbL9+RvJ7AA1X4pEM5JBgoAa7So837vMtdoBO0XoQ@mail.gmail.com>

>
> In addition I don't think we've ever added ipv6 ACL support either, so
> thats one that needs to be done at some point.
>

It seems undocumented but looking at the source code it looks like it has
been implemented.



On 3 March 2015 at 13:42, Brian West <brian at freeswitch.org> wrote:

> You'll need to use a domain name or use force-register-domain and
> force-register-db-domain to force the auth into a specific domain, the
> vanilla configs do this already so you've made extra steps to undo that.
>
> In addition I don't think we've ever added ipv6 ACL support either, so
> thats one that needs to be done at some point.
>
> On Tue, Mar 3, 2015 at 4:23 AM, Shisheer Teli <telishisheer at gmail.com>
> wrote:
>
>> Hi Team,
>>
>> My freeswitch server is on IPv6, and now i am able register extension
>> with IPv6 in freeswitch.
>>
>> but  i am unable to call from IPv6 extensions..
>>
>> Error:
>> 015-03-03 14:49:21.568064 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/internal-ipv6/1102@[clientipv6address]:5060
>> [60707716-c186-11e4-88f0-adeca182559b]
>> 2015-03-03 14:49:21.588040 [WARNING] sofia_reg.c:2827 Can't find user
>> [1102@[serveripv6address]] from clientipv6address
>> You must define a domain called '[serveripv6address]' in your directory
>> and add a user with the id="1102" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2015-03-03 14:49:21.588040 [NOTICE] sofia.c:2063 Hangup
>> sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_NEW]
>> [CALL_REJECTED]
>> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1641 Session 1
>> (sofia/internal-ipv6/1102@[clientipv6address]:5060) Ended
>> 2015-03-03 14:49:21.608037 [NOTICE] switch_core_session.c:1645 Close
>> Channel sofia/internal-ipv6/1102@[clientipv6address]:5060 [CS_DESTROY]
>>
>> Regards,
>> Shisheer
>>
>>
>> On Tue, Mar 3, 2015 at 2:39 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> but you didn't provide any information, so it's difficult to help.
>>>
>>> On Tue, Mar 3, 2015 at 7:03 AM, Shisheer Teli <telishisheer at gmail.com>
>>> wrote:
>>> > Hi Team,
>>> >
>>> > My freeswitch server is on IPv6, and now i am able register extension
>>> with
>>> > IPv6 in freeswitch.
>>> >
>>> > but  i am unable to call from IPv6 extensions..
>>> >
>>> > can help ..?
>>> >
>>> > Regards,
>>> > shisheer T
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Regards,
>> Shisheer Teli
>> Phone: +91-022 2278 2519 / 2121
>> shisheer at tifr.res.in
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/9540a0d3/attachment-0001.html 

From vipkilla at gmail.com  Thu Mar  5 15:56:33 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 5 Mar 2015 07:56:33 -0500
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
Message-ID: <CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>

It all depends on what you are trying to do with your module.
You can use a dialplan handler in your module (see mod_enum for example) to
route inbound calls using your custom dialplan.
You can use a state handler in your module and bind to channel states (much
like binding to events).
You can create a dialplan app in your module to execute code when the app
is called in dialplan
(Example: <application action="mycustom_module" data="${uuid} arg1 arg2"/>)
You can use an endpoint in your module to originate calls outbound (see
mod_lcr or mod_callcenter for an example)
Also, you can create an API for your module

IMO creating a module is much more powerful than using a script with ESL.
But if you are going to create a module, you really don't need to mess with
events (unless they are very specific events like CUSTOM::) because your
module has access to much of the freeswitch core.

Thanks.



On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> why at all do you need it to be a C module inside FreeSWITCH?
>
> Why not writing an ESL program which would subscribe to events and
> perform the needed actions?
>
> How about the following scenario:
>
> 1. In the XML dialplan, you execute "park" application on the incoming
> call.
>
> 2. Your program is listening to events via ESL, and it recognizes that
> a channel has been parked
>
> 3. Your program starts to playback the ringback tone into that channel
>
> 4. Your program performs all the needed lookups and sets needed
> variables on the channel
>
> 5. Your program transfers or bridges the call where needed.
>
> This is quite easy to implement in any programming language of your
> choice, easy to debug, and it's easily scalable. It can be done in a
> multi-threading fashion, like Go or Erlang, or even Java, and perform
> as many parallel calls as required.
>
> quite easy, and you don't have to mess with FreeSWITCH internals :)
>
>
>
>
>
>
> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> <jpablolorenzetti at hotmail.com> wrote:
> > Hi, i m writing a module in C that needs to check for certain
> information in
> > a
> > database for the caller and the destination number,
> > for this the module is subscribing to the CS_INIT channel events, so
> > everytime a channel is created
> > the module callback is called and it checks the numbers,
> > the problem is that the callback gets called twice,
> > for the creation of the a-leg of the call and the creation of the b-leg.
> > Is there any way to accomplish what i m trying to do ?
> > Am i doing it the wrong way?
> > I have already try getting testing for the flags in the channel but it
> did
> > not work,
> > testing of originator or originating does not yield anything ....
> >
> > i might be doing it wrong maybe ?
> >
> > Thanks!
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/215dfd2a/attachment.html 

From ssinyagin at gmail.com  Thu Mar  5 16:19:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 5 Mar 2015 14:19:51 +0100
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
	<CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>
Message-ID: <CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>

but for the task that OP has described, writing (and maintaining it in
the long term) a module is really an overkill. Plus, he would also
need to take care of multithreading within FreeSWITCH, as well as
memory management, etc.

Also, a module makes sense if it's some common task which can be
re-used by others and published as open source. If it's some
closed-source module for a specific enterprise task that Juan has, it
just doesn't make sense and too much risk for a long-term solution.





On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> It all depends on what you are trying to do with your module.
> You can use a dialplan handler in your module (see mod_enum for example) to
> route inbound calls using your custom dialplan.
> You can use a state handler in your module and bind to channel states (much
> like binding to events).
> You can create a dialplan app in your module to execute code when the app is
> called in dialplan
> (Example: <application action="mycustom_module" data="${uuid} arg1 arg2"/>)
> You can use an endpoint in your module to originate calls outbound (see
> mod_lcr or mod_callcenter for an example)
> Also, you can create an API for your module
>
> IMO creating a module is much more powerful than using a script with ESL.
> But if you are going to create a module, you really don't need to mess with
> events (unless they are very specific events like CUSTOM::) because your
> module has access to much of the freeswitch core.
>
> Thanks.
>
>
>
> On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>>
>> why at all do you need it to be a C module inside FreeSWITCH?
>>
>> Why not writing an ESL program which would subscribe to events and
>> perform the needed actions?
>>
>> How about the following scenario:
>>
>> 1. In the XML dialplan, you execute "park" application on the incoming
>> call.
>>
>> 2. Your program is listening to events via ESL, and it recognizes that
>> a channel has been parked
>>
>> 3. Your program starts to playback the ringback tone into that channel
>>
>> 4. Your program performs all the needed lookups and sets needed
>> variables on the channel
>>
>> 5. Your program transfers or bridges the call where needed.
>>
>> This is quite easy to implement in any programming language of your
>> choice, easy to debug, and it's easily scalable. It can be done in a
>> multi-threading fashion, like Go or Erlang, or even Java, and perform
>> as many parallel calls as required.
>>
>> quite easy, and you don't have to mess with FreeSWITCH internals :)
>>
>>
>>
>>
>>
>>
>> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
>> <jpablolorenzetti at hotmail.com> wrote:
>> > Hi, i m writing a module in C that needs to check for certain
>> > information in
>> > a
>> > database for the caller and the destination number,
>> > for this the module is subscribing to the CS_INIT channel events, so
>> > everytime a channel is created
>> > the module callback is called and it checks the numbers,
>> > the problem is that the callback gets called twice,
>> > for the creation of the a-leg of the call and the creation of the b-leg.
>> > Is there any way to accomplish what i m trying to do ?
>> > Am i doing it the wrong way?
>> > I have already try getting testing for the flags in the channel but it
>> > did
>> > not work,
>> > testing of originator or originating does not yield anything ....
>> >
>> > i might be doing it wrong maybe ?
>> >
>> > Thanks!
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From aqsyounas at gmail.com  Thu Mar  5 16:24:27 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 5 Mar 2015 18:24:27 +0500
Subject: [Freeswitch-users] freeswitch got killed
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315CD4A93@ex2.connex.local>
References: <CAGVOjVXaz7PM-D=UDbV4yHfC6_xDyLoDGcb6_M0oZeMAvP_fVQ@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315CD4A93@ex2.connex.local>
Message-ID: <CAGVOjVURXDYxUvUh9Ww2hPqg+M7fXT2eKqPTeWUUSXnXro6i2w@mail.gmail.com>

Thanks for your reply. I am getting more date to make sure actually it is
bug.

On 2 March 2015 at 18:43, Moishe Grunstein <max at nysolutions.com> wrote:

>
> https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aqs Younas
> *Sent:* Monday, March 2, 2015 7:16 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] freeswitch got killed
>
>
>
> Hi, users.
>
>
>
> I am playing streams with mod_vlc, but some streams make my switch killed.
>
> I am using the lasted git version.
>
> FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git
> 4909cdb 2015-02-24 20:58:26Z 64bit)
>
>   Logs that i see are these, also log files is attached.
>
>
>
> 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute
> log(${cur})
> 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute
> set(episode=0${last_matching_digits})
> 2015-03-02 05:37:49.554042 [NOTICE] switch_core_session.c:3000 Execute
> curl(http://206.225.05.12/rd_api/api/inboundcampaign/get_extension post
> ext=${episode}&did=${dst})
> 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
> log(${curl_response_data})
> 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
> set(error=No)
> 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
> set(cur=${curl_response_data})
> 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
> log(${cur})
> 2015-03-02 05:37:49.594042 [NOTICE] switch_core_session.c:3000 Execute
> transfer(${cur} XML play)
> 2015-03-02 05:37:49.594042 [NOTICE] switch_ivr.c:1861 Transfer
> sofia/external/19546006100 at 69.27.168.33:5060 to XML[
> 95.81.147.3/rfimonde/all/rfimonde-64k.mp3 at play]
> 2015-03-02 05:37:50.094042 [INFO] mod_dialplan_xml.c:635 Processing
> 19546006100 <19546006100>->95.81.147.3/rfimonde/all/rfimonde-64k.mp3 in
> context play
> 2015-03-02 05:37:50.094042 [INFO] switch_ivr_async.c:212 Digit parser
> DPTOOLS: Setting realm to 'moderator'
> 2015-03-02 05:37:50.114043 [NOTICE] mod_vlc.c:192 VLC Path is http
> http://95.81.147.3/rfimonde/all/rfimonde-64k.mp3
> [0x25e099b8] access_http access: Raw-audio server found, mp3 demuxer
> selected
> [0x7fa3fc2f69a8] mpgatofixed32 audio converter error: libmad error: bad
> main_data_begin pointer
> 2015-03-02 05:45:38.494035 [NOTICE] sofia.c:952 Hangup sofia/external/
> 18034805839 at 69.27.168.71:5060 [CS_EXECUTE] [NORMAL_CLEARING]
> 2015-03-02 05:45:38.534034 [INFO] mod_json_cdr.c:271 Process
> [f7c5c71a-438b-4c56-938e-34cb19766fd6.cdr.json]
> 2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1641 Session
> 2591 (sofia/external/18034805839 at 69.27.168.71:5060) Ended
> 2015-03-02 05:45:38.914042 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/external/18034805839 at 69.27.168.71:5060 [CS_DESTROY]
> [0x7fa47bf55668] Killed
>
> What is believe is that freeswitch must not be killed even if stream is
> bad.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/90fc083c/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/90fc083c/attachment-0001.jpg 

From aqsyounas at gmail.com  Thu Mar  5 16:28:42 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 5 Mar 2015 18:28:42 +0500
Subject: [Freeswitch-users] How can i create dynamic dialplan in
	freeswitch.
In-Reply-To: <CAKzWOxXa9M+Q+1wA-JVjrmd5_xXPUd3YggOMp+KOdF8BExFLTg@mail.gmail.com>
References: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>
	<CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>
	<CAKzWOxXa9M+Q+1wA-JVjrmd5_xXPUd3YggOMp+KOdF8BExFLTg@mail.gmail.com>
Message-ID: <CAGVOjVVWAgWVowMajPA1oFTcdRq2aqbUF1A4+_fjX_Fk3L+R=g@mail.gmail.com>

Thanks for your reply. I see mod_xml_curl using apache and so does
mod_httapi. An extensive back and forth switching would make a lot of load
on apache. So, i am creating something with mod_lua.

On 2 March 2015 at 23:42, Michael Collins <msc at freeswitch.org> wrote:

> See also chapter 9 of the FreeSWITCH 1.2 book, appropriately entitled,
> "Moving Beyond the Static XML Configuration."
>
> -MC
>
> On Mon, Mar 2, 2015 at 9:24 AM, Vik Killa <vipkilla at gmail.com> wrote:
>
>> Hi,
>> Look at mod_xml_curl to do a 'dynamic' dialplan.
>> Thanks.
>>
>> On Mon, Mar 2, 2015 at 12:21 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi, user.
>>>
>>> After working for more than 3 months while writing my dialplan in static
>>> xml file,but now wants to know how can i effectively create dynamic
>>> dialplan in freeswitch.
>>>
>>> Thanks.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/f562053e/attachment.html 

From aqsyounas at gmail.com  Thu Mar  5 16:38:41 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 5 Mar 2015 18:38:41 +0500
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
	hangup cause?
Message-ID: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>

Hi, list.

I see my freeswitch hanging a lot of calls with INCOMPATIBLE_DESTINATION as
hangup cause in my cdr though the DID they are hitting is a proper number.

Could someone please tells me why freeswitch is hanging calls with this
reason.

Thanks,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/64c1aac9/attachment.html 

From grcamauer at gmail.com  Thu Mar  5 16:52:24 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 5 Mar 2015 10:52:24 -0300
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
	hangup cause?
In-Reply-To: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>
References: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>
Message-ID: <08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>

Check to see that there is a common codec being offered by both sides of the call.

Guillermo

Sent from my iPhone

> On 5/3/2015, at 10:38, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Hi, list.
> 
> I see my freeswitch hanging a lot of calls with INCOMPATIBLE_DESTINATION as hangup cause in my cdr though the DID they are hitting is a proper number. 
> 
> Could someone please tells me why freeswitch is hanging calls with this reason.
> 
> Thanks,
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From aqsyounas at gmail.com  Thu Mar  5 16:59:30 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Thu, 5 Mar 2015 18:59:30 +0500
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
 hangup cause?
In-Reply-To: <08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>
References: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>
	<08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>
Message-ID: <CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>

My freeswitch just answers the call and plays a mp3 file. Can i do anything
to make this minimum,? because Vendor is sending us calls with most of them
get through but some just hangup with this cause.

Can i make my freeswitch to support maximum codecs?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/4457b838/attachment.html 

From jpablolorenzetti at hotmail.com  Thu Mar  5 17:23:08 2015
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 5 Mar 2015 14:23:08 +0000
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>,
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>,
	<CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>,
	<CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>
Message-ID: <DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>

Thank you very much guys for your contributions.

I m doing it as a module for couple of reasons, being the
most important (i believe) performance, because the module
i m working on is to do real time charging of voice calls on a switch
that is already  serving as a RBT service plus a bunch of IVR's to purchase
services, this is for a ~150K user base on a single machine (cold standby)
this switch is also scheduled to soon start providing hosted PBX services, 
so going the script direction
i personally dont see that as an option at all. I do use scripts for small no so much used
much simpler stuff though, e.g: a lua script takes care of authenticating users
when doing international calls from company extensions in the hosted PBX solution.

The other reason i chose to do
this as a module because C is the language i feel more comfortable with. 
i hope this clarifies i little bit this.

Moving on, right now i m developing on a test freeswitch that we have and yes i noticed
that subscribing to the CS_INIT event does represent a big problem
because i get notified for every single of those events that is generated on
freeswitch which would be very inconvenient because as i mentioned, the same
switch does many other things that i m not interested in, so i m going to try the advise
provided and try to do it in the dial plan, i will explore this option. 

thank you very much all!





> Date: Thu, 5 Mar 2015 14:19:51 +0100
> From: ssinyagin at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sessions and CS_INIT events
> 
> but for the task that OP has described, writing (and maintaining it in
> the long term) a module is really an overkill. Plus, he would also
> need to take care of multithreading within FreeSWITCH, as well as
> memory management, etc.
> 
> Also, a module makes sense if it's some common task which can be
> re-used by others and published as open source. If it's some
> closed-source module for a specific enterprise task that Juan has, it
> just doesn't make sense and too much risk for a long-term solution.
> 
> 
> 
> 
> 
> On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > It all depends on what you are trying to do with your module.
> > You can use a dialplan handler in your module (see mod_enum for example) to
> > route inbound calls using your custom dialplan.
> > You can use a state handler in your module and bind to channel states (much
> > like binding to events).
> > You can create a dialplan app in your module to execute code when the app is
> > called in dialplan
> > (Example: <application action="mycustom_module" data="${uuid} arg1 arg2"/>)
> > You can use an endpoint in your module to originate calls outbound (see
> > mod_lcr or mod_callcenter for an example)
> > Also, you can create an API for your module
> >
> > IMO creating a module is much more powerful than using a script with ESL.
> > But if you are going to create a module, you really don't need to mess with
> > events (unless they are very specific events like CUSTOM::) because your
> > module has access to much of the freeswitch core.
> >
> > Thanks.
> >
> >
> >
> > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> > wrote:
> >>
> >> why at all do you need it to be a C module inside FreeSWITCH?
> >>
> >> Why not writing an ESL program which would subscribe to events and
> >> perform the needed actions?
> >>
> >> How about the following scenario:
> >>
> >> 1. In the XML dialplan, you execute "park" application on the incoming
> >> call.
> >>
> >> 2. Your program is listening to events via ESL, and it recognizes that
> >> a channel has been parked
> >>
> >> 3. Your program starts to playback the ringback tone into that channel
> >>
> >> 4. Your program performs all the needed lookups and sets needed
> >> variables on the channel
> >>
> >> 5. Your program transfers or bridges the call where needed.
> >>
> >> This is quite easy to implement in any programming language of your
> >> choice, easy to debug, and it's easily scalable. It can be done in a
> >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> >> as many parallel calls as required.
> >>
> >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> >> <jpablolorenzetti at hotmail.com> wrote:
> >> > Hi, i m writing a module in C that needs to check for certain
> >> > information in
> >> > a
> >> > database for the caller and the destination number,
> >> > for this the module is subscribing to the CS_INIT channel events, so
> >> > everytime a channel is created
> >> > the module callback is called and it checks the numbers,
> >> > the problem is that the callback gets called twice,
> >> > for the creation of the a-leg of the call and the creation of the b-leg.
> >> > Is there any way to accomplish what i m trying to do ?
> >> > Am i doing it the wrong way?
> >> > I have already try getting testing for the flags in the channel but it
> >> > did
> >> > not work,
> >> > testing of originator or originating does not yield anything ....
> >> >
> >> > i might be doing it wrong maybe ?
> >> >
> >> > Thanks!
> >> >
> >> >
> >> >
> >> > _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://confluence.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/7f4cbaf2/attachment-0001.html 

From ssinyagin at gmail.com  Thu Mar  5 17:27:45 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 5 Mar 2015 15:27:45 +0100
Subject: [Freeswitch-users] Debian package freeswitch-all
Message-ID: <CACOYK=hXBS4CwPu9ozsVgdXwJRkATuD-QuRacJYGyh6Lby8xRg@mail.gmail.com>

hi,

I just noted that "freeswitch-all" package does not contain speex and
opus. And of course "freeswitch-mod-speex" and  "freeswitch-mod-opus"
conflict with it (I know that "freeswitch-meta-all" would solve the
problem, but I like the single-package approach).

Is it by intent or was it just missed out?


thanks


From vipkilla at gmail.com  Thu Mar  5 17:30:29 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 5 Mar 2015 09:30:29 -0500
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
	<CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>
	<CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>
	<DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>
Message-ID: <CAC-LwPMHFwCikZa+5GcU8J=EuNaJ0wY0nOeNo9afaqv5-iR+UQ@mail.gmail.com>

Juan,
You may want to use a state handler in your module instead of using events.


On Thu, Mar 5, 2015 at 9:23 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com>
wrote:

> Thank you very much guys for your contributions.
>
> I m doing it as a module for couple of reasons, being the
> most important (i believe) performance, because the module
> i m working on is to do real time charging of voice calls on a switch
> that is already  serving as a RBT service plus a bunch of IVR's to purchase
> services, this is for a ~150K user base on a single machine (cold standby)
> this switch is also scheduled to soon start providing hosted PBX services,
> so going the script direction
> i personally dont see that as an option at all. I do use scripts for small
> no so much used
> much simpler stuff though, e.g: a lua script takes care of authenticating
> users
> when doing international calls from company extensions in the hosted PBX
> solution.
>
> The other reason i chose to do
> this as a module because C is the language i feel more comfortable with.
> i hope this clarifies i little bit this.
>
> Moving on, right now i m developing on a test freeswitch that we have and
> yes i noticed
> that subscribing to the CS_INIT event does represent a big problem
> because i get notified for every single of those events that is generated
> on
> freeswitch which would be very inconvenient because as i mentioned, the
> same
> switch does many other things that i m not interested in, so i m going to
> try the advise
> provided and try to do it in the dial plan, i will explore this option.
>
> thank you very much all!
>
>
>
>
>
> > Date: Thu, 5 Mar 2015 14:19:51 +0100
> > From: ssinyagin at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] sessions and CS_INIT events
>
> >
> > but for the task that OP has described, writing (and maintaining it in
> > the long term) a module is really an overkill. Plus, he would also
> > need to take care of multithreading within FreeSWITCH, as well as
> > memory management, etc.
> >
> > Also, a module makes sense if it's some common task which can be
> > re-used by others and published as open source. If it's some
> > closed-source module for a specific enterprise task that Juan has, it
> > just doesn't make sense and too much risk for a long-term solution.
> >
> >
> >
> >
> >
> > On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > > It all depends on what you are trying to do with your module.
> > > You can use a dialplan handler in your module (see mod_enum for
> example) to
> > > route inbound calls using your custom dialplan.
> > > You can use a state handler in your module and bind to channel states
> (much
> > > like binding to events).
> > > You can create a dialplan app in your module to execute code when the
> app is
> > > called in dialplan
> > > (Example: <application action="mycustom_module" data="${uuid} arg1
> arg2"/>)
> > > You can use an endpoint in your module to originate calls outbound (see
> > > mod_lcr or mod_callcenter for an example)
> > > Also, you can create an API for your module
> > >
> > > IMO creating a module is much more powerful than using a script with
> ESL.
> > > But if you are going to create a module, you really don't need to mess
> with
> > > events (unless they are very specific events like CUSTOM::) because
> your
> > > module has access to much of the freeswitch core.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> > > wrote:
> > >>
> > >> why at all do you need it to be a C module inside FreeSWITCH?
> > >>
> > >> Why not writing an ESL program which would subscribe to events and
> > >> perform the needed actions?
> > >>
> > >> How about the following scenario:
> > >>
> > >> 1. In the XML dialplan, you execute "park" application on the incoming
> > >> call.
> > >>
> > >> 2. Your program is listening to events via ESL, and it recognizes that
> > >> a channel has been parked
> > >>
> > >> 3. Your program starts to playback the ringback tone into that channel
> > >>
> > >> 4. Your program performs all the needed lookups and sets needed
> > >> variables on the channel
> > >>
> > >> 5. Your program transfers or bridges the call where needed.
> > >>
> > >> This is quite easy to implement in any programming language of your
> > >> choice, easy to debug, and it's easily scalable. It can be done in a
> > >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> > >> as many parallel calls as required.
> > >>
> > >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> > >> <jpablolorenzetti at hotmail.com> wrote:
> > >> > Hi, i m writing a module in C that needs to check for certain
> > >> > information in
> > >> > a
> > >> > database for the caller and the destination number,
> > >> > for this the module is subscribing to the CS_INIT channel events, so
> > >> > everytime a channel is created
> > >> > the module callback is called and it checks the numbers,
> > >> > the problem is that the callback gets called twice,
> > >> > for the creation of the a-leg of the call and the creation of the
> b-leg.
> > >> > Is there any way to accomplish what i m trying to do ?
> > >> > Am i doing it the wrong way?
> > >> > I have already try getting testing for the flags in the channel but
> it
> > >> > did
> > >> > not work,
> > >> > testing of originator or originating does not yield anything ....
> > >> >
> > >> > i might be doing it wrong maybe ?
> > >> >
> > >> > Thanks!
> > >> >
> > >> >
> > >> >
> > >> >
> _________________________________________________________________________
> > >> > Professional FreeSWITCH Consulting Services:
> > >> > consulting at freeswitch.org
> > >> > http://www.freeswitchsolutions.com
> > >> >
> > >> > Official FreeSWITCH Sites
> > >> > http://www.freeswitch.org
> > >> > http://confluence.freeswitch.org
> > >> > http://www.cluecon.com
> > >> >
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> > http://www.freeswitch.org
> > >>
> > >>
> _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://confluence.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/7b4a3378/attachment.html 

From ssinyagin at gmail.com  Thu Mar  5 17:37:49 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 5 Mar 2015 15:37:49 +0100
Subject: [Freeswitch-users] Debian package freeswitch-all
In-Reply-To: <CACOYK=hXBS4CwPu9ozsVgdXwJRkATuD-QuRacJYGyh6Lby8xRg@mail.gmail.com>
References: <CACOYK=hXBS4CwPu9ozsVgdXwJRkATuD-QuRacJYGyh6Lby8xRg@mail.gmail.com>
Message-ID: <CACOYK=gK-BbDXX1kaRH-2+GR2dGUY_ixFNug+gO0SvrrZq6Q2Q@mail.gmail.com>

oops, I just noted that opus is present, but speex is not available in 1.4 debs.

Is speex removed by intent?



On Thu, Mar 5, 2015 at 3:27 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> hi,
>
> I just noted that "freeswitch-all" package does not contain speex and
> opus. And of course "freeswitch-mod-speex" and  "freeswitch-mod-opus"
> conflict with it (I know that "freeswitch-meta-all" would solve the
> problem, but I like the single-package approach).
>
> Is it by intent or was it just missed out?
>
>
> thanks


From s.safarov at gmail.com  Thu Mar  5 18:00:45 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 5 Mar 2015 15:00:45 +0000
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAHtxdDcJuQZm2igC9QcS-jAcOAFAmbTFJgE3Xg9A3X6B_m1sEg@mail.gmail.com>

I want to receive events from the erlang module and for that execute the
following commands

[root at fs1 xml]# erl -sname test -setcookie ClueCon
Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10]
[hipe] [kernel-poll:false]

Eshell V6.2.1  (abort with ^G)
(test at fs1)1>  {foo, fs1 at fs1} ! {event, 'CHANNEL_CREATE'}, receive Y -> Y
after 1000 -> timeout end.
ok
(test at fs1)2>


And second way
[root at fs1 xml]# erl -sname test -setcookie ClueCon
Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10]
[hipe] [kernel-poll:false]

Eshell V6.2.1  (abort with ^G)
(test at fs1)1>  {foo, fs1 at fs1} ! {event, 'ALL'}.
{event,'ALL'}
(test at fs1)2> receive Y -> Y after 1000 -> timeout end.
ok
(test at fs1)3>

And I can not get events.
What am I doing wrong?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/49dfd772/attachment-0001.html 

From mishehu at freeswitch.org  Thu Mar  5 18:49:13 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 05 Mar 2015 09:49:13 -0600
Subject: [Freeswitch-users] Debian package freeswitch-all
In-Reply-To: <CACOYK=gK-BbDXX1kaRH-2+GR2dGUY_ixFNug+gO0SvrrZq6Q2Q@mail.gmail.com>
References: <CACOYK=hXBS4CwPu9ozsVgdXwJRkATuD-QuRacJYGyh6Lby8xRg@mail.gmail.com>
	<CACOYK=gK-BbDXX1kaRH-2+GR2dGUY_ixFNug+gO0SvrrZq6Q2Q@mail.gmail.com>
Message-ID: <54F87AF9.9000906@freeswitch.org>

Speex is not its own separate module at the current time, and is 
implemented in the core.

-- 
Yossi Neiman


On 03/05/2015 08:37 AM, Stanislav Sinyagin wrote:
> oops, I just noted that opus is present, but speex is not available in 1.4 debs.
>
> Is speex removed by intent?
>
>
>
> On Thu, Mar 5, 2015 at 3:27 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>> hi,
>>
>> I just noted that "freeswitch-all" package does not contain speex and
>> opus. And of course "freeswitch-mod-speex" and  "freeswitch-mod-opus"
>> conflict with it (I know that "freeswitch-meta-all" would solve the
>> problem, but I like the single-package approach).
>>
>> Is it by intent or was it just missed out?
>>
>>
>> thanks
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Mar  5 18:55:11 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 05 Mar 2015 09:55:11 -0600
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
 hangup cause?
In-Reply-To: <CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>
References: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>	<08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>
	<CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>
Message-ID: <54F87C5F.6070208@freeswitch.org>

You can but then you may push your packets over the MTU and this has the 
potential to cause problems.  I suggest you take a look in the SDP 
information in the CDR's or get an active trace or pcap on calls that 
are ending with INCOMPATIBLE_DESTINATION.  I believe the SIP response 
code that FS will send back in those cases is 488.  When you look at the 
SDP for the codecs requested by the remote and compare them to what 
FreeSWITCH is offering and then you can see what it is that you need to 
activate.

-- 
Yossi Neiman


On 03/05/2015 07:59 AM, Aqs Younas wrote:
> My freeswitch just answers the call and plays a mp3 file. Can i do 
> anything to make this minimum,? because Vendor is sending us calls 
> with most of them get through but some just hangup with this cause.
>
> Can i make my freeswitch to support maximum codecs?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/4234a362/attachment.html 

From ssinyagin at gmail.com  Thu Mar  5 19:15:35 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 5 Mar 2015 17:15:35 +0100
Subject: [Freeswitch-users] Debian package freeswitch-all
In-Reply-To: <54F87AF9.9000906@freeswitch.org>
References: <CACOYK=hXBS4CwPu9ozsVgdXwJRkATuD-QuRacJYGyh6Lby8xRg@mail.gmail.com>
	<CACOYK=gK-BbDXX1kaRH-2+GR2dGUY_ixFNug+gO0SvrrZq6Q2Q@mail.gmail.com>
	<54F87AF9.9000906@freeswitch.org>
Message-ID: <CACOYK=j=aeHDrm8RzDYS9qj_4ahhQUVLfnBvS_RQoYvaHTODVg@mail.gmail.com>

I see, thanks!

On Thu, Mar 5, 2015 at 4:49 PM, I put the Who? in Mishehu
<mishehu at freeswitch.org> wrote:
> Speex is not its own separate module at the current time, and is
> implemented in the core.
>
> --
> Yossi Neiman
>
>
> On 03/05/2015 08:37 AM, Stanislav Sinyagin wrote:
>> oops, I just noted that opus is present, but speex is not available in 1.4 debs.
>>
>> Is speex removed by intent?
>>
>>
>>
>> On Thu, Mar 5, 2015 at 3:27 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>> hi,
>>>
>>> I just noted that "freeswitch-all" package does not contain speex and
>>> opus. And of course "freeswitch-mod-speex" and  "freeswitch-mod-opus"
>>> conflict with it (I know that "freeswitch-meta-all" would solve the
>>> problem, but I like the single-package approach).
>>>
>>> Is it by intent or was it just missed out?
>>>
>>>
>>> thanks
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From grcamauer at gmail.com  Thu Mar  5 19:57:20 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 5 Mar 2015 13:57:20 -0300
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
 hangup cause?
In-Reply-To: <CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>
References: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>
	<08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>
	<CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>
Message-ID: <CAPNTnnrUcBjRUTUAACREGJ=WSxpfcufjS133-SP7AaXYZDv6gQ@mail.gmail.com>

You might also want to look into playing native files (save a copy of your
MP3s in G729, Speex, etc. so that FS doesn't have to translate each time.
See mod_native_file (
https://freeswitch.org/confluence/display/FREESWITCH/mod_native_file).


Guillermo

On Thu, Mar 5, 2015 at 10:59 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> My freeswitch just answers the call and plays a mp3 file. Can i do
> anything to make this minimum,? because Vendor is sending us calls with
> most of them get through but some just hangup with this cause.
>
> Can i make my freeswitch to support maximum codecs?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/e2c29083/attachment.html 

From zoell at zoell.us  Thu Mar  5 20:06:41 2015
From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=)
Date: Thu, 5 Mar 2015 17:06:41 +0000
Subject: [Freeswitch-users] Set channel variables before bridge leg B hangup
Message-ID: <CAJrD+vfD1++606CPE9+zWf7gUCTRD5K-Q-rPfLr-5GcdmAp=NQ@mail.gmail.com>

Hi,

In lua I bridge two sessions. When leg B hangup the call I need to set up
some custom channel variables for odbc_cdr reporting.

freeswitch.bridge(session1, session2);
session2:execute("set", "custom_var1=asdf");

But when the set command tries to run, the log says "channel is hangup
already".

Is there any way to do this properly?

Many thanks,
Zoltan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/d999a45f/attachment.html 

From vipkilla at gmail.com  Thu Mar  5 20:10:37 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 5 Mar 2015 12:10:37 -0500
Subject: [Freeswitch-users] Set channel variables before bridge leg B
	hangup
In-Reply-To: <CAJrD+vfD1++606CPE9+zWf7gUCTRD5K-Q-rPfLr-5GcdmAp=NQ@mail.gmail.com>
References: <CAJrD+vfD1++606CPE9+zWf7gUCTRD5K-Q-rPfLr-5GcdmAp=NQ@mail.gmail.com>
Message-ID: <CAC-LwPNmM+h5UgmSWMpJfOjseAq5+2zu0QEF3HYRPkKKyQO9aw@mail.gmail.com>

You could try using the api_on_hangup to set a variable.
or there maybe an execute_on_hangup too.

On Thu, Mar 5, 2015 at 12:06 PM, Zolt?n Szab? <zoell at zoell.us> wrote:

> Hi,
>
> In lua I bridge two sessions. When leg B hangup the call I need to set up
> some custom channel variables for odbc_cdr reporting.
>
> freeswitch.bridge(session1, session2);
> session2:execute("set", "custom_var1=asdf");
>
> But when the set command tries to run, the log says "channel is hangup
> already".
>
> Is there any way to do this properly?
>
> Many thanks,
> Zoltan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/60cfb9af/attachment-0001.html 

From hkalyoncu at gmail.com  Thu Mar  5 20:14:10 2015
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Thu, 5 Mar 2015 19:14:10 +0200
Subject: [Freeswitch-users] How can i create dynamic dialplan in
	freeswitch.
In-Reply-To: <CAGVOjVVWAgWVowMajPA1oFTcdRq2aqbUF1A4+_fjX_Fk3L+R=g@mail.gmail.com>
References: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>
	<CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>
	<CAKzWOxXa9M+Q+1wA-JVjrmd5_xXPUd3YggOMp+KOdF8BExFLTg@mail.gmail.com>
	<CAGVOjVVWAgWVowMajPA1oFTcdRq2aqbUF1A4+_fjX_Fk3L+R=g@mail.gmail.com>
Message-ID: <CAPf+86V=eU_dQALm7K5+0uQGm33iypLiD7ucTfiFY7FXt=tqsA@mail.gmail.com>

another approach would be sticking with static xml dialplan and you can
dynamically generate and update it

On Thu, Mar 5, 2015 at 3:28 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks for your reply. I see mod_xml_curl using apache and so does
> mod_httapi. An extensive back and forth switching would make a lot of load
> on apache. So, i am creating something with mod_lua.
>
> On 2 March 2015 at 23:42, Michael Collins <msc at freeswitch.org> wrote:
>
>> See also chapter 9 of the FreeSWITCH 1.2 book, appropriately entitled,
>> "Moving Beyond the Static XML Configuration."
>>
>> -MC
>>
>> On Mon, Mar 2, 2015 at 9:24 AM, Vik Killa <vipkilla at gmail.com> wrote:
>>
>>> Hi,
>>> Look at mod_xml_curl to do a 'dynamic' dialplan.
>>> Thanks.
>>>
>>> On Mon, Mar 2, 2015 at 12:21 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>>
>>>> Hi, user.
>>>>
>>>> After working for more than 3 months while writing my dialplan in
>>>> static xml file,but now wants to know how can i effectively create dynamic
>>>> dialplan in freeswitch.
>>>>
>>>> Thanks.
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/8d66b591/attachment.html 

From s.safarov at gmail.com  Thu Mar  5 20:21:23 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 5 Mar 2015 17:21:23 +0000
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAHtxdDcJuQZm2igC9QcS-jAcOAFAmbTFJgE3Xg9A3X6B_m1sEg@mail.gmail.com>
References: <CAHtxdDcJuQZm2igC9QcS-jAcOAFAmbTFJgE3Xg9A3X6B_m1sEg@mail.gmail.com>
Message-ID: <CAHtxdDecmapRh-bCP=OgFnehvhtJ-F0xeg9szjd+_x+LjOfC0Q@mail.gmail.com>

In the console I get errors
2015-03-05 17:03:18.232604 [ERR] ei_helpers.c:274 Invalid process type!

What is the process uset there?

On Thu, Mar 5, 2015 at 3:00 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> I want to receive events from the erlang module and for that execute the
> following commands
>
> [root at fs1 xml]# erl -sname test -setcookie ClueCon
> Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10]
> [hipe] [kernel-poll:false]
>
> Eshell V6.2.1  (abort with ^G)
> (test at fs1)1>  {foo, fs1 at fs1} ! {event, 'CHANNEL_CREATE'}, receive Y -> Y
> after 1000 -> timeout end.
> ok
> (test at fs1)2>
>
>
> And second way
> [root at fs1 xml]# erl -sname test -setcookie ClueCon
> Erlang/OTP 17 [erts-6.2.1] [source] [64-bit] [smp:2:2] [async-threads:10]
> [hipe] [kernel-poll:false]
>
> Eshell V6.2.1  (abort with ^G)
> (test at fs1)1>  {foo, fs1 at fs1} ! {event, 'ALL'}.
> {event,'ALL'}
> (test at fs1)2> receive Y -> Y after 1000 -> timeout end.
> ok
> (test at fs1)3>
>
> And I can not get events.
> What am I doing wrong?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/754b21a3/attachment.html 

From mishehu at freeswitch.org  Thu Mar  5 20:21:37 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 05 Mar 2015 11:21:37 -0600
Subject: [Freeswitch-users] What does cause 'INCOMPATIBLE_DESTINATION' a
 hangup cause?
In-Reply-To: <CAPNTnnrUcBjRUTUAACREGJ=WSxpfcufjS133-SP7AaXYZDv6gQ@mail.gmail.com>
References: <CAGVOjVX+Ke4poYbi=Ld-k7ufNp65htKAEeVxMxBKZQKWarrX5g@mail.gmail.com>	<08FA92EB-A951-4CB7-92DB-10094B424CA8@gmail.com>	<CAGVOjVXeRQrbnadvqE7Msr6XK8GGmG+FokGgtJw3PNJVXr0NOA@mail.gmail.com>
	<CAPNTnnrUcBjRUTUAACREGJ=WSxpfcufjS133-SP7AaXYZDv6gQ@mail.gmail.com>
Message-ID: <54F890A1.7030603@freeswitch.org>

In order to not confuse the original poster, what you are describing is 
completely separate from the issue that is being experienced.  It should 
also be noted that not all codecs will support direct injection by 
mod_native_file also.

-- 
Yossi Neiman


On 03/05/2015 10:57 AM, Guillermo Ruiz Camauer wrote:
> You might also want to look into playing native files (save a copy of 
> your MP3s in G729, Speex, etc. so that FS doesn't have to translate 
> each time.  See mod_native_file 
> (https://freeswitch.org/confluence/display/FREESWITCH/mod_native_file).
>
>
> Guillermo
>
> On Thu, Mar 5, 2015 at 10:59 AM, Aqs Younas <aqsyounas at gmail.com 
> <mailto:aqsyounas at gmail.com>> wrote:
>
>     My freeswitch just answers the call and plays a mp3 file. Can i do
>     anything to make this minimum,? because Vendor is sending us calls
>     with most of them get through but some just hangup with this cause.
>
>     Can i make my freeswitch to support maximum codecs?
>
>     _________________________________________________________________________
>     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://confluence.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
>
>
>
>
> -- 
> Guillermo Ruiz Camauer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/392c2819/attachment-0001.html 

From mishehu at freeswitch.org  Thu Mar  5 20:24:10 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 05 Mar 2015 11:24:10 -0600
Subject: [Freeswitch-users] How can i create dynamic dialplan in
	freeswitch.
In-Reply-To: <CAPf+86V=eU_dQALm7K5+0uQGm33iypLiD7ucTfiFY7FXt=tqsA@mail.gmail.com>
References: <CAGVOjVWNb7JWZV=kxHR7vekOdYam=Ep88H=EOs=2GS+Cmn7XsA@mail.gmail.com>	<CAC-LwPMEkuC8mnt-FWPpnDadMbcngiiJgfeptKboOnE+ZOTHvQ@mail.gmail.com>	<CAKzWOxXa9M+Q+1wA-JVjrmd5_xXPUd3YggOMp+KOdF8BExFLTg@mail.gmail.com>	<CAGVOjVVWAgWVowMajPA1oFTcdRq2aqbUF1A4+_fjX_Fk3L+R=g@mail.gmail.com>
	<CAPf+86V=eU_dQALm7K5+0uQGm33iypLiD7ucTfiFY7FXt=tqsA@mail.gmail.com>
Message-ID: <54F8913A.4040809@freeswitch.org>

This is only practical if you make changes once in a long while.  If 
your call handling is dependent on conditions that are unique to every 
call, this option will not be viable.

-- 
Yossi Neiman


On 03/05/2015 11:14 AM, huseyin kalyoncu wrote:
> another approach would be sticking with static xml dialplan and you 
> can dynamically generate and update it
>
> On Thu, Mar 5, 2015 at 3:28 PM, Aqs Younas <aqsyounas at gmail.com 
> <mailto:aqsyounas at gmail.com>> wrote:
>
>     Thanks for your reply. I see mod_xml_curl using apache and so does
>     mod_httapi. An extensive back and forth switching would make a lot
>     of load on apache. So, i am creating something with mod_lua.
>
>     On 2 March 2015 at 23:42, Michael Collins <msc at freeswitch.org
>     <mailto:msc at freeswitch.org>> wrote:
>
>         See also chapter 9 of the FreeSWITCH 1.2 book, appropriately
>         entitled, "Moving Beyond the Static XML Configuration."
>
>         -MC
>
>         On Mon, Mar 2, 2015 at 9:24 AM, Vik Killa <vipkilla at gmail.com
>         <mailto:vipkilla at gmail.com>> wrote:
>
>             Hi,
>             Look at mod_xml_curl to do a 'dynamic' dialplan.
>             Thanks.
>
>             On Mon, Mar 2, 2015 at 12:21 PM, Aqs Younas
>             <aqsyounas at gmail.com <mailto:aqsyounas at gmail.com>> wrote:
>
>                 Hi, user.
>
>                 After working for more than 3 months while writing my
>                 dialplan in static xml file,but now wants to know how
>                 can i effectively create dynamic dialplan in freeswitch.
>
>                 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://confluence.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://confluence.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://confluence.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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/c0eda509/attachment.html 

From luis.daniel.lucio at gmail.com  Thu Mar  5 01:30:25 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 4 Mar 2015 17:30:25 -0500
Subject: [Freeswitch-users] Real-time billing application for the
 FreeSWITCH (mod_lua, mod_perl or ESL)
In-Reply-To: <1425507467266-7596150.post@n2.nabble.com>
References: <002101ce952e$6e3fc3f0$4abf4bd0$@verizon.net>
	<1425507467266-7596150.post@n2.nabble.com>
Message-ID: <CAFLo2QzSswbR4ZQ7O2as_Dan6efenfemYRG6NqFdEWp0b8=d9A@mail.gmail.com>

There are many ways to do the billing. In my case, I use mod_nibblebill
only to monitor to cut the call if they run out of credit and I interact
with mod_xml_cdr to bill when the call has just finished.

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

2015-03-04 17:17 GMT-05:00 jorgemariodlc <jorgemariodlc at gmail.com>:

> I actually working in it, I found it yesterday you need to add those lines
> (/autoload_configs/lua.conf.xml):
>
> <configuration name="lua.conf" description="LUA Configuration">
>   <settings>
>
>     <hook event="CHANNEL_ANSWER" script="answer.lua"/>
>     <hook event="CHANNEL_HANGUP_COMPLETE" script="hangup.lua"/>
>
>   </settings>
> </configuration>
>
> Check this link to know what information is in each event, because it's
> important to handle the direction-call (Outbound, Inbound)
> https://wiki.freeswitch.org/wiki/Event_List
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Real-time-billing-application-for-the-FreeSWITCH-mod-lua-mod-perl-or-ESL-tp7593788p7596150.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150304/232f513c/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Thu Mar  5 11:24:39 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Thu, 5 Mar 2015 13:54:39 +0530
Subject: [Freeswitch-users] Flooded with Stun Errors
Message-ID: <C4DE512B-31F0-478D-BAC7-43BE2299D1BA@gmail.com>

Hi,

I am getting flood of following messages.

2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00
2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid syncing 1 audio packet(s)
2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003 at df7jal23ls0d.invalid audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00
2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00
2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid syncing 1 audio packet(s)
2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00
2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid syncing 1 audio packet(s)
2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID

Requesting an insight what could have been causing this.

Regards,

Naveen Khanna
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/59366777/attachment.html 

From osblinnikov at gmail.com  Thu Mar  5 11:51:56 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Thu, 5 Mar 2015 09:51:56 +0100
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
Message-ID: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>

Hi,

I've made a simple Android Java application utilizing JAIN SIP,
webrtc.org android
library and connected to FreeSwitch via UDP.

But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client
this SDP is not managed well by FreeSwitch for establishment WebRTC
PeerConnection.

When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);`
in my Android Application with the SDP from FreeSwitch I get:

"onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS
fingerprint."

At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/)
and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure
audio flow.

Here is initial SDP from Chrome (http://tryit.jssip.net/):

v=0
o=- 6887715720880489867 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.122.1
a=rtcp:38359 IN IP4 192.168.122.1
a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
generation 0
.......
a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
active generation 0
a=ice-ufrag:bwrCv9yS8rCY12Az
a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
a=ice-options:google-ice
a=*fingerprint*:sha-256
52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
4e8f212e-746a-47bb-bc62-4a42d4e9e84e
a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
m=video 38359 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.122.1
a=rtcp:38359 IN IP4 192.168.122.1
a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
generation 0
............
a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
active generation 0
a=ice-ufrag:bwrCv9yS8rCY12Az
a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
a=ice-options:google-ice
a=*fingerprint*:sha-256
52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100


Here is SDP received from FreeSwitch in JAIN SIP via UDP:

v=0
o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
s=FreeSWITCH
c=IN IP4 192.168.131.253
t=0 0
m=audio 16390 RTP/AVP 111 0 8 101 13
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 16388 RTP/AVP 100
a=rtpmap:100 VP8/90000


I suppose that FreeSwitch wants to see WebRTC connection only on the
WebSocket ports and it doesn't know that my UDP client is actually WebRTC
client.

So I'm wondering if it possible to connect SIP client to the WebSocket port
via TCP using standard SIP client and never upgrade connection to WebSocket?

Regards,
Oleg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/5935ea23/attachment.html 

From mike at jerris.com  Thu Mar  5 20:46:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2015 12:46:28 -0500
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl> <, >
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
	<,
	> <CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>
	<,
	> <CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>
	<DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>
Message-ID: <60002D19-0F29-4BCE-9DA3-9726D952B45F@jerris.com>

Given the similarity in purpose, I would look closely at how mod_nibblebill interfaces with freeswitch.  It sounds like your interface needs are nearly identical.


> On Mar 5, 2015, at 9:23 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:
> 
> Thank you very much guys for your contributions.
> 
> I m doing it as a module for couple of reasons, being the
> most important (i believe) performance, because the module
> i m working on is to do real time charging of voice calls on a switch
> that is already  serving as a RBT service plus a bunch of IVR's to purchase
> services, this is for a ~150K user base on a single machine (cold standby)
> this switch is also scheduled to soon start providing hosted PBX services, 
> so going the script direction
> i personally dont see that as an option at all. I do use scripts for small no so much used
> much simpler stuff though, e.g: a lua script takes care of authenticating users
> when doing international calls from company extensions in the hosted PBX solution.
> 
> The other reason i chose to do
> this as a module because C is the language i feel more comfortable with. 
> i hope this clarifies i little bit this.
> 
> Moving on, right now i m developing on a test freeswitch that we have and yes i noticed
> that subscribing to the CS_INIT event does represent a big problem
> because i get notified for every single of those events that is generated on
> freeswitch which would be very inconvenient because as i mentioned, the same
> switch does many other things that i m not interested in, so i m going to try the advise
> provided and try to do it in the dial plan, i will explore this option. 
> 
> thank you very much all!
> 
> 
> 
> 
> 
> > Date: Thu, 5 Mar 2015 14:19:51 +0100
> > From: ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>
> > To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] sessions and CS_INIT events
> > 
> > but for the task that OP has described, writing (and maintaining it in
> > the long term) a module is really an overkill. Plus, he would also
> > need to take care of multithreading within FreeSWITCH, as well as
> > memory management, etc.
> > 
> > Also, a module makes sense if it's some common task which can be
> > re-used by others and published as open source. If it's some
> > closed-source module for a specific enterprise task that Juan has, it
> > just doesn't make sense and too much risk for a long-term solution.
> > 
> > 
> > 
> > 
> > 
> > On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > > It all depends on what you are trying to do with your module.
> > > You can use a dialplan handler in your module (see mod_enum for example) to
> > > route inbound calls using your custom dialplan.
> > > You can use a state handler in your module and bind to channel states (much
> > > like binding to events).
> > > You can create a dialplan app in your module to execute code when the app is
> > > called in dialplan
> > > (Example: <application action="mycustom_module" data="${uuid} arg1 arg2"/>)
> > > You can use an endpoint in your module to originate calls outbound (see
> > > mod_lcr or mod_callcenter for an example)
> > > Also, you can create an API for your module
> > >
> > > IMO creating a module is much more powerful than using a script with ESL.
> > > But if you are going to create a module, you really don't need to mess with
> > > events (unless they are very specific events like CUSTOM::) because your
> > > module has access to much of the freeswitch core.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> > > wrote:
> > >>
> > >> why at all do you need it to be a C module inside FreeSWITCH?
> > >>
> > >> Why not writing an ESL program which would subscribe to events and
> > >> perform the needed actions?
> > >>
> > >> How about the following scenario:
> > >>
> > >> 1. In the XML dialplan, you execute "park" application on the incoming
> > >> call.
> > >>
> > >> 2. Your program is listening to events via ESL, and it recognizes that
> > >> a channel has been parked
> > >>
> > >> 3. Your program starts to playback the ringback tone into that channel
> > >>
> > >> 4. Your program performs all the needed lookups and sets needed
> > >> variables on the channel
> > >>
> > >> 5. Your program transfers or bridges the call where needed.
> > >>
> > >> This is quite easy to implement in any programming language of your
> > >> choice, easy to debug, and it's easily scalable. It can be done in a
> > >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> > >> as many parallel calls as required.
> > >>
> > >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> > >> <jpablolorenzetti at hotmail.com> wrote:
> > >> > Hi, i m writing a module in C that needs to check for certain
> > >> > information in
> > >> > a
> > >> > database for the caller and the destination number,
> > >> > for this the module is subscribing to the CS_INIT channel events, so
> > >> > everytime a channel is created
> > >> > the module callback is called and it checks the numbers,
> > >> > the problem is that the callback gets called twice,
> > >> > for the creation of the a-leg of the call and the creation of the b-leg.
> > >> > Is there any way to accomplish what i m trying to do ?
> > >> > Am i doing it the wrong way?
> > >> > I have already try getting testing for the flags in the channel but it
> > >> > did
> > >> > not work,
> > >> > testing of originator or originating does not yield anything ....
> > >> >
> > >> > i might be doing it wrong maybe ?
> > >> >
> > >> > Thanks!
> > >> >
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/15e54de4/attachment-0001.html 

From danb.lists at gmail.com  Thu Mar  5 20:48:38 2015
From: danb.lists at gmail.com (DanB)
Date: Thu, 05 Mar 2015 18:48:38 +0100
Subject: [Freeswitch-users] Real-time billing application for the,
 FreeSWITCH (mod_lua, mod_perl or ESL)
In-Reply-To: <mailman.6726.1425576982.581.freeswitch-users@lists.freeswitch.org>
References: <mailman.6726.1425576982.581.freeswitch-users@lists.freeswitch.org>
Message-ID: <54F896F6.1020602@gmail.com>

One more thing to consider when you build your real-time billing 
application is call authorize before connect.
If you only treat answer and hangup you can end up with the call without 
balance going through and eating some seconds at each connect (which in 
case of DoS requests can result in quite serious amounts for you).

In CGRateS we use park application in dialplan to put the call on hold 
before being authorized by the same CHANNEL_PARK event and loop the call 
through dialplan back when we are done checking it.

Just my two cents,
DanB

On 05.03.2015 18:36, freeswitch-users-request at lists.freeswitch.org wrote:
> 015-03-04 17:17 GMT-05:00 jorgemariodlc<jorgemariodlc at gmail.com>:
>
>> >I actually working in it, I found it yesterday you need to add those lines
>> >(/autoload_configs/lua.conf.xml):
>> >
>> ><configuration name="lua.conf" description="LUA Configuration">
>> >   <settings>
>> >
>> >     <hook event="CHANNEL_ANSWER" script="answer.lua"/>
>> >     <hook event="CHANNEL_HANGUP_COMPLETE" script="hangup.lua"/>
>> >
>> >   </settings>
>> ></configuration>
>> >
>> >Check this link to know what information is in each event, because it's
>> >important to handle the direction-call (Outbound, Inbound)
>> >https://wiki.freeswitch.org/wiki/Event_List



From mike at jerris.com  Thu Mar  5 20:51:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2015 12:51:30 -0500
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
Message-ID: <0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>

you need to tell freeswitch to send a webrtc compatible SDP.

https://wiki.freeswitch.org/wiki/Variable_media_webrtc <https://wiki.freeswitch.org/wiki/Variable_media_webrtc>


> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> Hi,
> 
> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org <http://webrtc.org/> android library and connected to FreeSwitch via UDP.
> 
> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection.
> 
> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:
> 
> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
> 
> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/ <http://tryit.jssip.net/>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.
> 
> Here is initial SDP from Chrome (http://tryit.jssip.net/ <http://tryit.jssip.net/>):
> 
> v=0
> o=- 6887715720880489867 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
> .......
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> m=video 38359 RTP/SAVPF 100 116 117 96
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
> ............
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
> a=setup:actpass
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> a=recvonly
> a=rtcp-mux
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 nack pli
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> 
> 
> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
> 
> v=0
> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
> s=FreeSWITCH
> c=IN IP4 192.168.131.253
> t=0 0
> m=audio 16390 RTP/AVP 111 0 8 101 13
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> m=video 16388 RTP/AVP 100
> a=rtpmap:100 VP8/90000
> 
> 
> I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. 
> 
> So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket?
> 
> Regards,
> Oleg
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/8004ccf9/attachment.html 

From mike at jerris.com  Thu Mar  5 20:56:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2015 12:56:53 -0500
Subject: [Freeswitch-users] Flooded with Stun Errors
In-Reply-To: <C4DE512B-31F0-478D-BAC7-43BE2299D1BA@gmail.com>
References: <C4DE512B-31F0-478D-BAC7-43BE2299D1BA@gmail.com>
Message-ID: <BEB71057-02BC-4A6B-8AD0-524039DA0542@jerris.com>

these are extra debug you get when you set debug-level in switch.conf or when using "fsctl debug_level" api command.

> On Mar 5, 2015, at 3:24 AM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> Hi,
> 
> I am getting flood of following messages.
> 
> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
> 2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00
> 2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
> 2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003 at df7jal23ls0d.invalid <mailto:2003 at df7jal23ls0d.invalid> audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00
> 2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00
> 2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
> 2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00
> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
> 2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
> 
> Requesting an insight what could have been causing this.
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/4e705b9d/attachment-0001.html 

From jpablolorenzetti at hotmail.com  Thu Mar  5 21:05:51 2015
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Thu, 5 Mar 2015 18:05:51 +0000
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <60002D19-0F29-4BCE-9DA3-9726D952B45F@jerris.com>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl> <,
	>, <CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>,
	<, , >
	<CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>, <,
	, >
	<CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>,
	<DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>,
	<60002D19-0F29-4BCE-9DA3-9726D952B45F@jerris.com>
Message-ID: <DUB130-W42AFF5EB5971D0AF2A9198D91F0@phx.gbl>

Hi, yes i had a look at it, and yes the needs are similar, i used it at the beginning
to get started, and i m using as a reference at this point but it seems that the use cases
are different . thanks!



From: mike at jerris.com
Date: Thu, 5 Mar 2015 12:46:28 -0500
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sessions and CS_INIT events

Given the similarity in purpose, I would look closely at how mod_nibblebill interfaces with freeswitch.  It sounds like your interface needs are nearly identical.

On Mar 5, 2015, at 9:23 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com> wrote:Thank you very much guys for your contributions.

I m doing it as a module for couple of reasons, being the
most important (i believe) performance, because the module
i m working on is to do real time charging of voice calls on a switch
that is already  serving as a RBT service plus a bunch of IVR's to purchase
services, this is for a ~150K user base on a single machine (cold standby)
this switch is also scheduled to soon start providing hosted PBX services, 
so going the script direction
i personally dont see that as an option at all. I do use scripts for small no so much used
much simpler stuff though, e.g: a lua script takes care of authenticating users
when doing international calls from company extensions in the hosted PBX solution.

The other reason i chose to do
this as a module because C is the language i feel more comfortable with. 
i hope this clarifies i little bit this.

Moving on, right now i m developing on a test freeswitch that we have and yes i noticed
that subscribing to the CS_INIT event does represent a big problem
because i get notified for every single of those events that is generated on
freeswitch which would be very inconvenient because as i mentioned, the same
switch does many other things that i m not interested in, so i m going to try the advise
provided and try to do it in the dial plan, i will explore this option. 

thank you very much all!





> Date: Thu, 5 Mar 2015 14:19:51 +0100
> From: ssinyagin at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sessions and CS_INIT events
> 
> but for the task that OP has described, writing (and maintaining it in
> the long term) a module is really an overkill. Plus, he would also
> need to take care of multithreading within FreeSWITCH, as well as
> memory management, etc.
> 
> Also, a module makes sense if it's some common task which can be
> re-used by others and published as open source. If it's some
> closed-source module for a specific enterprise task that Juan has, it
> just doesn't make sense and too much risk for a long-term solution.
> 
> 
> 
> 
> 
> On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > It all depends on what you are trying to do with your module.
> > You can use a dialplan handler in your module (see mod_enum for example) to
> > route inbound calls using your custom dialplan.
> > You can use a state handler in your module and bind to channel states (much
> > like binding to events).
> > You can create a dialplan app in your module to execute code when the app is
> > called in dialplan
> > (Example: <application action="mycustom_module" data="${uuid} arg1 arg2"/>)
> > You can use an endpoint in your module to originate calls outbound (see
> > mod_lcr or mod_callcenter for an example)
> > Also, you can create an API for your module
> >
> > IMO creating a module is much more powerful than using a script with ESL.
> > But if you are going to create a module, you really don't need to mess with
> > events (unless they are very specific events like CUSTOM::) because your
> > module has access to much of the freeswitch core.
> >
> > Thanks.
> >
> >
> >
> > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> > wrote:
> >>
> >> why at all do you need it to be a C module inside FreeSWITCH?
> >>
> >> Why not writing an ESL program which would subscribe to events and
> >> perform the needed actions?
> >>
> >> How about the following scenario:
> >>
> >> 1. In the XML dialplan, you execute "park" application on the incoming
> >> call.
> >>
> >> 2. Your program is listening to events via ESL, and it recognizes that
> >> a channel has been parked
> >>
> >> 3. Your program starts to playback the ringback tone into that channel
> >>
> >> 4. Your program performs all the needed lookups and sets needed
> >> variables on the channel
> >>
> >> 5. Your program transfers or bridges the call where needed.
> >>
> >> This is quite easy to implement in any programming language of your
> >> choice, easy to debug, and it's easily scalable. It can be done in a
> >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> >> as many parallel calls as required.
> >>
> >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> >> <jpablolorenzetti at hotmail.com> wrote:
> >> > Hi, i m writing a module in C that needs to check for certain
> >> > information in
> >> > a
> >> > database for the caller and the destination number,
> >> > for this the module is subscribing to the CS_INIT channel events, so
> >> > everytime a channel is created
> >> > the module callback is called and it checks the numbers,
> >> > the problem is that the callback gets called twice,
> >> > for the creation of the a-leg of the call and the creation of the b-leg.
> >> > Is there any way to accomplish what i m trying to do ?
> >> > Am i doing it the wrong way?
> >> > I have already try getting testing for the flags in the channel but it
> >> > did
> >> > not work,
> >> > testing of originator or originating does not yield anything ....
> >> >
> >> > i might be doing it wrong maybe ?
> >> >
> >> > Thanks!
> >> >
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/110dd885/attachment.html 

From tfred31 at yahoo.com  Fri Mar  6 01:05:40 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Thu, 5 Mar 2015 14:05:40 -0800
Subject: [Freeswitch-users] Conference Announce Count Inline - Not working
Message-ID: <1425593140.19093.YahooMailBasic@web160201.mail.bf1.yahoo.com>

I am following the instructions on:
   "This example is a very quick and dirty dialplan and conference config that lets you hear how many callers are in conference."
       https://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline

Within my existing    conf\autoload_configs\conference.conf.xml   I added the new  caller controls:   <group name="plain">  as shown in the referenced page.

Additionally within the same file I edited the   <profile name="default">   to use the new:   <param name="caller-controls" value="plain"/> 


I then created a new XML file:    conf/dialplan/default/01_Announce_Conf_Count.xml   containing the code shown in the referenced page.

Lastly I edited this file to change the last 'application' line beginning with:    application="say"  to the following
      <action application="phrase" data="callers_in_conf,${conf_count}"/>    
so as to play the existing macro phrase which exists in:   conf\lang\en\ivr\sounds.xml    
It should be playing existing wav files and, therefore not need, TTS


But when I enter the conference..
1. The first caller in gets a 'voice' message played indicating:   "You are currently the only person in this conference"
2. But subsequent callers get nothing but a tone upon entry into the conference.   The intended macro phrase is not playing.

What needs to change to make the 'Announce Conference Participant Count"  work.

Any assistance would be greatly appreciated.

Thanks






From ing.antonyam at gmail.com  Fri Mar  6 02:58:30 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Thu, 5 Mar 2015 17:58:30 -0600
Subject: [Freeswitch-users] freeswitch support
Message-ID: <CABFbAiVDwMZpc2z+y1ynpZVhfseX1hC+jEM-5-JL7RG4L9ZU=Q@mail.gmail.com>

I have configured my fs with xml_curl module in the directory part, may
serve 100,000 extensions from a single FS or is there any limitation as to
the software?

IN the hardware I have a server with the following specifications:

Brand: DELL
RAM: 4GB
CPU: Intel (R) Xeon (R) CPU E5405 @ 2.00GHz
cores: 4
DD: 50GB

regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150305/e5ba2695/attachment.html 

From zoell at zoell.us  Fri Mar  6 11:53:47 2015
From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=)
Date: Fri, 6 Mar 2015 08:53:47 +0000
Subject: [Freeswitch-users] Set channel variables before bridge leg B
	hangup
In-Reply-To: <CAC-LwPNmM+h5UgmSWMpJfOjseAq5+2zu0QEF3HYRPkKKyQO9aw@mail.gmail.com>
References: <CAJrD+vfD1++606CPE9+zWf7gUCTRD5K-Q-rPfLr-5GcdmAp=NQ@mail.gmail.com>
	<CAC-LwPNmM+h5UgmSWMpJfOjseAq5+2zu0QEF3HYRPkKKyQO9aw@mail.gmail.com>
Message-ID: <CAJrD+vcvPGNepvDSNCffE5Mxw4Lx4+ZxfxJEPksUdMktrpW9AA@mail.gmail.com>

How can I reference the session variable in the hook lua script?

Thank you

2015-03-05 17:10 GMT+00:00 Vik Killa <vipkilla at gmail.com>:

> You could try using the api_on_hangup to set a variable.
> or there maybe an execute_on_hangup too.
>
> On Thu, Mar 5, 2015 at 12:06 PM, Zolt?n Szab? <zoell at zoell.us> wrote:
>
>> Hi,
>>
>> In lua I bridge two sessions. When leg B hangup the call I need to set up
>> some custom channel variables for odbc_cdr reporting.
>>
>> freeswitch.bridge(session1, session2);
>> session2:execute("set", "custom_var1=asdf");
>>
>> But when the set command tries to run, the log says "channel is hangup
>> already".
>>
>> Is there any way to do this properly?
>>
>> Many thanks,
>> Zoltan
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/346b314b/attachment-0001.html 

From naveen.khanna.bm at gmail.com  Fri Mar  6 12:02:27 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Fri, 6 Mar 2015 14:32:27 +0530
Subject: [Freeswitch-users] Flooded with Stun Errors
In-Reply-To: <BEB71057-02BC-4A6B-8AD0-524039DA0542@jerris.com>
References: <C4DE512B-31F0-478D-BAC7-43BE2299D1BA@gmail.com>
	<BEB71057-02BC-4A6B-8AD0-524039DA0542@jerris.com>
Message-ID: <4D1EE950-323D-4F14-967C-7E5EE7FB2B37@gmail.com>

Thanks.

Should I consider this as normal? Is there any thing which can be done so that these alerts do not occur? An inside on on the reason will help me curb these alerts.

Regards,

Naveen Khanna


> On 05-Mar-2015, at 11:26 pm, Michael Jerris <mike at jerris.com> wrote:
> 
> these are extra debug you get when you set debug-level in switch.conf or when using "fsctl debug_level" api command.
> 
>> On Mar 5, 2015, at 3:24 AM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>> 
>> Hi,
>> 
>> I am getting flood of following messages.
>> 
>> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>> 2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00
>> 2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>> 2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003 at df7jal23ls0d.invalid <mailto:2003 at df7jal23ls0d.invalid> audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00
>> 2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00
>> 2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>> 2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00
>> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>> 2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>> 
>> Requesting an insight what could have been causing this.
>> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/61a693e4/attachment.html 

From naveen.khanna.bm at gmail.com  Fri Mar  6 12:33:54 2015
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Fri, 6 Mar 2015 15:03:54 +0530
Subject: [Freeswitch-users] Stale Channels build up
Message-ID: <8D471D2B-CF39-4071-BF34-ECE55F736F4B@gmail.com>

Hi,

I am getting Stale Channels build up in my Freeswitch installation.

We have a deployed a Dialer solution which is based on mod_callcenter, and a continuous stream of call are generated using Originate api command. As time progress the hung channels build increase and moment it reaches a threshold of 420+ of stale channels the entire site goes down. The Sip connections / extensions get disconnect and there is no means that they can get connected again.

However, switch continues receive incoming calls, but extensions / agents sitting behind mod_callcenter are unable to answer calls. The only solution that I have is to restart the switch.

I am using : Freeswitch 1.4 enabled with WebRTC support, SipmML5 based Sip Client and postgres as database. The solution is deployed on CentOS 6.6 64 bit.


Regards,

Naveen Khanna
 

From osblinnikov at gmail.com  Fri Mar  6 13:05:54 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Fri, 6 Mar 2015 11:05:54 +0100
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
Message-ID: <CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>

thank you very much Michael, it magically works.

On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com> wrote:

> you need to tell freeswitch to send a webrtc compatible SDP.
>
> https://wiki.freeswitch.org/wiki/Variable_media_webrtc
>
>
> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
>
> Hi,
>
> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org android
> library and connected to FreeSwitch via UDP.
>
> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client
> this SDP is not managed well by FreeSwitch for establishment WebRTC
> PeerConnection.
>
> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);`
> in my Android Application with the SDP from FreeSwitch I get:
>
> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS
> fingerprint."
>
> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/)
> and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure
> audio flow.
>
> Here is initial SDP from Chrome (http://tryit.jssip.net/):
>
> v=0
> o=- 6887715720880489867 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
> generation 0
> .......
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
> active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=*fingerprint*:sha-256
> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
> m=video 38359 RTP/SAVPF 100 116 117 96
> c=IN IP4 192.168.122.1
> a=rtcp:38359 IN IP4 192.168.122.1
> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
> generation 0
> ............
> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
> active generation 0
> a=ice-ufrag:bwrCv9yS8rCY12Az
> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
> a=ice-options:google-ice
> a=*fingerprint*:sha-256
> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
> a=setup:actpass
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=recvonly
> a=rtcp-mux
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 nack pli
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
>
>
> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>
> v=0
> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
> s=FreeSWITCH
> c=IN IP4 192.168.131.253
> t=0 0
> m=audio 16390 RTP/AVP 111 0 8 101 13
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> m=video 16388 RTP/AVP 100
> a=rtpmap:100 VP8/90000
>
>
> I suppose that FreeSwitch wants to see WebRTC connection only on the
> WebSocket ports and it doesn't know that my UDP client is actually WebRTC
> client.
>
> So I'm wondering if it possible to connect SIP client to the WebSocket
> port via TCP using standard SIP client and never upgrade connection to
> WebSocket?
>
> Regards,
> Oleg
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Oleg Blinnikov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/bdf9d486/attachment-0001.html 

From aqsyounas at gmail.com  Fri Mar  6 14:42:27 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 6 Mar 2015 16:42:27 +0500
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is malformed]
Message-ID: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>

Hi, users.

After power failure on my server, now when I start my freeswitch  I see
these errors logs on my freeswitch.

state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='18572166595',dialplan='XML',context='public',presence_id='',presence_data=''
where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set state='CS_EXECUTE' where
uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set application='sched_hangup',application_data='+10800
alloted_timeout',presence_id='',presence_data='' where
uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set
application='answer',application_data='',presence_id='',presence_data=''
where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set
read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set
read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[database disk image is malformed]
update channels set
read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'

How can I resolve this on my server?

I was using mysql database to dump my cdr using mod_json_cdr.

Thanks for you help.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/87f52c41/attachment.html 

From nbhatti at gmail.com  Fri Mar  6 14:54:47 2015
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Fri, 6 Mar 2015 14:54:47 +0300
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is
 malformed]
In-Reply-To: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>
References: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>
Message-ID: <etPan.54f9958a.625558ec.36e@BusyBox.local>


You seem to be using SQLite for core db. Remove the db files from your FreeSWITCH install location /db and restart FreeSWITCH. ?

?
Thanks,
Muhammad Naseer Bhatti


From:?Aqs Younas <aqsyounas at gmail.com>
Reply:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
Date:?March 6, 2015 at 2:43:22 PM
To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
Subject:? [Freeswitch-users] NATIVE SQL ERR [database disk image is malformed]  

Hi, users.

After power failure on my server, now when I start my freeswitch? I see these errors logs on my freeswitch.

state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='18572166595',dialplan='XML',context='public',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set state='CS_EXECUTE' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set application='sched_hangup',application_data='+10800 alloted_timeout',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set application='answer',application_data='',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'

How can I resolve this on my server?

I was using mysql database to dump my cdr using mod_json_cdr.

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

Official FreeSWITCH Sites  
http://www.freeswitch.org  
http://confluence.freeswitch.org  
http://www.cluecon.com  

FreeSWITCH-users mailing list  
FreeSWITCH-users at lists.freeswitch.org  
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users  
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users  
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/cfe3b411/attachment.html 

From ssinyagin at gmail.com  Fri Mar  6 15:41:34 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 6 Mar 2015 13:41:34 +0100
Subject: [Freeswitch-users] freeswitch support
In-Reply-To: <CABFbAiVDwMZpc2z+y1ynpZVhfseX1hC+jEM-5-JL7RG4L9ZU=Q@mail.gmail.com>
References: <CABFbAiVDwMZpc2z+y1ynpZVhfseX1hC+jEM-5-JL7RG4L9ZU=Q@mail.gmail.com>
Message-ID: <CACOYK=jxHYNHbvCU2+EoWG7ri2uC=musiosh7uo3b-RwqAMVKQ@mail.gmail.com>

with 100K paying customers, you should be able to afford a redundant
solution and bigger boxes :)

say, with 1:20 ratio, at peak hours you would get 5000 simultaneous
calls. It's quite a lot for a single, moderately equipped server. Even
without transcoding, that's quite a lot of work for such a CPU.

Then, 100k registered users, let's say with 3600 second expiry time,
would send you 27 REGISTER messages per second. Plus, most of them
will flood you with SIP OPTIONS pings -- even with one ping per
minute, you would need to expect 1600 pings per second. So, your CPU
would be quite busy with processing SIP messages as well.

so, the answer is no. You need to invest into hardware and redundant
and distributed architecture. A Kamailio cluster in front of
FreeSWITCH would offload most of SIP ping traffic and can do some
other jobs. Then, several FreeSWITCH boxes would work in
fault-tolerant and load-balancing configuration.... Something like
that. It just needs a proper investment and a proper engineering team.







On Fri, Mar 6, 2015 at 12:58 AM, Antony Aguirre Morales
<ing.antonyam at gmail.com> wrote:
> I have configured my fs with xml_curl module in the directory part, may
> serve 100,000 extensions from a single FS or is there any limitation as to
> the software?
>
> IN the hardware I have a server with the following specifications:
>
> Brand: DELL
> RAM: 4GB
> CPU: Intel (R) Xeon (R) CPU E5405 @ 2.00GHz
> cores: 4
> DD: 50GB
>
> regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From paul.atreides83 at googlemail.com  Fri Mar  6 16:43:15 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Fri, 6 Mar 2015 14:43:15 +0100
Subject: [Freeswitch-users] Early Dial / Sip 484
In-Reply-To: <CAEJMVkA5HToXh-LQAyauL_d2fAmxg7tz9J3ZM11=KbKCfqE1tg@mail.gmail.com>
References: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>
	<CAEJMVkA5HToXh-LQAyauL_d2fAmxg7tz9J3ZM11=KbKCfqE1tg@mail.gmail.com>
Message-ID: <CAP6oyxHD4gx7xoBq-cRqHq5H=D_-ukWYsX1Q-9e24HKNH1_Xng@mail.gmail.com>

 <action application="respond" data="484 Address Incomplete"/>
I put it at the bottom of the dial plan but it seems that the freeswitch is
not waiting long enough for
me to enter the numbers. I get send right away to the operator. Is there
any condition I have
to check so that I know the dialing has finished?


On Sun, Mar 1, 2015 at 4:07 PM, Brian West <brian at freeswitch.org> wrote:

> Yes, just use the respond app at the bottom of your dial plan with 484 as
> the argument
>
>
> On Sunday, March 1, 2015, Paul Atreides <paul.atreides83 at googlemail.com>
> wrote:
>
>> Hi,
>>
>> does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it
>> up in the dialplan?
>>
>> Thanks
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/bf321b04/attachment.html 

From regis.freeswitch.org at tornad.net  Fri Mar  6 17:11:38 2015
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 6 Mar 2015 15:11:38 +0100
Subject: [Freeswitch-users] Freeswitch RTP handling payload RFC 4040 /
	clearmode
Message-ID: <CAHWtr1R-QDCgAMperAqxTm0dsYrajSTc6MjwNuyb20MwCj6NiQ@mail.gmail.com>

Hi,

Does FS support RFC 4040 (https://tools.ietf.org/html/rfc4040) clearmode ?

"This document describes how to carry 64 kbit/s channel data

   transparently in RTP packets, using a pseudo-codec called
   "Clearmode".  It also serves as registration for a related MIME type
   called "audio/clearmode"."


Not found on wiki or confluence..

Regards,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/60ec8cf7/attachment-0001.html 

From steveayre at gmail.com  Fri Mar  6 17:43:07 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 6 Mar 2015 14:43:07 +0000
Subject: [Freeswitch-users] Early Dial / Sip 484
In-Reply-To: <CAP6oyxHD4gx7xoBq-cRqHq5H=D_-ukWYsX1Q-9e24HKNH1_Xng@mail.gmail.com>
References: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>
	<CAEJMVkA5HToXh-LQAyauL_d2fAmxg7tz9J3ZM11=KbKCfqE1tg@mail.gmail.com>
	<CAP6oyxHD4gx7xoBq-cRqHq5H=D_-ukWYsX1Q-9e24HKNH1_Xng@mail.gmail.com>
Message-ID: <CAFiqYunz9CT+GMLDtUu5nBRqPwn6n9ynbQakOHtWptp0OyUv4g@mail.gmail.com>

Early dial attempts a new SIP call each time you enter a digit. Respond
ends the 'call' for each digit. 484 tells your phone that there isn't
enough digits yet so you can continue entering digits rather than
displaying an error and ending the call.

You need to conditionally return 484 until you know you have sufficient
digits to bridge the call to the operator. For example to return 484 until
you have at least 10 digits:

<extension>
  <condition field="destination_number" data="^\d{10}">
    <action application="bridge"
data="sofia/gateway/operator/${destination_number}"/>
    <action application="respond" data="484"/>
  </condition>
</extension>

On 6 March 2015 at 13:43, Paul Atreides <paul.atreides83 at googlemail.com>
wrote:

>  <action application="respond" data="484 Address Incomplete"/>
> I put it at the bottom of the dial plan but it seems that the freeswitch
> is not waiting long enough for
> me to enter the numbers. I get send right away to the operator. Is there
> any condition I have
> to check so that I know the dialing has finished?
>
>
> On Sun, Mar 1, 2015 at 4:07 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes, just use the respond app at the bottom of your dial plan with 484 as
>> the argument
>>
>>
>> On Sunday, March 1, 2015, Paul Atreides <paul.atreides83 at googlemail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it
>>> up in the dialplan?
>>>
>>> Thanks
>>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/48c5613f/attachment.html 

From krice at freeswitch.org  Fri Mar  6 18:00:53 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 06 Mar 2015 15:00:53 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <54f9c125f26e7_53268f733479429@resque-worker-ip-10-167-66-21.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/ab501522/attachment.html 

From mike at jerris.com  Fri Mar  6 18:43:09 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 6 Mar 2015 10:43:09 -0500
Subject: [Freeswitch-users] Flooded with Stun Errors
In-Reply-To: <4D1EE950-323D-4F14-967C-7E5EE7FB2B37@gmail.com>
References: <C4DE512B-31F0-478D-BAC7-43BE2299D1BA@gmail.com>
	<BEB71057-02BC-4A6B-8AD0-524039DA0542@jerris.com>
	<4D1EE950-323D-4F14-967C-7E5EE7FB2B37@gmail.com>
Message-ID: <311220C6-C34F-43B8-A0FA-73DB2306A12C@jerris.com>

If you don't turn them on then you won't get them.  This is extra debug that is not on by default.


> On Mar 6, 2015, at 4:02 AM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> Thanks.
> 
> Should I consider this as normal? Is there any thing which can be done so that these alerts do not occur? An inside on on the reason will help me curb these alerts.
> 
> Regards,
> 
> Naveen Khanna
> 
> 
>> On 05-Mar-2015, at 11:26 pm, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> these are extra debug you get when you set debug-level in switch.conf or when using "fsctl debug_level" api command.
>> 
>>> On Mar 5, 2015, at 3:24 AM, Naveen Khanna <naveen.khanna.bm at gmail.com <mailto:naveen.khanna.bm at gmail.com>> wrote:
>>> 
>>> Hi,
>>> 
>>> I am getting flood of following messages.
>>> 
>>> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>>> 2015-03-05 13:49:25.766474 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>>> 2015-03-05 13:49:25.926473 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2042 flaws: 30 mos: 4.48 v: 23.30 9.52/400.00
>>> 2015-03-05 13:49:25.946476 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>>> 2015-03-05 13:49:25.986475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2003 at df7jal23ls0d.invalid <mailto:2003 at df7jal23ls0d.invalid> audio stat 99.00 241/242 flaws: 1 mos: 4.49 v: 55.49 10.00/400.00
>>> 2015-03-05 13:49:26.146475 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2043 flaws: 31 mos: 4.48 v: 23.29 9.52/400.00
>>> 2015-03-05 13:49:26.166475 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>>> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>>> 2015-03-05 13:49:26.286473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>>> 2015-03-05 13:49:26.366474 [ALERT] switch_rtp.c:1678 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> audio stat 98.00 2012/2044 flaws: 32 mos: 4.48 v: 23.28 9.52/400.00
>>> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>>> 2015-03-05 13:49:26.426473 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>>> 2015-03-05 13:49:26.486474 [ALERT] switch_rtp.c:5440 sofia/internal/sip:2002 at df7jal23ls0d.invalid <mailto:2002 at df7jal23ls0d.invalid> syncing 1 audio packet(s)
>>> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:875 STUN PACKET TYPE: BINDING_RESPONSE
>>> 2015-03-05 13:49:26.666475 [ALERT] switch_rtp.c:878 |---: STUN ATTR INVALID
>>> 
>>> Requesting an insight what could have been causing this.
>>> 
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/ab2950ef/attachment.html 

From mike at jerris.com  Fri Mar  6 18:43:54 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 6 Mar 2015 10:43:54 -0500
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
	<CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
Message-ID: <29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>

Always nice to hear that we are magic!  

> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> thank you very much Michael, it magically works.
> 
> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> you need to tell freeswitch to send a webrtc compatible SDP.
> 
> https://wiki.freeswitch.org/wiki/Variable_media_webrtc <https://wiki.freeswitch.org/wiki/Variable_media_webrtc>
> 
> 
>> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>> 
>> Hi,
>> 
>> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org <http://webrtc.org/> android library and connected to FreeSwitch via UDP.
>> 
>> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection.
>> 
>> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:
>> 
>> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
>> 
>> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/ <http://tryit.jssip.net/>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.
>> 
>> Here is initial SDP from Chrome (http://tryit.jssip.net/ <http://tryit.jssip.net/>):
>> 
>> v=0
>> o=- 6887715720880489867 2 IN IP4 127.0.0.1
>> s=-
>> t=0 0
>> a=group:BUNDLE audio video
>> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
>> c=IN IP4 192.168.122.1
>> a=rtcp:38359 IN IP4 192.168.122.1
>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>> .......
>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>> a=ice-ufrag:bwrCv9yS8rCY12Az
>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>> a=ice-options:google-ice
>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>> a=setup:actpass
>> a=mid:audio
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>> a=sendrecv
>> a=rtcp-mux
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=maxptime:60
>> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
>> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>> m=video 38359 RTP/SAVPF 100 116 117 96
>> c=IN IP4 192.168.122.1
>> a=rtcp:38359 IN IP4 192.168.122.1
>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>> ............
>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>> a=ice-ufrag:bwrCv9yS8rCY12Az
>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>> a=ice-options:google-ice
>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>> a=setup:actpass
>> a=mid:video
>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>> a=recvonly
>> a=rtcp-mux
>> a=rtpmap:100 VP8/90000
>> a=rtcp-fb:100 ccm fir
>> a=rtcp-fb:100 nack
>> a=rtcp-fb:100 nack pli
>> a=rtcp-fb:100 goog-remb
>> a=rtpmap:116 red/90000
>> a=rtpmap:117 ulpfec/90000
>> a=rtpmap:96 rtx/90000
>> a=fmtp:96 apt=100
>> 
>> 
>> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>> 
>> v=0
>> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
>> s=FreeSWITCH
>> c=IN IP4 192.168.131.253
>> t=0 0
>> m=audio 16390 RTP/AVP 111 0 8 101 13
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> m=video 16388 RTP/AVP 100
>> a=rtpmap:100 VP8/90000
>> 
>> 
>> I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. 
>> 
>> So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/f36bd26d/attachment-0001.html 

From 18665301040 at 163.com  Fri Mar  6 16:43:45 2015
From: 18665301040 at 163.com (james.zhu)
Date: Fri, 6 Mar 2015 21:43:45 +0800 (CST)
Subject: [Freeswitch-users] Does Sangoma Analog card support callerid in
	dtmf format?
Message-ID: <3aa66618.23bf5.14bef53dbdc.Coremail.18665301040@163.com>

Hello:
My customer installed Sangoma A200 with FreeTDM. A200 can not get the callerid in dtmf format, But if the callerid send by FSK format, the callerid can be received. I also take a loot at the code for callerid process in freetdm, it seems lack of supporting the callerid in dtmf format.  Mr.Moises also mentioned that three year ago: 
http://freeswitch-users.2379917.n2.nabble.com/Caller-id-on-incoming-FXO-with-freetdm-td5931555.html
I am not very sure the problem is still there. I post this issue to with logs, please help me check that:
https://freeswitch.org/jira/browse/OPENZAP-235?jql=project%20%3D%20OPENZAP%20AND%20resolution%20%3D%20Unresolved%20AND%20issuetype%20%3D%20%22New%20Feature%22%20ORDER%20BY%20priority%20DESC
I think callerid in dtmf/fsk format should be a standard feature, hope it can be resolved soon. 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/30b271ed/attachment.html 

From mike at jerris.com  Fri Mar  6 18:46:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 6 Mar 2015 10:46:30 -0500
Subject: [Freeswitch-users] Freeswitch RTP handling payload RFC 4040 /
	clearmode
In-Reply-To: <CAHWtr1R-QDCgAMperAqxTm0dsYrajSTc6MjwNuyb20MwCj6NiQ@mail.gmail.com>
References: <CAHWtr1R-QDCgAMperAqxTm0dsYrajSTc6MjwNuyb20MwCj6NiQ@mail.gmail.com>
Message-ID: <EF716551-37B6-4D5D-9FBD-F536C9EDBD94@jerris.com>

We haven't done anything to support this.  It conceptually could work in passthrough.  What are you trying to do, just bridge this across a freeswitch or actually terminate it to something?


> On Mar 6, 2015, at 9:11 AM, Regis M <regis.freeswitch.org at tornad.net> wrote:
> 
> Hi,
> 
> Does FS support RFC 4040 (https://tools.ietf.org/html/rfc4040 <https://tools.ietf.org/html/rfc4040>) clearmode ?
> 
> "This document describes how to carry 64 kbit/s channel data
>    transparently in RTP packets, using a pseudo-codec called
>    "Clearmode".  It also serves as registration for a related MIME type
>    called "audio/clearmode"."
> 
> Not found on wiki or confluence..
> 
> Regards,
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/f745a62f/attachment.html 

From regis.freeswitch.org at tornad.net  Fri Mar  6 19:15:06 2015
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 6 Mar 2015 17:15:06 +0100
Subject: [Freeswitch-users] Freeswitch RTP handling payload RFC 4040 /
	clearmode
In-Reply-To: <EF716551-37B6-4D5D-9FBD-F536C9EDBD94@jerris.com>
References: <CAHWtr1R-QDCgAMperAqxTm0dsYrajSTc6MjwNuyb20MwCj6NiQ@mail.gmail.com>
	<EF716551-37B6-4D5D-9FBD-F536C9EDBD94@jerris.com>
Message-ID: <CAHWtr1RTuzoLuEZWZ3mTfaup1oMGPF-WqvO4HPvzZgXOFwbrOQ@mail.gmail.com>

I have seen a request for a SIP trunk with clearmode support, so it was
transcoding's need with no passtrhu as the customer search a trunk with
this feature. So we must be able to terminate it, and change codec to a
classic one.

Thanks for the answer.. Not a critical need :)

Regards,


2015-03-06 16:46 GMT+01:00 Michael Jerris <mike at jerris.com>:

> We haven't done anything to support this.  It conceptually could work in
> passthrough.  What are you trying to do, just bridge this across a
> freeswitch or actually terminate it to something?
>
>
> On Mar 6, 2015, at 9:11 AM, Regis M <regis.freeswitch.org at tornad.net>
> wrote:
>
> Hi,
>
> Does FS support RFC 4040 (https://tools.ietf.org/html/rfc4040) clearmode ?
>
> "This document describes how to carry 64 kbit/s channel data
>
>    transparently in RTP packets, using a pseudo-codec called
>    "Clearmode".  It also serves as registration for a related MIME type
>    called "audio/clearmode"."
>
>
> Not found on wiki or confluence..
>
> Regards,
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/463e7d1c/attachment.html 

From richard.mace at gmail.com  Fri Mar  6 22:33:59 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Fri, 6 Mar 2015 19:33:59 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
Message-ID: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>

Hi All,
I did a fresh install of both Debian and FreeSWITCH today, following the
article here:
https://freeswitch.org/confluence/display/FREESWITCH/Debian

However, after installation, fs_cli will not connect. Any ideas?

Thanks

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/66ff3d27/attachment.html 

From bote_radio at botecomm.com  Fri Mar  6 23:02:44 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 6 Mar 2015 15:02:44 -0500
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
	on	Debian
In-Reply-To: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
Message-ID: <00c301d05848$833f55c0$89be0140$@botecomm.com>

On a fresh FS installation fs_cli only connects to 127.0.01 localhost. 

 

To connect from a remote machine put a valid routable interface address (although I have 0.0.0.0 in mine) in 

conf/autoload_configs/event_socket.conf.xml

 

and change the password and maybe even the port depending on the crackability of your network.

 

Then you?ll probably want to configure a profile configuration file with tight permissions to avoid having to type the parameters on the command line every time you start fs_cli.

 

Check the ?command-line Interface fs_cli? Confluence page for all the details.

 

Bote

 

 

From: Richard Mace
Sent: Friday, 06 March, 2015 14:34
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on Debian

 

Hi All,

I did a fresh install of both Debian and FreeSWITCH today, following the article here:

https://freeswitch.org/confluence/display/FREESWITCH/Debian

 

However, after installation, fs_cli will not connect. Any ideas?

 

Thanks

 

Richard

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/add22dbf/attachment-0001.html 

From anthony.minessale at gmail.com  Fri Mar  6 23:22:08 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Mar 2015 14:22:08 -0600
Subject: [Freeswitch-users] sessions and CS_INIT events
In-Reply-To: <DUB130-W42AFF5EB5971D0AF2A9198D91F0@phx.gbl>
References: <DUB130-W40BCF228CE6AF25DC1336AD91F0@phx.gbl>
	<CACOYK=iTNFK=ADoLvBJ+_u00Ho4S0eVJovZY0DL3B3nfbW4q5g@mail.gmail.com>
	<CAC-LwPMG9SK4aRPrVn3gz+uqAx8B4t2im3Ri07jPDAoURxkDNw@mail.gmail.com>
	<CACOYK=i9jYgviJBZ5vDax0oVyK0jQg6WpU=w-FCi3HArmS-ojQ@mail.gmail.com>
	<DUB130-W599E29F23E429E74C12A1FD91F0@phx.gbl>
	<60002D19-0F29-4BCE-9DA3-9726D952B45F@jerris.com>
	<DUB130-W42AFF5EB5971D0AF2A9198D91F0@phx.gbl>
Message-ID: <CAKbxfG-PwyqEbsY-B+Xxhsb13ZFGkXagVH8w4tBrJ9u48y7f0A@mail.gmail.com>

You probably want to look at the dialplan module interface as well.  And
pay attention to the direction variable to distinguish inbound from out
bound Especially at the routing state not init.


On Thu, Mar 5, 2015 at 12:05 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Hi, yes i had a look at it, and yes the needs are similar, i used it at
> the beginning
> to get started, and i m using as a reference at this point but it seems
> that the use cases
> are different . thanks!
>
>
>
> ------------------------------
> From: mike at jerris.com
> Date: Thu, 5 Mar 2015 12:46:28 -0500
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sessions and CS_INIT events
>
> Given the similarity in purpose, I would look closely at how
> mod_nibblebill interfaces with freeswitch.  It sounds like your interface
> needs are nearly identical.
>
>
> On Mar 5, 2015, at 9:23 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com>
> wrote:
>
> Thank you very much guys for your contributions.
>
> I m doing it as a module for couple of reasons, being the
> most important (i believe) performance, because the module
> i m working on is to do real time charging of voice calls on a switch
> that is already  serving as a RBT service plus a bunch of IVR's to purchase
> services, this is for a ~150K user base on a single machine (cold standby)
> this switch is also scheduled to soon start providing hosted PBX services,
>
> so going the script direction
> i personally dont see that as an option at all. I do use scripts for small
> no so much used
> much simpler stuff though, e.g: a lua script takes care of authenticating
> users
> when doing international calls from company extensions in the hosted PBX
> solution.
>
> The other reason i chose to do
> this as a module because C is the language i feel more comfortable with.
> i hope this clarifies i little bit this.
>
> Moving on, right now i m developing on a test freeswitch that we have and
> yes i noticed
> that subscribing to the CS_INIT event does represent a big problem
> because i get notified for every single of those events that is generated
> on
> freeswitch which would be very inconvenient because as i mentioned, the
> same
> switch does many other things that i m not interested in, so i m going to
> try the advise
> provided and try to do it in the dial plan, i will explore this option.
>
> thank you very much all!
>
>
>
>
>
> > Date: Thu, 5 Mar 2015 14:19:51 +0100
> > From: ssinyagin at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] sessions and CS_INIT events
> >
> > but for the task that OP has described, writing (and maintaining it in
> > the long term) a module is really an overkill. Plus, he would also
> > need to take care of multithreading within FreeSWITCH, as well as
> > memory management, etc.
> >
> > Also, a module makes sense if it's some common task which can be
> > re-used by others and published as open source. If it's some
> > closed-source module for a specific enterprise task that Juan has, it
> > just doesn't make sense and too much risk for a long-term solution.
> >
> >
> >
> >
> >
> > On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > > It all depends on what you are trying to do with your module.
> > > You can use a dialplan handler in your module (see mod_enum for
> example) to
> > > route inbound calls using your custom dialplan.
> > > You can use a state handler in your module and bind to channel states
> (much
> > > like binding to events).
> > > You can create a dialplan app in your module to execute code when the
> app is
> > > called in dialplan
> > > (Example: <application action="mycustom_module" data="${uuid} arg1
> arg2"/>)
> > > You can use an endpoint in your module to originate calls outbound (see
> > > mod_lcr or mod_callcenter for an example)
> > > Also, you can create an API for your module
> > >
> > > IMO creating a module is much more powerful than using a script with
> ESL.
> > > But if you are going to create a module, you really don't need to mess
> with
> > > events (unless they are very specific events like CUSTOM::) because
> your
> > > module has access to much of the freeswitch core.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> > > wrote:
> > >>
> > >> why at all do you need it to be a C module inside FreeSWITCH?
> > >>
> > >> Why not writing an ESL program which would subscribe to events and
> > >> perform the needed actions?
> > >>
> > >> How about the following scenario:
> > >>
> > >> 1. In the XML dialplan, you execute "park" application on the incoming
> > >> call.
> > >>
> > >> 2. Your program is listening to events via ESL, and it recognizes that
> > >> a channel has been parked
> > >>
> > >> 3. Your program starts to playback the ringback tone into that channel
> > >>
> > >> 4. Your program performs all the needed lookups and sets needed
> > >> variables on the channel
> > >>
> > >> 5. Your program transfers or bridges the call where needed.
> > >>
> > >> This is quite easy to implement in any programming language of your
> > >> choice, easy to debug, and it's easily scalable. It can be done in a
> > >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> > >> as many parallel calls as required.
> > >>
> > >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> > >> <jpablolorenzetti at hotmail.com> wrote:
> > >> > Hi, i m writing a module in C that needs to check for certain
> > >> > information in
> > >> > a
> > >> > database for the caller and the destination number,
> > >> > for this the module is subscribing to the CS_INIT channel events, so
> > >> > everytime a channel is created
> > >> > the module callback is called and it checks the numbers,
> > >> > the problem is that the callback gets called twice,
> > >> > for the creation of the a-leg of the call and the creation of the
> b-leg.
> > >> > Is there any way to accomplish what i m trying to do ?
> > >> > Am i doing it the wrong way?
> > >> > I have already try getting testing for the flags in the channel but
> it
> > >> > did
> > >> > not work,
> > >> > testing of originator or originating does not yield anything ....
> > >> >
> > >> > i might be doing it wrong maybe ?
> > >> >
> > >> > Thanks!
> > >> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
> http://www.freeswitch.org http://confluence.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/eaa333cc/attachment.html 

From richard.mace at gmail.com  Fri Mar  6 23:33:25 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Fri, 6 Mar 2015 20:33:25 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <00c301d05848$833f55c0$89be0140$@botecomm.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
Message-ID: <CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>

Hi,

Sorry, I should have clarified that this is running locally on the machine
running FreeSWITCH.

Richard

On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:

> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>
>
>
> To connect from a remote machine put a valid routable interface address
> (although I have 0.0.0.0 in mine) in
>
> conf/autoload_configs/event_socket.conf.xml
>
>
>
> and change the password and maybe even the port depending on the
> crackability of your network.
>
>
>
> Then you?ll probably want to configure a profile configuration file with
> tight permissions to avoid having to type the parameters on the command
> line every time you start fs_cli.
>
>
>
> Check the ?command-line Interface fs_cli? Confluence page for all the
> details.
>
>
>
> Bote
>
>
>
>
>
> *From:* Richard Mace
> *Sent:* Friday, 06 March, 2015 14:34
> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install on
> Debian
>
>
>
> Hi All,
>
> I did a fresh install of both Debian and FreeSWITCH today, following the
> article here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>
>
>
> However, after installation, fs_cli will not connect. Any ideas?
>
>
>
> Thanks
>
>
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/2eaf900c/attachment-0001.html 

From brian at freeswitch.org  Fri Mar  6 23:42:54 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 6 Mar 2015 15:42:54 -0500
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
Message-ID: <CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>

remove

::1     localhost ip6-localhost ip6-loopback


from /etc/hosts


its a bug in debian.

On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com> wrote:

> Hi,
>
> Sorry, I should have clarified that this is running locally on the machine
> running FreeSWITCH.
>
> Richard
>
> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>
>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>
>>
>>
>> To connect from a remote machine put a valid routable interface address
>> (although I have 0.0.0.0 in mine) in
>>
>> conf/autoload_configs/event_socket.conf.xml
>>
>>
>>
>> and change the password and maybe even the port depending on the
>> crackability of your network.
>>
>>
>>
>> Then you?ll probably want to configure a profile configuration file with
>> tight permissions to avoid having to type the parameters on the command
>> line every time you start fs_cli.
>>
>>
>>
>> Check the ?command-line Interface fs_cli? Confluence page for all the
>> details.
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>> *From:* Richard Mace
>> *Sent:* Friday, 06 March, 2015 14:34
>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install
>> on Debian
>>
>>
>>
>> Hi All,
>>
>> I did a fresh install of both Debian and FreeSWITCH today, following the
>> article here:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>
>>
>>
>> However, after installation, fs_cli will not connect. Any ideas?
>>
>>
>>
>> Thanks
>>
>>
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/fb08f6ed/attachment.html 

From brian at freeswitch.org  Sat Mar  7 00:21:01 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 6 Mar 2015 16:21:01 -0500
Subject: [Freeswitch-users] Early Dial / Sip 484
In-Reply-To: <CAFiqYunz9CT+GMLDtUu5nBRqPwn6n9ynbQakOHtWptp0OyUv4g@mail.gmail.com>
References: <CAP6oyxHXuh3fO9UOtH2R27iEnd6fYH7G9ZBM41w9dmhviq-ZJg@mail.gmail.com>
	<CAEJMVkA5HToXh-LQAyauL_d2fAmxg7tz9J3ZM11=KbKCfqE1tg@mail.gmail.com>
	<CAP6oyxHD4gx7xoBq-cRqHq5H=D_-ukWYsX1Q-9e24HKNH1_Xng@mail.gmail.com>
	<CAFiqYunz9CT+GMLDtUu5nBRqPwn6n9ynbQakOHtWptp0OyUv4g@mail.gmail.com>
Message-ID: <CAEJMVkDdKGxWwVc1mBJPR3J3VdiAaoZA-EmjVz8jPnFTWTBiNQ@mail.gmail.com>

You can do this at the very bottom of your context, so if nothing matches
yet it'll 484 till something else further up does.  I've not tested this in
a while but I recall this being how I set it up.
<extension>
  <condition>
    <action application="respond" data="484"/>
  </condition>
</extension>

On Fri, Mar 6, 2015 at 9:43 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Early dial attempts a new SIP call each time you enter a digit. Respond
> ends the 'call' for each digit. 484 tells your phone that there isn't
> enough digits yet so you can continue entering digits rather than
> displaying an error and ending the call.
>
> You need to conditionally return 484 until you know you have sufficient
> digits to bridge the call to the operator. For example to return 484 until
> you have at least 10 digits:
>
> <extension>
>   <condition field="destination_number" data="^\d{10}">
>     <action application="bridge"
> data="sofia/gateway/operator/${destination_number}"/>
>     <action application="respond" data="484"/>
>   </condition>
> </extension>
>
> On 6 March 2015 at 13:43, Paul Atreides <paul.atreides83 at googlemail.com>
> wrote:
>
>>  <action application="respond" data="484 Address Incomplete"/>
>> I put it at the bottom of the dial plan but it seems that the freeswitch
>> is not waiting long enough for
>> me to enter the numbers. I get send right away to the operator. Is there
>> any condition I have
>> to check so that I know the dialing has finished?
>>
>>
>> On Sun, Mar 1, 2015 at 4:07 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Yes, just use the respond app at the bottom of your dial plan with 484
>>> as the argument
>>>
>>>
>>> On Sunday, March 1, 2015, Paul Atreides <paul.atreides83 at googlemail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set
>>>> it up in the dialplan?
>>>>
>>>> Thanks
>>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150306/df873639/attachment-0001.html 

From montecillodavid.spingine at gmail.com  Sat Mar  7 06:27:59 2015
From: montecillodavid.spingine at gmail.com (David Montecillo)
Date: Sat, 7 Mar 2015 11:27:59 +0800
Subject: [Freeswitch-users] issue terminating outbound calls in freeswitch
Message-ID: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>

Hi Guys,

Im using GOIP(GSM Over IP) to make outbound calls in freeswitch but I have
a problem terminating the call. If the recipient ends the call from its end
the call terminates normally but whenever I end a call from my end the GOIP
thinks its still engage in a call so when I try to make another outbound
call it fails. I need to reset the GOIP to make another call.

Regards,
Dave Monte
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/fcc55054/attachment.html 

From jpablolorenzetti at hotmail.com  Sat Mar  7 09:29:15 2015
From: jpablolorenzetti at hotmail.com (Juan Pablo L.)
Date: Sat, 7 Mar 2015 00:29:15 -0600
Subject: [Freeswitch-users] sessions and CS_INIT events
Message-ID: <DUB405-EAS4320134A075DC06EDE3C278D91D0@phx.gbl>

Thank you very much for the advise. I accidentally found it to be useful as well.

--- Original Message ---

From: "Anthony Minessale" <anthony.minessale at gmail.com>
Sent: March 6, 2015 2:23 PM
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] sessions and CS_INIT events

You probably want to look at the dialplan module interface as well.  And
pay attention to the direction variable to distinguish inbound from out
bound Especially at the routing state not init.


On Thu, Mar 5, 2015 at 12:05 PM, Juan Pablo L. <jpablolorenzetti at hotmail.com
> wrote:

> Hi, yes i had a look at it, and yes the needs are similar, i used it at
> the beginning
> to get started, and i m using as a reference at this point but it seems
> that the use cases
> are different . thanks!
>
>
>
> ------------------------------
> From: mike at jerris.com
> Date: Thu, 5 Mar 2015 12:46:28 -0500
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sessions and CS_INIT events
>
> Given the similarity in purpose, I would look closely at how
> mod_nibblebill interfaces with freeswitch.  It sounds like your interface
> needs are nearly identical.
>
>
> On Mar 5, 2015, at 9:23 AM, Juan Pablo L. <jpablolorenzetti at hotmail.com>
> wrote:
>
> Thank you very much guys for your contributions.
>
> I m doing it as a module for couple of reasons, being the
> most important (i believe) performance, because the module
> i m working on is to do real time charging of voice calls on a switch
> that is already  serving as a RBT service plus a bunch of IVR's to purchase
> services, this is for a ~150K user base on a single machine (cold standby)
> this switch is also scheduled to soon start providing hosted PBX services,
>
> so going the script direction
> i personally dont see that as an option at all. I do use scripts for small
> no so much used
> much simpler stuff though, e.g: a lua script takes care of authenticating
> users
> when doing international calls from company extensions in the hosted PBX
> solution.
>
> The other reason i chose to do
> this as a module because C is the language i feel more comfortable with.
> i hope this clarifies i little bit this.
>
> Moving on, right now i m developing on a test freeswitch that we have and
> yes i noticed
> that subscribing to the CS_INIT event does represent a big problem
> because i get notified for every single of those events that is generated
> on
> freeswitch which would be very inconvenient because as i mentioned, the
> same
> switch does many other things that i m not interested in, so i m going to
> try the advise
> provided and try to do it in the dial plan, i will explore this option.
>
> thank you very much all!
>
>
>
>
>
> > Date: Thu, 5 Mar 2015 14:19:51 +0100
> > From: ssinyagin at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] sessions and CS_INIT events
> >
> > but for the task that OP has described, writing (and maintaining it in
> > the long term) a module is really an overkill. Plus, he would also
> > need to take care of multithreading within FreeSWITCH, as well as
> > memory management, etc.
> >
> > Also, a module makes sense if it's some common task which can be
> > re-used by others and published as open source. If it's some
> > closed-source module for a specific enterprise task that Juan has, it
> > just doesn't make sense and too much risk for a long-term solution.
> >
> >
> >
> >
> >
> > On Thu, Mar 5, 2015 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
> > > It all depends on what you are trying to do with your module.
> > > You can use a dialplan handler in your module (see mod_enum for
> example) to
> > > route inbound calls using your custom dialplan.
> > > You can use a state handler in your module and bind to channel states
> (much
> > > like binding to events).
> > > You can create a dialplan app in your module to execute code when the
> app is
> > > called in dialplan
> > > (Example: <application action="mycustom_module" data="${uuid} arg1
> arg2"/>)
> > > You can use an endpoint in your module to originate calls outbound (see
> > > mod_lcr or mod_callcenter for an example)
> > > Also, you can create an API for your module
> > >
> > > IMO creating a module is much more powerful than using a script with
> ESL.
> > > But if you are going to create a module, you really don't need to mess
> with
> > > events (unless they are very specific events like CUSTOM::) because
> your
> > > module has access to much of the freeswitch core.
> > >
> > > Thanks.
> > >
> > >
> > >
> > > On Thu, Mar 5, 2015 at 5:07 AM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> > > wrote:
> > >>
> > >> why at all do you need it to be a C module inside FreeSWITCH?
> > >>
> > >> Why not writing an ESL program which would subscribe to events and
> > >> perform the needed actions?
> > >>
> > >> How about the following scenario:
> > >>
> > >> 1. In the XML dialplan, you execute "park" application on the incoming
> > >> call.
> > >>
> > >> 2. Your program is listening to events via ESL, and it recognizes that
> > >> a channel has been parked
> > >>
> > >> 3. Your program starts to playback the ringback tone into that channel
> > >>
> > >> 4. Your program performs all the needed lookups and sets needed
> > >> variables on the channel
> > >>
> > >> 5. Your program transfers or bridges the call where needed.
> > >>
> > >> This is quite easy to implement in any programming language of your
> > >> choice, easy to debug, and it's easily scalable. It can be done in a
> > >> multi-threading fashion, like Go or Erlang, or even Java, and perform
> > >> as many parallel calls as required.
> > >>
> > >> quite easy, and you don't have to mess with FreeSWITCH internals :)
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> On Thu, Mar 5, 2015 at 4:07 AM, Juan Pablo L.
> > >> <jpablolorenzetti at hotmail.com> wrote:
> > >> > Hi, i m writing a module in C that needs to check for certain
> > >> > information in
> > >> > a
> > >> > database for the caller and the destination number,
> > >> > for this the module is subscribing to the CS_INIT channel events, so
> > >> > everytime a channel is created
> > >> > the module callback is called and it checks the numbers,
> > >> > the problem is that the callback gets called twice,
> > >> > for the creation of the a-leg of the call and the creation of the
> b-leg.
> > >> > Is there any way to accomplish what i m trying to do ?
> > >> > Am i doing it the wrong way?
> > >> > I have already try getting testing for the flags in the channel but
> it
> > >> > did
> > >> > not work,
> > >> > testing of originator or originating does not yield anything ....
> > >> >
> > >> > i might be doing it wrong maybe ?
> > >> >
> > >> > Thanks!
> > >> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
> http://www.freeswitch.org http://confluence.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/e79b44aa/attachment.html 
-------------- next part --------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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 Mar  7 11:42:45 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 7 Mar 2015 08:42:45 +0000
Subject: [Freeswitch-users] Regex
Message-ID: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>

Hi,
Can I just confirm that the following:

    <extension name="wph_out_work">
      <condition field="destination_number" expression="^9(0\d{10})$">
        <action application="bridge" data="sofia/gateway/wph-office/$1"/>
      </condition>
    </extension>


Would route all calls that were 90 and then another 10 digits, to the
gateway wph-office by just passing the number beginning with the 0

Thanks

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/b2fb7639/attachment-0001.html 

From flokrrr at gmail.com  Sat Mar  7 12:02:54 2015
From: flokrrr at gmail.com (Florent Krieg)
Date: Sat, 7 Mar 2015 10:02:54 +0100
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
References: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
Message-ID: <CAABKsuRqjGsLef-BC2Urc-5y+LQjkxXeDPUVTB0mwuH7aSA-2g@mail.gmail.com>

Yes it would indeed.
Le 7 mars 2015 09:46, "Richard Mace" <richard.mace at gmail.com> a ?crit :

> Hi,
> Can I just confirm that the following:
>
>     <extension name="wph_out_work">
>       <condition field="destination_number" expression="^9(0\d{10})$">
>         <action application="bridge" data="sofia/gateway/wph-office/$1"/>
>       </condition>
>     </extension>
>
>
> Would route all calls that were 90 and then another 10 digits, to the
> gateway wph-office by just passing the number beginning with the 0
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/6256ee80/attachment.html 

From richard.mace at gmail.com  Sat Mar  7 12:09:49 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 7 Mar 2015 09:09:49 +0000
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAABKsuRqjGsLef-BC2Urc-5y+LQjkxXeDPUVTB0mwuH7aSA-2g@mail.gmail.com>
References: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
	<CAABKsuRqjGsLef-BC2Urc-5y+LQjkxXeDPUVTB0mwuH7aSA-2g@mail.gmail.com>
Message-ID: <CAK+KT3-4vHsQg8GxYhhsqvcECnn2HoEvS_nsU_ByaW06+emKbA@mail.gmail.com>

Thanks Florent,
There must be another reason for my calls not going through the gateway
then :)

Richard
On 7 Mar 2015 09:05, "Florent Krieg" <flokrrr at gmail.com> wrote:

> Yes it would indeed.
> Le 7 mars 2015 09:46, "Richard Mace" <richard.mace at gmail.com> a ?crit :
>
>> Hi,
>> Can I just confirm that the following:
>>
>>     <extension name="wph_out_work">
>>       <condition field="destination_number" expression="^9(0\d{10})$">
>>         <action application="bridge" data="sofia/gateway/wph-office/$1"/>
>>       </condition>
>>     </extension>
>>
>>
>> Would route all calls that were 90 and then another 10 digits, to the
>> gateway wph-office by just passing the number beginning with the 0
>>
>> Thanks
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/6b58c503/attachment.html 

From richard.mace at gmail.com  Sat Mar  7 12:57:00 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 7 Mar 2015 09:57:00 +0000
Subject: [Freeswitch-users] SIP Trunk
Message-ID: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>

Hi,
I have a trunk that currently works with Asterisk, and I am trying to get
it working with FreeSWITCH. The Asterisk config is:

[out_trunk]
disallow=all
host=sip.voip-unlimited.net
username=username
fromuser=username
secret=password
type=peer
dtmfmode=rfc2833
nat=no
context=incoming-sip
insecure=invite
allow=alaw
fromdomain=voip-unlimited.net

Any idea how I would configure the same in within FreeSWITCH please, as my
current attempt doesn't seem to be working?

Thanks very much in advance

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/11cce23c/attachment.html 

From regis.freeswitch.org at tornad.net  Sat Mar  7 18:03:46 2015
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Sat, 7 Mar 2015 16:03:46 +0100
Subject: [Freeswitch-users] SIP Trunk
In-Reply-To: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
References: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
Message-ID: <CAHWtr1QJ-vSLxXXe7byYdiDQ31Y+Bbbs4KVCWubD9Xq2TCUYBw@mail.gmail.com>

What is your current freeswitch config for this trunk ?
normal config generaly works out of the box...



2015-03-07 10:57 GMT+01:00 Richard Mace <richard.mace at gmail.com>:

> Hi,
> I have a trunk that currently works with Asterisk, and I am trying to get
> it working with FreeSWITCH. The Asterisk config is:
>
> [out_trunk]
> disallow=all
> host=sip.voip-unlimited.net
> username=username
> fromuser=username
> secret=password
> type=peer
> dtmfmode=rfc2833
> nat=no
> context=incoming-sip
> insecure=invite
> allow=alaw
> fromdomain=voip-unlimited.net
>
> Any idea how I would configure the same in within FreeSWITCH please, as my
> current attempt doesn't seem to be working?
>
> Thanks very much in advance
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/6b77c16e/attachment.html 

From bote_radio at botecomm.com  Sat Mar  7 18:11:05 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sat, 7 Mar 2015 10:11:05 -0500
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAK+KT3-4vHsQg8GxYhhsqvcECnn2HoEvS_nsU_ByaW06+emKbA@mail.gmail.com>
References: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>	<CAABKsuRqjGsLef-BC2Urc-5y+LQjkxXeDPUVTB0mwuH7aSA-2g@mail.gmail.com>
	<CAK+KT3-4vHsQg8GxYhhsqvcECnn2HoEvS_nsU_ByaW06+emKbA@mail.gmail.com>
Message-ID: <01cd01d058e8$ef40e950$cdc2bbf0$@botecomm.com>

The FreeSWITCH debug log output is quite informative, although it can be daunting at first.

 

My guess is that you have not configured a gateway under conf/sip_profiles/external/gateway-name.xml containing at least

<gateway name=?wph_out_work?>

which is the lookup key for that gateway named in the sofia line.

 

The name of the XML file is not important, just so that you can identify it. The rest of those settings from Asterisk populate the corresponding fields; even though they come from Asterisk, I doubt that FreeSWITCH will reject them J

 

If you?ve already done this, then the log files will show the bridge app being fed that sofia dial string and the lines immediately following should tell you what happened next.

 

Or perhaps that extension in the dialplan is not even being matched in the first place?

 

Hope this helps.

 

Bote

 

 

From: Richard Mace
Sent: Saturday, 07 March, 2015 04:10
Subject: Re: [Freeswitch-users] Regex

 

Thanks Florent,
There must be another reason for my calls not going through the gateway then :)

Richard

On 7 Mar 2015 09:05, "Florent Krieg" <flokrrr at gmail.com> wrote:

Yes it would indeed.

Le 7 mars 2015 09:46, "Richard Mace" <richard.mace at gmail.com> a ?crit :

Hi,

Can I just confirm that the following:

 

    <extension name="wph_out_work">

      <condition field="destination_number" expression="^9(0\d{10})$">

        <action application="bridge" data="sofia/gateway/wph-office/$1"/>

      </condition>

    </extension>

 

 

Would route all calls that were 90 and then another 10 digits, to the gateway wph-office by just passing the number beginning with the 0

 

Thanks

 

Richard

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/b2b6bccc/attachment-0001.html 

From regis.freeswitch.org at tornad.net  Sat Mar  7 18:14:00 2015
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Sat, 7 Mar 2015 16:14:00 +0100
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
References: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
Message-ID: <CAHWtr1TG3xN_11ET4bv9RgidELfOBVEbOr4vTwqcT27N1XRb-A@mail.gmail.com>

Yes it is...  https://regex101.com/r/kX6xK8/1

2015-03-07 9:42 GMT+01:00 Richard Mace <richard.mace at gmail.com>:

> Hi,
> Can I just confirm that the following:
>
>     <extension name="wph_out_work">
>       <condition field="destination_number" expression="^9(0\d{10})$">
>         <action application="bridge" data="sofia/gateway/wph-office/$1"/>
>       </condition>
>     </extension>
>
>
> Would route all calls that were 90 and then another 10 digits, to the
> gateway wph-office by just passing the number beginning with the 0
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/188f8a1c/attachment.html 

From paul.atreides83 at googlemail.com  Sat Mar  7 20:03:32 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Sat, 7 Mar 2015 18:03:32 +0100
Subject: [Freeswitch-users] Transfer back to origin on failure
Message-ID: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>

When I do a blind transfer then I want freeswitch to call back the origin
who initiated the call.
But I am not able the capture the transfer event?

<condition field="destination_number" expression="^att_xfer$">
<condition field="destination_number" expression="^is_transfer$">

They seem do be ignored by the dialplan. Is there a list what kind of
values destionation_number can have besides the called numbers?


I am doing the transfer with a grandstream gxp2140
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/c84b686b/attachment.html 

From richard.mace at gmail.com  Sun Mar  8 00:52:19 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 7 Mar 2015 21:52:19 +0000
Subject: [Freeswitch-users] SIP Trunk
In-Reply-To: <CAHWtr1QJ-vSLxXXe7byYdiDQ31Y+Bbbs4KVCWubD9Xq2TCUYBw@mail.gmail.com>
References: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
	<CAHWtr1QJ-vSLxXXe7byYdiDQ31Y+Bbbs4KVCWubD9Xq2TCUYBw@mail.gmail.com>
Message-ID: <CAK+KT3-MGYBUn0qfZCd2Ae1Oyc10NpcwBc7x4964wN8+oh+=aA@mail.gmail.com>

Hi Regis,
It's as follows:

<include>
  <gateway name="wph-office">
  <!--/// account username *required* ///-->
  <param name="username" value="username"/>
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <param name="realm" value="voip-unlimited.net"/>
  <!--/// username to use in from: *optional* same as  username, if blank
///-->
  <param name="from-user" value="username"/>
  <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
  <param name="from-domain" value="voip-unlimited.net"/>
  <!--/// account password *required* ///-->
  <param name="password" value="password"/>
  <param name="register" value="true"/>
  </gateway>
</include>

The real username and password has obviously been changed in this example

Thanks


On 7 March 2015 at 15:03, Regis M <regis.freeswitch.org at tornad.net> wrote:

> What is your current freeswitch config for this trunk ?
> normal config generaly works out of the box...
>
>
>
> 2015-03-07 10:57 GMT+01:00 Richard Mace <richard.mace at gmail.com>:
>
>> Hi,
>> I have a trunk that currently works with Asterisk, and I am trying to get
>> it working with FreeSWITCH. The Asterisk config is:
>>
>> [out_trunk]
>> disallow=all
>> host=sip.voip-unlimited.net
>> username=username
>> fromuser=username
>> secret=password
>> type=peer
>> dtmfmode=rfc2833
>> nat=no
>> context=incoming-sip
>> insecure=invite
>> allow=alaw
>> fromdomain=voip-unlimited.net
>>
>> Any idea how I would configure the same in within FreeSWITCH please, as
>> my current attempt doesn't seem to be working?
>>
>> Thanks very much in advance
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/864b1232/attachment.html 

From dragic.dusan at gmail.com  Sun Mar  8 01:13:26 2015
From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=)
Date: Sat, 7 Mar 2015 23:13:26 +0100
Subject: [Freeswitch-users] SIP Trunk
In-Reply-To: <CAK+KT3-MGYBUn0qfZCd2Ae1Oyc10NpcwBc7x4964wN8+oh+=aA@mail.gmail.com>
References: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
	<CAHWtr1QJ-vSLxXXe7byYdiDQ31Y+Bbbs4KVCWubD9Xq2TCUYBw@mail.gmail.com>
	<CAK+KT3-MGYBUn0qfZCd2Ae1Oyc10NpcwBc7x4964wN8+oh+=aA@mail.gmail.com>
Message-ID: <CAPHNFcobvHoj3XtUvzmuhWd1u-Z6yOGJ1fnXAY0OxvAUVnLQRw@mail.gmail.com>

If the sip proxy/registrar is at sip.voip-unlimited.net you probably
need <param name="proxy" value="sip.voip-unlimited.net"/>

On 7 March 2015 at 22:52, Richard Mace <richard.mace at gmail.com> wrote:
> Hi Regis,
> It's as follows:
>
> <include>
>   <gateway name="wph-office">
>   <!--/// account username *required* ///-->
>   <param name="username" value="username"/>
>   <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>   <param name="realm" value="voip-unlimited.net"/>
>   <!--/// username to use in from: *optional* same as  username, if blank
> ///-->
>   <param name="from-user" value="username"/>
>   <!--/// domain to use in from: *optional* same as  realm, if blank ///-->
>   <param name="from-domain" value="voip-unlimited.net"/>
>   <!--/// account password *required* ///-->
>   <param name="password" value="password"/>
>   <param name="register" value="true"/>
>   </gateway>
> </include>
>
> The real username and password has obviously been changed in this example
>
> Thanks
>
>
> On 7 March 2015 at 15:03, Regis M <regis.freeswitch.org at tornad.net> wrote:
>>
>> What is your current freeswitch config for this trunk ?
>> normal config generaly works out of the box...
>>
>>
>>
>> 2015-03-07 10:57 GMT+01:00 Richard Mace <richard.mace at gmail.com>:
>>>
>>> Hi,
>>> I have a trunk that currently works with Asterisk, and I am trying to get
>>> it working with FreeSWITCH. The Asterisk config is:
>>>
>>> [out_trunk]
>>> disallow=all
>>> host=sip.voip-unlimited.net
>>> username=username
>>> fromuser=username
>>> secret=password
>>> type=peer
>>> dtmfmode=rfc2833
>>> nat=no
>>> context=incoming-sip
>>> insecure=invite
>>> allow=alaw
>>> fromdomain=voip-unlimited.net
>>>
>>> Any idea how I would configure the same in within FreeSWITCH please, as
>>> my current attempt doesn't seem to be working?
>>>
>>> Thanks very much in advance
>>>
>>> Richard
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Du?an Dragi?


From martinschoepfer at gmx.de  Sun Mar  8 01:47:00 2015
From: martinschoepfer at gmx.de (=?iso-8859-1?Q?Martin_Sch=F6pfer_=28GMX=29?=)
Date: Sat, 7 Mar 2015 23:47:00 +0100
Subject: [Freeswitch-users] Freeswitch --> Faxing issue
Message-ID: <004601d05928$a15b6630$e4123290$@gmx.de>

Hello,

 

can anyone help me by my config for faxing with freeswitch. I have testet al
functions of spandsp also to set the software Modems active and receive
faxes with hylafax but nothing works.

 

That?s the function I want to get

 

inbound


 Sip-provider ? Freeswitch ? Cisco SPA3102 Line 1(T.28 enabled) ? hp fax or
another 

For now i?m getting the message ?no fax detected? from my faxing maschine

Outbound

Hp fax or another ? Cisco SPA3102 Line 1 ? Freeswitch ? Sip-provider 
.

There I get the message phone busy always ?I know the other faxing maschine
isn busy that?s my second?

 

The softmodem are also always busy I can?t check the function by cu ?l FS
because it tells me line in use.

 

Freeswitch Version is the latest master branch of 1.4

 

Thanks for your advice

 

Martin

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150307/cb8b107c/attachment-0001.html 

From nandy1925 at gmail.com  Sun Mar  8 05:31:54 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 8 Mar 2015 10:31:54 +0800
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is
	malformed]
In-Reply-To: <etPan.54f9958a.625558ec.36e@BusyBox.local>
References: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>
	<etPan.54f9958a.625558ec.36e@BusyBox.local>
Message-ID: <CAHKqMb7PP8dQot1vvPyByO17NmD8LxOYmH2rfKiCvRZ3nOY5Ew@mail.gmail.com>

I made a backup copy of the SQLite file. I made a script that 1) shutdowns
FS; 2) copy the backup copy; 3) start FS again.

I encounter this problem several times. So, is it better if I switch to
PostgreSQL or MySQL?

Tks,
/Nandy

On Fri, Mar 6, 2015 at 7:54 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
wrote:

>
> You seem to be using SQLite for core db. Remove the db files from your
> FreeSWITCH install location /db and restart FreeSWITCH.
>
> ?
> Thanks,
> Muhammad Naseer Bhatti
>
>
> From: Aqs Younas <aqsyounas at gmail.com> <aqsyounas at gmail.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
> <freeswitch-users at lists.freeswitch.org>
> Date: March 6, 2015 at 2:43:22 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
> <freeswitch-users at lists.freeswitch.org>
> Subject:  [Freeswitch-users] NATIVE SQL ERR [database disk image is
> malformed]
>
>   Hi, users.
>
> After power failure on my server, now when I start my freeswitch  I see
> these errors logs on my freeswitch.
>
> state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003
> ',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='
> 18572166595',dialplan='XML',context='public',presence_id='',presence_data=''
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set state='CS_EXECUTE' where
> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set application='sched_hangup',application_data='+10800
> alloted_timeout',presence_id='',presence_data='' where
> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> application='answer',application_data='',presence_id='',presence_data=''
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>
> How can I resolve this on my server?
>
> I was using mysql database to dump my cdr using mod_json_cdr.
>
> Thanks for you help.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/9c95ada9/attachment.html 

From max at nysolutions.com  Sun Mar  8 05:58:31 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 8 Mar 2015 02:58:31 +0000
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image
	is	malformed]
In-Reply-To: <CAHKqMb7PP8dQot1vvPyByO17NmD8LxOYmH2rfKiCvRZ3nOY5Ew@mail.gmail.com>
References: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>
	<etPan.54f9958a.625558ec.36e@BusyBox.local>
	<CAHKqMb7PP8dQot1vvPyByO17NmD8LxOYmH2rfKiCvRZ3nOY5Ew@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315CF1357@ex2.connex.local>

Although more common with SQlite, any Db can get corrupt from an unclean shutdown, you should look in to why you the server is losing power, get a UPS that can cleanly shut down the server in an even of a long power outage.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nandy Dagondon
Sent: Saturday, March 7, 2015 9:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] NATIVE SQL ERR [database disk image is malformed]

I made a backup copy of the SQLite file. I made a script that 1) shutdowns FS; 2) copy the backup copy; 3) start FS again.

I encounter this problem several times. So, is it better if I switch to PostgreSQL or MySQL?

Tks,
/Nandy

On Fri, Mar 6, 2015 at 7:54 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com<mailto:nbhatti at gmail.com>> wrote:

You seem to be using SQLite for core db. Remove the db files from your FreeSWITCH install location /db and restart FreeSWITCH.

?
Thanks,
Muhammad Naseer Bhatti


From: Aqs Younas <aqsyounas at gmail.com><mailto:aqsyounas at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>><mailto:freeswitch-users at lists.freeswitch.org>
Date: March 6, 2015 at 2:43:22 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>><mailto:freeswitch-users at lists.freeswitch.org>
Subject:  [Freeswitch-users] NATIVE SQL ERR [database disk image is malformed]


Hi, users.

After power failure on my server, now when I start my freeswitch  I see these errors logs on my freeswitch.

state='CS_ROUTING',cid_name='12397284003<tel:12397284003>',cid_num='12397284003<tel:12397284003>',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='18572166595<tel:18572166595>',dialplan='XML',context='public',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set state='CS_EXECUTE' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set application='sched_hangup',application_data='+10800 alloted_timeout',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set application='answer',application_data='',presence_id='',presence_data='' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [database disk image is malformed]
update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
How can I resolve this on my server?
I was using mysql database to dump my cdr using mod_json_cdr.
Thanks for you help.
_________________________________________________________________________
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://confluence.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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/a1d100e9/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/a1d100e9/attachment-0001.jpg 

From italorossib at gmail.com  Sun Mar  8 06:35:12 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Sun, 8 Mar 2015 00:35:12 -0300
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAHWtr1TG3xN_11ET4bv9RgidELfOBVEbOr4vTwqcT27N1XRb-A@mail.gmail.com>
References: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
	<CAHWtr1TG3xN_11ET4bv9RgidELfOBVEbOr4vTwqcT27N1XRb-A@mail.gmail.com>
Message-ID: <CAJR7f2Gg+tTQ74zP=ZdWGGvHn4Ceo0LmTGXCwzXbqgvvJKUKKA@mail.gmail.com>

Regis,

You know that the regex is ok, but you need to find if this specific
extension in your dialplan is being tested during the call routing, if you
post your dialplan and your debug logs you'll get better answers.
Em 07/03/2015 12:14, "Regis M" <regis.freeswitch.org at tornad.net> escreveu:

> Yes it is...  https://regex101.com/r/kX6xK8/1
>
> 2015-03-07 9:42 GMT+01:00 Richard Mace <richard.mace at gmail.com>:
>
>> Hi,
>> Can I just confirm that the following:
>>
>>     <extension name="wph_out_work">
>>       <condition field="destination_number" expression="^9(0\d{10})$">
>>         <action application="bridge" data="sofia/gateway/wph-office/$1"/>
>>       </condition>
>>     </extension>
>>
>>
>> Would route all calls that were 90 and then another 10 digits, to the
>> gateway wph-office by just passing the number beginning with the 0
>>
>> Thanks
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/f31e5458/attachment.html 

From richard.mace at gmail.com  Sun Mar  8 09:44:05 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Sun, 8 Mar 2015 06:44:05 +0000
Subject: [Freeswitch-users] SIP Trunk
In-Reply-To: <CAPHNFcobvHoj3XtUvzmuhWd1u-Z6yOGJ1fnXAY0OxvAUVnLQRw@mail.gmail.com>
References: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
	<CAHWtr1QJ-vSLxXXe7byYdiDQ31Y+Bbbs4KVCWubD9Xq2TCUYBw@mail.gmail.com>
	<CAK+KT3-MGYBUn0qfZCd2Ae1Oyc10NpcwBc7x4964wN8+oh+=aA@mail.gmail.com>
	<CAPHNFcobvHoj3XtUvzmuhWd1u-Z6yOGJ1fnXAY0OxvAUVnLQRw@mail.gmail.com>
Message-ID: <CAK+KT3-1zSuH_8GYgAGHc+8pFZig+Y0kBVKhx5X-nrutbc--Uw@mail.gmail.com>

Thanks Dusan,
That was the answer. All working now :)

Richard

On 7 March 2015 at 22:13, Du?an Dragi? <dragic.dusan at gmail.com> wrote:

> If the sip proxy/registrar is at sip.voip-unlimited.net you probably
> need <param name="proxy" value="sip.voip-unlimited.net"/>
>
> On 7 March 2015 at 22:52, Richard Mace <richard.mace at gmail.com> wrote:
> > Hi Regis,
> > It's as follows:
> >
> > <include>
> >   <gateway name="wph-office">
> >   <!--/// account username *required* ///-->
> >   <param name="username" value="username"/>
> >   <!--/// auth realm: *optional* same as gateway name, if blank ///-->
> >   <param name="realm" value="voip-unlimited.net"/>
> >   <!--/// username to use in from: *optional* same as  username, if blank
> > ///-->
> >   <param name="from-user" value="username"/>
> >   <!--/// domain to use in from: *optional* same as  realm, if blank
> ///-->
> >   <param name="from-domain" value="voip-unlimited.net"/>
> >   <!--/// account password *required* ///-->
> >   <param name="password" value="password"/>
> >   <param name="register" value="true"/>
> >   </gateway>
> > </include>
> >
> > The real username and password has obviously been changed in this example
> >
> > Thanks
> >
> >
> > On 7 March 2015 at 15:03, Regis M <regis.freeswitch.org at tornad.net>
> wrote:
> >>
> >> What is your current freeswitch config for this trunk ?
> >> normal config generaly works out of the box...
> >>
> >>
> >>
> >> 2015-03-07 10:57 GMT+01:00 Richard Mace <richard.mace at gmail.com>:
> >>>
> >>> Hi,
> >>> I have a trunk that currently works with Asterisk, and I am trying to
> get
> >>> it working with FreeSWITCH. The Asterisk config is:
> >>>
> >>> [out_trunk]
> >>> disallow=all
> >>> host=sip.voip-unlimited.net
> >>> username=username
> >>> fromuser=username
> >>> secret=password
> >>> type=peer
> >>> dtmfmode=rfc2833
> >>> nat=no
> >>> context=incoming-sip
> >>> insecure=invite
> >>> allow=alaw
> >>> fromdomain=voip-unlimited.net
> >>>
> >>> Any idea how I would configure the same in within FreeSWITCH please, as
> >>> my current attempt doesn't seem to be working?
> >>>
> >>> Thanks very much in advance
> >>>
> >>> Richard
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Du?an Dragi?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/3f366cf7/attachment.html 

From nandy1925 at gmail.com  Sun Mar  8 14:45:32 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 8 Mar 2015 19:45:32 +0800
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is
	malformed]
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315CF1357@ex2.connex.local>
References: <CAGVOjVVqWJUv0UQ_fo_Vr67hPvof2fWFUFtQo95LNHp8W3+0qg@mail.gmail.com>
	<etPan.54f9958a.625558ec.36e@BusyBox.local>
	<CAHKqMb7PP8dQot1vvPyByO17NmD8LxOYmH2rfKiCvRZ3nOY5Ew@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315CF1357@ex2.connex.local>
Message-ID: <CAHKqMb6pixy=jy3hae3VPNnimNEFqwQVDdx7_tqpmWYFfy2gmA@mail.gmail.com>

Hi Moishe,

The corruption happened even without power failures.  I begin to suspect
that an electrical surge or noise is causing this corruption. A line
conditioner is needed in this case.

Thanks for your input re other databases.

/Nandy


On Sun, Mar 8, 2015 at 10:58 AM, Moishe Grunstein <max at nysolutions.com>
wrote:

>  Although more common with SQlite, any Db can get corrupt from an unclean
> shutdown, you should look in to why you the server is losing power, get a
> UPS that can cleanly shut down the server in an even of a long power outage.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nandy
> Dagondon
> *Sent:* Saturday, March 7, 2015 9:32 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] NATIVE SQL ERR [database disk image is
> malformed]
>
>
>
> I made a backup copy of the SQLite file. I made a script that 1) shutdowns
> FS; 2) copy the backup copy; 3) start FS again.
>
>
>
> I encounter this problem several times. So, is it better if I switch to
> PostgreSQL or MySQL?
>
>
>    Tks,
>
> /Nandy
>
>
>
> On Fri, Mar 6, 2015 at 7:54 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
> wrote:
>
>
>
> You seem to be using SQLite for core db. Remove the db files from your
> FreeSWITCH install location /db and restart FreeSWITCH.
>
>
>
> ?
>
> Thanks,
>
> Muhammad Naseer Bhatti
>
>
>
>
> From: Aqs Younas <aqsyounas at gmail.com> <aqsyounas at gmail.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
> <freeswitch-users at lists.freeswitch.org>
> Date: March 6, 2015 at 2:43:22 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
> <freeswitch-users at lists.freeswitch.org>
> Subject:  [Freeswitch-users] NATIVE SQL ERR [database disk image is
> malformed]
>
>
>
>      Hi, users.
>
> After power failure on my server, now when I start my freeswitch  I see
> these errors logs on my freeswitch.
>
> state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003
> ',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='
> 18572166595',dialplan='XML',context='public',presence_id='',presence_data=''
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set state='CS_EXECUTE' where
> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set application='sched_hangup',application_data='+10800
> alloted_timeout',presence_id='',presence_data='' where
> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> application='answer',application_data='',presence_id='',presence_data=''
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
> 2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [database disk image is malformed]
> update channels set
> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>
> How can I resolve this on my server?
>
> I was using mysql database to dump my cdr using mod_json_cdr.
>
> Thanks for you help.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/803ab740/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/803ab740/attachment-0001.jpg 

From s.safarov at gmail.com  Sun Mar  8 17:39:03 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 08 Mar 2015 14:39:03 +0000
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is
	malformed]
Message-ID: <CAHtxdDfviY=veVMCgi0ktf5E3NRuyZO4-jK3YoFxYo5a_biY+A@mail.gmail.com>

Nandy, online UPS will protect you data from this error

??, 8 ????? 2015, 14:47, Nandy Dagondon <nandy1925 at gmail.com>:

> Hi Moishe,
>
> The corruption happened even without power failures.  I begin to suspect
> that an electrical surge or noise is causing this corruption. A line
> conditioner is needed in this case.
>
> Thanks for your input re other databases.
>
> /Nandy
>
>
>
> On Sun, Mar 8, 2015 at 10:58 AM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>>  Although more common with SQlite, any Db can get corrupt from an
>> unclean shutdown, you should look in to why you the server is losing power,
>> get a UPS that can cleanly shut down the server in an even of a long power
>> outage.
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Moishe Grunstein
>>
>> Tornado Computer Systems, Inc.
>>
>> 212.400.7650 888.IPPBX.US
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>>
>> Computer Networking * Managed Services * IP Video Surveillance * Network
>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>> Security * Site Surveys * CMS
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nandy
>> Dagondon
>> *Sent:* Saturday, March 7, 2015 9:32 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] NATIVE SQL ERR [database disk image is
>> malformed]
>>
>>
>>
>> I made a backup copy of the SQLite file. I made a script that 1)
>> shutdowns FS; 2) copy the backup copy; 3) start FS again.
>>
>>
>>
>> I encounter this problem several times. So, is it better if I switch to
>> PostgreSQL or MySQL?
>>
>>
>>    Tks,
>>
>> /Nandy
>>
>>
>>
>> On Fri, Mar 6, 2015 at 7:54 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com>
>> wrote:
>>
>>
>>
>> You seem to be using SQLite for core db. Remove the db files from your
>> FreeSWITCH install location /db and restart FreeSWITCH.
>>
>>
>>
>> ?
>>
>> Thanks,
>>
>> Muhammad Naseer Bhatti
>>
>>
>>
>>
>> From: Aqs Younas <aqsyounas at gmail.com> <aqsyounas at gmail.com>
>> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: March 6, 2015 at 2:43:22 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
>> <freeswitch-users at lists.freeswitch.org>
>> Subject:  [Freeswitch-users] NATIVE SQL ERR [database disk image is
>> malformed]
>>
>>
>>
>>      Hi, users.
>>
>> After power failure on my server, now when I start my freeswitch  I see
>> these errors logs on my freeswitch.
>>
>> state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003
>> ',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='
>> 18572166595',dialplan='XML',context='public',presence_id='',presence_data=''
>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set state='CS_EXECUTE' where
>> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set application='sched_hangup',application_data='+10800
>> alloted_timeout',presence_id='',presence_data='' where
>> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set
>> application='answer',application_data='',presence_id='',presence_data=''
>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set
>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set
>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>> 2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [database disk image is malformed]
>> update channels set
>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>
>> How can I resolve this on my server?
>>
>> I was using mysql database to dump my cdr using mod_json_cdr.
>>
>> Thanks for you help.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/be4242ca/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/be4242ca/attachment-0002.jpg 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/be4242ca/attachment-0003.jpg 

From aademattia at Comcast.net  Sun Mar  8 22:35:36 2015
From: aademattia at Comcast.net (Andrew)
Date: Sun, 8 Mar 2015 15:35:36 -0400
Subject: [Freeswitch-users] illegal instruction
Message-ID: <002201d059d7$11a76c50$34f644f0$@Comcast.net>

Hi,

I have an odd issue.  I was going to run out of the box FreeSWITCH on a
windows server and just by double clicking on freeswitch.exe I get a crash.
When I did a debug I found it was illegal instruction error.

 

If I remove mod_spandsp The program starts up fine.  I then try to do a tone
detection and then the program crashes again.  What would cause the same
release code to work on one server but crash on another server?

 

Andrew

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/15cd2a89/attachment.html 

From paul.atreides83 at googlemail.com  Sun Mar  8 23:11:34 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Sun, 8 Mar 2015 21:11:34 +0100
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
References: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
Message-ID: <CAP6oyxF_CErJR9FjtZHpU-ibZ7Zj7XycQ=XSmuaysRNOyctsXQ@mail.gmail.com>

2015-03-08 21:07:52.092450 [NOTICE] switch_ivr.c:1854 Transfer
sofia/internal/sip:14 at 10.0.200.14:5060 to XML[13 at default]
2015-03-08 21:07:52.112451 [NOTICE] switch_ivr_bridge.c:1608 Hangup
sofia/internal/18 at 192.168.176.6 [CS_EXECUTE] [NORMAL_CLEARING]
2015-03-08 21:07:52.132453 [NOTICE] switch_core_session.c:1633 Session 1
(sofia/internal/18 at 192.168.176.6) Ended
2015-03-08 21:07:52.132453 [NOTICE] switch_core_session.c:1637 Close
Channel sofia/internal/18 at 192.168.176.6 [CS_DESTROY]
2015-03-08 21:07:52.132453 [INFO] mod_dialplan_xml.c:635 Processing
<18>->13 in context default
2015-03-08 21:07:52.132453 [INFO] switch_channel.c:3062 sofia/internal/
sip:14 at 10.0.200.14:5060 Flipping CID from "" <18> to "Outbound Call" <14>
2015-03-08 21:07:52.152454 [NOTICE] switch_channel.c:1055 New Channel
sofia/internal/sip:13 at 10.0.200.13:5060
[8b43385b-ec35-4baf-9ace-395670cd2a98]
2015-03-08 21:07:52.192457 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/
sip:13 at 10.0.200.13:5060!

Does this help?


On Sat, Mar 7, 2015 at 6:03 PM, Paul Atreides <
paul.atreides83 at googlemail.com> wrote:

> When I do a blind transfer then I want freeswitch to call back the origin
> who initiated the call.
> But I am not able the capture the transfer event?
>
> <condition field="destination_number" expression="^att_xfer$">
> <condition field="destination_number" expression="^is_transfer$">
>
> They seem do be ignored by the dialplan. Is there a list what kind of
> values destionation_number can have besides the called numbers?
>
>
> I am doing the transfer with a grandstream gxp2140
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/e338ef00/attachment.html 

From jleung at v10networks.ca  Mon Mar  9 00:03:58 2015
From: jleung at v10networks.ca (Jeff Leung)
Date: Sun, 8 Mar 2015 14:03:58 -0700
Subject: [Freeswitch-users] illegal instruction
In-Reply-To: <002201d059d7$11a76c50$34f644f0$@Comcast.net>
References: <002201d059d7$11a76c50$34f644f0$@Comcast.net>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA01109DA@IMBER.smb.curriegrad2004.ca>

I'm suspecting the code was compiled with CPU specific extensions which
the current one does not support. Try recompiling FreeSWITCH without the
SMID extensions for mod_spandsp and see if that helps.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Andrew
> Sent: Sunday, March 8, 2015 12:36
> To: 'FreeSWITCH Users Help'
> Subject: [Freeswitch-users] illegal instruction
> 
> Hi,
> 
> I have an odd issue.  I was going to run out of the box FreeSWITCH on
a windows server
> and just by double clicking on freeswitch.exe I get a crash.  When I
did a debug I found it
> was illegal instruction error.
> 
> 
> 
> If I remove mod_spandsp The program starts up fine.  I then try to do
a tone detection
> and then the program crashes again.  What would cause the same release
code to work
> on one server but crash on another server?
> 
> 
> 
> Andrew
> 
> 
> 
> 



From aronp at guaranteedplus.com  Mon Mar  9 00:32:18 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Sun, 8 Mar 2015 17:32:18 -0400
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
Message-ID: <CANJp-yiPojcD90P3vcNW=CS_Yc+2Fgmk0sTnhzcTLY59WNxTww@mail.gmail.com>

@brian Why remove if you use ipv6? just need to make sure there is a line
127.0.0.1 localhost

On Fri, Mar 6, 2015 at 3:42 PM, Brian West <brian at freeswitch.org> wrote:

> remove
>
> ::1     localhost ip6-localhost ip6-loopback
>
>
> from /etc/hosts
>
>
> its a bug in debian.
>
> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
> wrote:
>
>> Hi,
>>
>> Sorry, I should have clarified that this is running locally on the
>> machine running FreeSWITCH.
>>
>> Richard
>>
>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>
>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>
>>>
>>>
>>> To connect from a remote machine put a valid routable interface address
>>> (although I have 0.0.0.0 in mine) in
>>>
>>> conf/autoload_configs/event_socket.conf.xml
>>>
>>>
>>>
>>> and change the password and maybe even the port depending on the
>>> crackability of your network.
>>>
>>>
>>>
>>> Then you?ll probably want to configure a profile configuration file with
>>> tight permissions to avoid having to type the parameters on the command
>>> line every time you start fs_cli.
>>>
>>>
>>>
>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>> details.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>> *From:* Richard Mace
>>> *Sent:* Friday, 06 March, 2015 14:34
>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install
>>> on Debian
>>>
>>>
>>>
>>> Hi All,
>>>
>>> I did a fresh install of both Debian and FreeSWITCH today, following the
>>> article here:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>
>>>
>>>
>>> However, after installation, fs_cli will not connect. Any ideas?
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>> Richard
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/65b47787/attachment-0001.html 

From aronp at guaranteedplus.com  Mon Mar  9 00:39:43 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Sun, 8 Mar 2015 17:39:43 -0400
Subject: [Freeswitch-users] git push invalid format
Message-ID: <CANJp-yinPDUT383x-t8pmzR_HdAY4bqMg5HqgtOfU87P51M-6A@mail.gmail.com>

Hi,

I'm trying to push the freeswitch git repo to my github, but I get the
following error


remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
format - expected 'committer' line
remote: fatal: Error in object
channel_by_id: 0: bad id: channel free
Received window adjust for non-open channel 0.
error: pack-objects died of signal 13

This is caused by having multiple authors on a commit (which in general is
not allowed by git) and github verifies the commits and rejects it.

here is the output of git fsck

Checking object directories: 100% (256/256), done.
error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
expected 'committer' line
error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
expected 'committer' line
Checking objects: 100% (254227/254227), done.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/553b71ab/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar  9 01:00:13 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Sun, 8 Mar 2015 23:00:13 +0100
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAP6oyxF_CErJR9FjtZHpU-ibZ7Zj7XycQ=XSmuaysRNOyctsXQ@mail.gmail.com>
References: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
	<CAP6oyxF_CErJR9FjtZHpU-ibZ7Zj7XycQ=XSmuaysRNOyctsXQ@mail.gmail.com>
Message-ID: <CAP6oyxFTpK1e6uW1SZ9hiNDSSuC48ZoKVXoAQbMx11XdijNmvQ@mail.gmail.com>

Now I tried to catch the blind transfer event with

    <extension name="blind_redirect">
      <condition field="sip_refer_to" expression="^\S+$">

      </condition>
    </extension>

but that ain't working neither :(

On Sun, Mar 8, 2015 at 9:11 PM, Paul Atreides <
paul.atreides83 at googlemail.com> wrote:

>
> 2015-03-08 21:07:52.092450 [NOTICE] switch_ivr.c:1854 Transfer
> sofia/internal/sip:14 at 10.0.200.14:5060 to XML[13 at default]
> 2015-03-08 21:07:52.112451 [NOTICE] switch_ivr_bridge.c:1608 Hangup
> sofia/internal/18 at 192.168.176.6 [CS_EXECUTE] [NORMAL_CLEARING]
> 2015-03-08 21:07:52.132453 [NOTICE] switch_core_session.c:1633 Session 1
> (sofia/internal/18 at 192.168.176.6) Ended
> 2015-03-08 21:07:52.132453 [NOTICE] switch_core_session.c:1637 Close
> Channel sofia/internal/18 at 192.168.176.6 [CS_DESTROY]
> 2015-03-08 21:07:52.132453 [INFO] mod_dialplan_xml.c:635 Processing
> <18>->13 in context default
> 2015-03-08 21:07:52.132453 [INFO] switch_channel.c:3062 sofia/internal/
> sip:14 at 10.0.200.14:5060 Flipping CID from "" <18> to "Outbound Call" <14>
> 2015-03-08 21:07:52.152454 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/sip:13 at 10.0.200.13:5060
> [8b43385b-ec35-4baf-9ace-395670cd2a98]
> 2015-03-08 21:07:52.192457 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/
> sip:13 at 10.0.200.13:5060!
>
> Does this help?
>
>
> On Sat, Mar 7, 2015 at 6:03 PM, Paul Atreides <
> paul.atreides83 at googlemail.com> wrote:
>
>> When I do a blind transfer then I want freeswitch to call back the origin
>> who initiated the call.
>> But I am not able the capture the transfer event?
>>
>> <condition field="destination_number" expression="^att_xfer$">
>> <condition field="destination_number" expression="^is_transfer$">
>>
>> They seem do be ignored by the dialplan. Is there a list what kind of
>> values destionation_number can have besides the called numbers?
>>
>>
>> I am doing the transfer with a grandstream gxp2140
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/cbc36a11/attachment.html 

From bote_radio at botecomm.com  Mon Mar  9 01:14:56 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sun, 8 Mar 2015 18:14:56 -0400
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CANJp-yinPDUT383x-t8pmzR_HdAY4bqMg5HqgtOfU87P51M-6A@mail.gmail.com>
References: <CANJp-yinPDUT383x-t8pmzR_HdAY4bqMg5HqgtOfU87P51M-6A@mail.gmail.com>
Message-ID: <02f601d059ed$4ffa0150$efee03f0$@botecomm.com>

This is only a guess as I have never used github, but I believe I overhead that this error is one of the big reasons that FreeSWITCH runs its own repository. In other words, it is expected and unlikely to be corrected since they have known about it for a long time and the error persists.

 

Others who know better should correct me, though.

 

Bote

 

 

From: Podrigal, Aron
Sent: Sunday, 08 March, 2015 17:40
Subject: [Freeswitch-users] git push invalid format

 

Hi,

 

I'm trying to push the freeswitch git repo to my github, but I get the following error

 

 

remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid format - expected 'committer' line

remote: fatal: Error in object

channel_by_id: 0: bad id: channel free

Received window adjust for non-open channel 0.

error: pack-objects died of signal 13

 

This is caused by having multiple authors on a commit (which in general is not allowed by git) and github verifies the commits and rejects it.

 

here is the output of git fsck

 

Checking object directories: 100% (256/256), done.

error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format - expected 'committer' line

error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format - expected 'committer' line

Checking objects: 100% (254227/254227), done.

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/098f7322/attachment.html 

From krice at freeswitch.org  Mon Mar  9 03:34:58 2015
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 08 Mar 2015 18:34:58 -0600
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CANJp-yinPDUT383x-t8pmzR_HdAY4bqMg5HqgtOfU87P51M-6A@mail.gmail.com>
Message-ID: <D12246D2.98025%krice@freeswitch.org>

This is a known issue with github and will not be fixed



On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com> wrote:

> Hi,
> 
> I'm trying to push the freeswitch git repo to my github, but I get the
> following error
> 
> 
> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid format
> - expected 'committer' line
> remote: fatal: Error in object
> channel_by_id: 0: bad id: channel free
> Received window adjust for non-open channel 0.
> error: pack-objects died of signal 13
> 
> This is caused by having multiple authors on a commit (which in general is not
> allowed by git) and github verifies the commits and rejects it.
> 
> here is the output of git fsck
> 
> Checking object directories: 100% (256/256), done.
> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
> expected 'committer' line
> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
> expected 'committer' line
> Checking objects: 100% (254227/254227), done.
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150308/86e96790/attachment.html 

From steveayre at gmail.com  Mon Mar  9 03:10:11 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 Mar 2015 00:10:11 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CANJp-yiPojcD90P3vcNW=CS_Yc+2Fgmk0sTnhzcTLY59WNxTww@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CANJp-yiPojcD90P3vcNW=CS_Yc+2Fgmk0sTnhzcTLY59WNxTww@mail.gmail.com>
Message-ID: <CAFiqYu=+WUTuWfY0sMiCo4xdGz1sDCUkfwP3rAUOTT1gzSShTg@mail.gmail.com>

Given Brian's comment that suggests that mod_event_socket is probably
configured to listen on "127.0.0.1" but fs_cli connects to "localhost" and
so tries to connect to ::1 instead, which FreeSWITCH isn't listening on
since it's listening for v4 only. So you could workaround that by
telling mod_event_socket to listen on "::"



On 8 March 2015 at 21:32, Podrigal, Aron <aronp at guaranteedplus.com> wrote:

> @brian Why remove if you use ipv6? just need to make sure there is a line
> 127.0.0.1 localhost
>
> On Fri, Mar 6, 2015 at 3:42 PM, Brian West <brian at freeswitch.org> wrote:
>
>> remove
>>
>> ::1     localhost ip6-localhost ip6-loopback
>>
>>
>> from /etc/hosts
>>
>>
>> its a bug in debian.
>>
>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> Sorry, I should have clarified that this is running locally on the
>>> machine running FreeSWITCH.
>>>
>>> Richard
>>>
>>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>>
>>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>>
>>>>
>>>>
>>>> To connect from a remote machine put a valid routable interface address
>>>> (although I have 0.0.0.0 in mine) in
>>>>
>>>> conf/autoload_configs/event_socket.conf.xml
>>>>
>>>>
>>>>
>>>> and change the password and maybe even the port depending on the
>>>> crackability of your network.
>>>>
>>>>
>>>>
>>>> Then you?ll probably want to configure a profile configuration file
>>>> with tight permissions to avoid having to type the parameters on the
>>>> command line every time you start fs_cli.
>>>>
>>>>
>>>>
>>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>>> details.
>>>>
>>>>
>>>>
>>>> Bote
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *From:* Richard Mace
>>>> *Sent:* Friday, 06 March, 2015 14:34
>>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install
>>>> on Debian
>>>>
>>>>
>>>>
>>>> Hi All,
>>>>
>>>> I did a fresh install of both Debian and FreeSWITCH today, following
>>>> the article here:
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>>
>>>>
>>>>
>>>> However, after installation, fs_cli will not connect. Any ideas?
>>>>
>>>>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> Richard
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/c5f4dc49/attachment-0001.html 

From ssinyagin at gmail.com  Mon Mar  9 04:01:31 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 9 Mar 2015 02:01:31 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAFiqYu=+WUTuWfY0sMiCo4xdGz1sDCUkfwP3rAUOTT1gzSShTg@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CANJp-yiPojcD90P3vcNW=CS_Yc+2Fgmk0sTnhzcTLY59WNxTww@mail.gmail.com>
	<CAFiqYu=+WUTuWfY0sMiCo4xdGz1sDCUkfwP3rAUOTT1gzSShTg@mail.gmail.com>
Message-ID: <CACOYK=ho5hKedpk6bGe_8FMnLz_oV-QwU7txXMunbiH2JPWLwQ@mail.gmail.com>

I've just set up an IPv6-only machine (127.0.0.1 is the only ipv4 address
available), and freeswitch with default configuration:

    <param name="listen-ip" value="127.0.0.1"/>
    <param name="listen-port" value="8021"/>

The result is that nobody is listening on port 8021 at all (netstat -an |
grep 8021).

Changing the address to ::1 in autoload_configs/event_socket.conf.xml has
helped, although fs_cli now needs the host:
fs_cli -H localhost

Should I open a Jira ticket? I don't see any reason why the server failed
to listen to  127.0.0.1:8021, without any error message.









On Mon, Mar 9, 2015 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Given Brian's comment that suggests that mod_event_socket is probably
> configured to listen on "127.0.0.1" but fs_cli connects to "localhost" and
> so tries to connect to ::1 instead, which FreeSWITCH isn't listening on
> since it's listening for v4 only. So you could workaround that by
> telling mod_event_socket to listen on "::"
>
>
>
> On 8 March 2015 at 21:32, Podrigal, Aron <aronp at guaranteedplus.com> wrote:
>
>> @brian Why remove if you use ipv6? just need to make sure there is a line
>> 127.0.0.1 localhost
>>
>> On Fri, Mar 6, 2015 at 3:42 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> remove
>>>
>>> ::1     localhost ip6-localhost ip6-loopback
>>>
>>>
>>> from /etc/hosts
>>>
>>>
>>> its a bug in debian.
>>>
>>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Sorry, I should have clarified that this is running locally on the
>>>> machine running FreeSWITCH.
>>>>
>>>> Richard
>>>>
>>>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>>>
>>>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>>>
>>>>>
>>>>>
>>>>> To connect from a remote machine put a valid routable interface
>>>>> address (although I have 0.0.0.0 in mine) in
>>>>>
>>>>> conf/autoload_configs/event_socket.conf.xml
>>>>>
>>>>>
>>>>>
>>>>> and change the password and maybe even the port depending on the
>>>>> crackability of your network.
>>>>>
>>>>>
>>>>>
>>>>> Then you?ll probably want to configure a profile configuration file
>>>>> with tight permissions to avoid having to type the parameters on the
>>>>> command line every time you start fs_cli.
>>>>>
>>>>>
>>>>>
>>>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>>>> details.
>>>>>
>>>>>
>>>>>
>>>>> Bote
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> *From:* Richard Mace
>>>>> *Sent:* Friday, 06 March, 2015 14:34
>>>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh
>>>>> install on Debian
>>>>>
>>>>>
>>>>>
>>>>> Hi All,
>>>>>
>>>>> I did a fresh install of both Debian and FreeSWITCH today, following
>>>>> the article here:
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>>>
>>>>>
>>>>>
>>>>> However, after installation, fs_cli will not connect. Any ideas?
>>>>>
>>>>>
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>>
>>>>> Richard
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/49e60184/attachment.html 

From nandy1925 at gmail.com  Mon Mar  9 04:44:13 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 9 Mar 2015 09:44:13 +0800
Subject: [Freeswitch-users] NATIVE SQL ERR [database disk image is
	malformed]
In-Reply-To: <CAHtxdDfviY=veVMCgi0ktf5E3NRuyZO4-jK3YoFxYo5a_biY+A@mail.gmail.com>
References: <CAHtxdDfviY=veVMCgi0ktf5E3NRuyZO4-jK3YoFxYo5a_biY+A@mail.gmail.com>
Message-ID: <CAHKqMb4J4F1BG81aPK9atbnbDrT+u9K-zxeGE8cBjg=2n8YjSg@mail.gmail.com>

Hi Sergey,


Yes, a UPS esp intelligent types, shuts down FS properly.  But not all UPS
has line conditioners which filters out nasty noise, surge and sags esp
common-mode types. It's these disturbances which enter servers that cause
some bits in the memory (perhaps) to clubber the database file.

No doubt a UPS helps.

Tks,
/Nandy

On Sun, Mar 8, 2015 at 10:39 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Nandy, online UPS will protect you data from this error
>
> ??, 8 ????? 2015, 14:47, Nandy Dagondon <nandy1925 at gmail.com>:
>
> Hi Moishe,
>>
>> The corruption happened even without power failures.  I begin to suspect
>> that an electrical surge or noise is causing this corruption. A line
>> conditioner is needed in this case.
>>
>> Thanks for your input re other databases.
>>
>> /Nandy
>>
>>
>>
>> On Sun, Mar 8, 2015 at 10:58 AM, Moishe Grunstein <max at nysolutions.com>
>> wrote:
>>
>>>  Although more common with SQlite, any Db can get corrupt from an
>>> unclean shutdown, you should look in to why you the server is losing power,
>>> get a UPS that can cleanly shut down the server in an even of a long power
>>> outage.
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Moishe Grunstein
>>>
>>> Tornado Computer Systems, Inc.
>>>
>>> 212.400.7650 888.IPPBX.US
>>> *Service Request Email: support at nysolutions.com
>>> <support at nysolutions.com> *
>>>
>>> [image: cid:image001.jpg at 01C72F94.9EE45D60]
>>> <http://www.nysolutions.com/>
>>>
>>> Computer Networking * Managed Services * IP Video Surveillance * Network
>>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>>> Security * Site Surveys * CMS
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nandy
>>> Dagondon
>>> *Sent:* Saturday, March 7, 2015 9:32 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] NATIVE SQL ERR [database disk image
>>> is malformed]
>>>
>>>
>>>
>>> I made a backup copy of the SQLite file. I made a script that 1)
>>> shutdowns FS; 2) copy the backup copy; 3) start FS again.
>>>
>>>
>>>
>>> I encounter this problem several times. So, is it better if I switch to
>>> PostgreSQL or MySQL?
>>>
>>>
>>>    Tks,
>>>
>>> /Nandy
>>>
>>>
>>>
>>> On Fri, Mar 6, 2015 at 7:54 PM, Muhammad Naseer Bhatti <
>>> nbhatti at gmail.com> wrote:
>>>
>>>
>>>
>>> You seem to be using SQLite for core db. Remove the db files from your
>>> FreeSWITCH install location /db and restart FreeSWITCH.
>>>
>>>
>>>
>>> ?
>>>
>>> Thanks,
>>>
>>> Muhammad Naseer Bhatti
>>>
>>>
>>>
>>>
>>> From: Aqs Younas <aqsyounas at gmail.com> <aqsyounas at gmail.com>
>>> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: March 6, 2015 at 2:43:22 PM
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Subject:  [Freeswitch-users] NATIVE SQL ERR [database disk image is
>>> malformed]
>>>
>>>
>>>
>>>      Hi, users.
>>>
>>> After power failure on my server, now when I start my freeswitch  I see
>>> these errors logs on my freeswitch.
>>>
>>> state='CS_ROUTING',cid_name='12397284003',cid_num='12397284003
>>> ',callee_name='',callee_num='',sent_callee_name='',sent_callee_num='',ip_addr='216.55.169.165',dest='
>>> 18572166595',dialplan='XML',context='public',presence_id='',presence_data=''
>>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:19.447128 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set state='CS_EXECUTE' where
>>> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:19.646659 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set application='sched_hangup',application_data='+10800
>>> alloted_timeout',presence_id='',presence_data='' where
>>> uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:19.846678 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set
>>> application='answer',application_data='',presence_id='',presence_data=''
>>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:20.046670 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set
>>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:20.246667 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set
>>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>> 2015-03-06 11:29:20.447279 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>>> [database disk image is malformed]
>>> update channels set
>>> read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000'
>>> where uuid='d1dabf5d-b0e5-49e1-a2a4-64fab587214b'
>>>
>>> How can I resolve this on my server?
>>>
>>> I was using mysql database to dump my cdr using mod_json_cdr.
>>>
>>> Thanks for you help.
>>>
>>> _________________________________________________________________________
>>>
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/450325e2/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/450325e2/attachment-0001.jpg 

From richard.mace at gmail.com  Mon Mar  9 09:49:28 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Mon, 9 Mar 2015 06:49:28 +0000
Subject: [Freeswitch-users] Incoming call - ring call group - then voicemail
Message-ID: <CAK+KT38GMxQ4N2-6e_aocAJo6+bhLyt9emvH1xEz3f-YyYjnEQ@mail.gmail.com>

Hi All,
I am trying to get an incoming call to ring the group 1000 in the domain
wph.co.uk and then go into the voicemail for 200. Is the following code
close because it doesn't seem to work?

      <action application="set" data="call_timeout=20"/>
      <action application="set" data="continue_on_fail=true"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge" data="sofia/wph.co.uk/group/1000"/>
      <action application="answer"/>
      <action application="voicemail" data="default wph.co.uk 200?/>

Thanks in advance

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/9c0c5886/attachment.html 

From steveayre at gmail.com  Mon Mar  9 13:11:24 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 Mar 2015 10:11:24 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CACOYK=ho5hKedpk6bGe_8FMnLz_oV-QwU7txXMunbiH2JPWLwQ@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CANJp-yiPojcD90P3vcNW=CS_Yc+2Fgmk0sTnhzcTLY59WNxTww@mail.gmail.com>
	<CAFiqYu=+WUTuWfY0sMiCo4xdGz1sDCUkfwP3rAUOTT1gzSShTg@mail.gmail.com>
	<CACOYK=ho5hKedpk6bGe_8FMnLz_oV-QwU7txXMunbiH2JPWLwQ@mail.gmail.com>
Message-ID: <CAFiqYumMQ9RAYpXK_urvyaPBGbSVRs4BCdgKkxz5VmFLMLGdgQ@mail.gmail.com>

127.0.0.1 is hardcoded as a default in libs/esl/fs_cli.c, should you want
to submit a patch push request.

You could also override that default in /etc/fs_cli.conf or ~/.fs_cli_conf

On 9 March 2015 at 01:01, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> I've just set up an IPv6-only machine (127.0.0.1 is the only ipv4 address
> available), and freeswitch with default configuration:
>
>     <param name="listen-ip" value="127.0.0.1"/>
>     <param name="listen-port" value="8021"/>
>
> The result is that nobody is listening on port 8021 at all (netstat -an |
> grep 8021).
>
> Changing the address to ::1 in autoload_configs/event_socket.conf.xml has
> helped, although fs_cli now needs the host:
> fs_cli -H localhost
>
> Should I open a Jira ticket? I don't see any reason why the server failed
> to listen to  127.0.0.1:8021, without any error message.
>
>
>
>
>
>
>
>
>
> On Mon, Mar 9, 2015 at 1:10 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Given Brian's comment that suggests that mod_event_socket is probably
>> configured to listen on "127.0.0.1" but fs_cli connects to "localhost" and
>> so tries to connect to ::1 instead, which FreeSWITCH isn't listening on
>> since it's listening for v4 only. So you could workaround that by
>> telling mod_event_socket to listen on "::"
>>
>>
>>
>> On 8 March 2015 at 21:32, Podrigal, Aron <aronp at guaranteedplus.com>
>> wrote:
>>
>>> @brian Why remove if you use ipv6? just need to make sure there is a
>>> line 127.0.0.1 localhost
>>>
>>> On Fri, Mar 6, 2015 at 3:42 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> remove
>>>>
>>>> ::1     localhost ip6-localhost ip6-loopback
>>>>
>>>>
>>>> from /etc/hosts
>>>>
>>>>
>>>> its a bug in debian.
>>>>
>>>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Sorry, I should have clarified that this is running locally on the
>>>>> machine running FreeSWITCH.
>>>>>
>>>>> Richard
>>>>>
>>>>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>>>>
>>>>>> On a fresh FS installation fs_cli only connects to 127.0.01
>>>>>> localhost.
>>>>>>
>>>>>>
>>>>>>
>>>>>> To connect from a remote machine put a valid routable interface
>>>>>> address (although I have 0.0.0.0 in mine) in
>>>>>>
>>>>>> conf/autoload_configs/event_socket.conf.xml
>>>>>>
>>>>>>
>>>>>>
>>>>>> and change the password and maybe even the port depending on the
>>>>>> crackability of your network.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Then you?ll probably want to configure a profile configuration file
>>>>>> with tight permissions to avoid having to type the parameters on the
>>>>>> command line every time you start fs_cli.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>>>>> details.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Bote
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Richard Mace
>>>>>> *Sent:* Friday, 06 March, 2015 14:34
>>>>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh
>>>>>> install on Debian
>>>>>>
>>>>>>
>>>>>>
>>>>>> Hi All,
>>>>>>
>>>>>> I did a fresh install of both Debian and FreeSWITCH today, following
>>>>>> the article here:
>>>>>>
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>>>>
>>>>>>
>>>>>>
>>>>>> However, after installation, fs_cli will not connect. Any ideas?
>>>>>>
>>>>>>
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>> Richard
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/b600b3f9/attachment-0001.html 

From s.safarov at gmail.com  Mon Mar  9 13:13:03 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 9 Mar 2015 13:13:03 +0300
Subject: [Freeswitch-users] module dependency
In-Reply-To: <0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
	<CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>
	<0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
Message-ID: <CAHtxdDcwcwRQeJXe9JQKM=x54ydcHq6qsKi0q7JcLH=VXvuxGg@mail.gmail.com>

Hi Michael
I has created modified version of switch_load_timezones function in
attached file.
I has make first step - "swap out the pointers", but do know not how to
make correctly thread synchronization. Can you give me reference to
function and source file where located mutex protection of dialplan reload?

I has read apr_thread_mutex_lock and apr_thread_rwlock_rdlock functions
description
apr_thread_mutex_lock
http://apr.apache.org/docs/apr/1.3/group__apr__thread__mutex.html#g1430fd10d8d260c0e3832c959742a977
apr_thread_rwlock_rdlock
http://apr.apache.org/docs/apr/1.4/group__apr__thread__rwlock.html#gad44a106cd9a81eef362d31837ca7018f

May be usage switch_thread_rwlock_t datatype
and switch_thread_rwlock_rdlock function will be more featured? This allow
multiple threads make switch_lookup_timezone.


On Tue, Mar 3, 2015 at 10:20 PM, Michael Jerris <mike at jerris.com> wrote:

> yes it will require code changes there.  I wouldn't make an idle loop
> tho.  I would do something to swap out the pointers with the new ones and
> protect it all with a mutex.  I think we do something similar with dialplan
> reload.
>
>
> On Mar 3, 2015, at 1:35 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> Will it help addition of the configuration update flag of module
> CORE_SOFTTIMER_MODULE.
> And to add idle loop 'into the function switch_lookup_timezone until
> 'update is complete?
>
> On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> That is ALWAYS loaded before any other modules, so that not being loaded
>> after.  Whats happening here, is the reload signal triggers the timezones
>> to reload asynchronously.  This will require a code change to swap those
>> out in some way that doesn't leave them empty for a short period, properly
>> protected against race conditions.  This code is in switch_time.c.
>>
>>
>> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>> >
>> > Please help me declare module dependency
>> > I has extended module radius_cdr by timezone support and from time to
>> time is getting following error
>> >
>> > freeswitch at internal> reload mod_radius_cdr
>> > +OK Reloading XML
>> > +OK module unloaded
>> > +OK module loaded
>> >
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935
>> Stopping: mod_radius_cdr
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955
>> mod_radius_cdr unloaded.
>> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
>> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone
>> 'Asia/Tokyo' not found!
>> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find
>> timezone Asia/Tokyo
>> > , Setting timezone to GMT
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465
>> Successfully Loaded [mod_radius_cdr]
>> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded
>> 1781 definitions
>> >
>> >
>> > Module currently depend of loaded configuradion of
>> CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded
>> CORE_SOFTTIMER_MODULE configuration.
>> >
>> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded
>> before mod_radius_cdr?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/3d7f0963/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: switch_load_timezones-mod.c
Type: text/x-csrc
Size: 1438 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/3d7f0963/attachment.bin 

From richard.mace at gmail.com  Mon Mar  9 13:37:36 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Mon, 9 Mar 2015 10:37:36 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
Message-ID: <CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>

Hi Brian,
Removed the line, and rebooted, but still getting:

root at FreeSWITCH:~# fs_cli
[ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]

Richard

On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:

> remove
>
> ::1     localhost ip6-localhost ip6-loopback
>
>
> from /etc/hosts
>
>
> its a bug in debian.
>
> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
> wrote:
>
>> Hi,
>>
>> Sorry, I should have clarified that this is running locally on the
>> machine running FreeSWITCH.
>>
>> Richard
>>
>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>
>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>
>>>
>>>
>>> To connect from a remote machine put a valid routable interface address
>>> (although I have 0.0.0.0 in mine) in
>>>
>>> conf/autoload_configs/event_socket.conf.xml
>>>
>>>
>>>
>>> and change the password and maybe even the port depending on the
>>> crackability of your network.
>>>
>>>
>>>
>>> Then you?ll probably want to configure a profile configuration file with
>>> tight permissions to avoid having to type the parameters on the command
>>> line every time you start fs_cli.
>>>
>>>
>>>
>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>> details.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>> *From:* Richard Mace
>>> *Sent:* Friday, 06 March, 2015 14:34
>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install
>>> on Debian
>>>
>>>
>>>
>>> Hi All,
>>>
>>> I did a fresh install of both Debian and FreeSWITCH today, following the
>>> article here:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>
>>>
>>>
>>> However, after installation, fs_cli will not connect. Any ideas?
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>> Richard
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/bfbb9a76/attachment-0001.html 

From davidwaf at gmail.com  Mon Mar  9 13:54:19 2015
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 9 Mar 2015 12:54:19 +0200
Subject: [Freeswitch-users] Chat: invalid profile
Message-ID: <CAK9Ws2dTyUrKsDsHRnP1PNq2Vd+zPCH+k0yqRGbhE2h7zzNhQA@mail.gmail.com>

Hello all,
I used to have the following command working with default installation.


chat sip|1000 at xxx.xxx.xxx.xxx|1001 at xxx.xxx.xxx.xxx|hello sopranos


After binding the chatplan to xml curl, the command no longer works, i get :

2015-03-09 12:47:41.238102 [ERR] sofia_presence.c:200 Chat proto [global]
from [1000 at xxx.xxx.xxx.xxx]
to [1000 at xxx.xxx.xxx.xxx]
hello sopranos
Invalid Profile xxx.xxx.xxx.xxx

 xxx.xxx.xxx.xxx is the same IP through.  Am not sure where to start
looking. Help.

-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/4d8139dd/attachment.html 

From cmrienzo at gmail.com  Mon Mar  9 15:01:48 2015
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Mon, 9 Mar 2015 08:01:48 -0400
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <D12246D2.98025%krice@freeswitch.org>
References: <D12246D2.98025%krice@freeswitch.org>
Message-ID: <E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>

I switched to bitbucket.org just for the FreeSWITCH repo to work around this. 


> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
> 
> This is a known issue with github and will not be fixed 
> 
> 
> 
> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com> wrote:
> 
> Hi,
> 
> I'm trying to push the freeswitch git repo to my github, but I get the following error
> 
> 
> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid format - expected 'committer' line
> remote: fatal: Error in object
> channel_by_id: 0: bad id: channel free
> Received window adjust for non-open channel 0.
> error: pack-objects died of signal 13
> 
> This is caused by having multiple authors on a commit (which in general is not allowed by git) and github verifies the commits and rejects it.
> 
> here is the output of git fsck
> 
> Checking object directories: 100% (256/256), done.
> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format - expected 'committer' line
> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format - expected 'committer' line
> Checking objects: 100% (254227/254227), done.
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> Twitter: @FreeSWITCH
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/17f4be2a/attachment.html 

From ben at langfeld.co.uk  Mon Mar  9 16:13:49 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 9 Mar 2015 10:13:49 -0300
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>
References: <D12246D2.98025%krice@freeswitch.org>
	<E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>
Message-ID: <CAAyX+KEksE4a==CYZ4oc9BtuKSezBx8fx9pVxqC8t4nwJaQ=9g@mail.gmail.com>

I'm curious about this one... If git fsck complains about the issue, what
is the justification for saying that Github is broken? How were these
commits created with a format that git itself complains about?

On 9 March 2015 at 09:01, <cmrienzo at gmail.com> wrote:

> I switched to bitbucket.org just for the FreeSWITCH repo to work around
> this.
>
>
> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>
> This is a known issue with github and will not be fixed
>
>
>
> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com> wrote:
>
> Hi,
>
> I'm trying to push the freeswitch git repo to my github, but I get the
> following error
>
>
> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
> format - expected 'committer' line
> remote: fatal: Error in object
> channel_by_id: 0: bad id: channel free
> Received window adjust for non-open channel 0.
> error: pack-objects died of signal 13
>
> This is caused by having multiple authors on a commit (which in general is
> not allowed by git) and github verifies the commits and rejects it.
>
> here is the output of git fsck
>
> Checking object directories: 100% (256/256), done.
> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
> expected 'committer' line
> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
> expected 'committer' line
> Checking objects: 100% (254227/254227), done.
>
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> Twitter: @FreeSWITCH
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/21e83c64/attachment.html 

From tfred31 at yahoo.com  Mon Mar  9 16:23:01 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Mon, 9 Mar 2015 06:23:01 -0700
Subject: [Freeswitch-users] Regex
In-Reply-To: <CAK+KT3_0C=S-qC+sb_WZKMsXhwGaA+2Euj_Ohuc__eerZdV4vw@mail.gmail.com>
Message-ID: <1425907381.30413.YahooMailBasic@web160205.mail.bf1.yahoo.com>

First, I use the following site (there are many others as well) to test any RegEx expressions that I intend to use
    https://www.regex101.com/
On that site I select the "Unit Tests", enter a number of test strings into the Add Test, and then click on the right arrow above the test strings to execute the test

But looking at the RegEx that you have written, it would REQUIRE: that the string begin with both 9 & 0 and then be followed by 10 digits.
The 'captured' value would be that within the parenthesis  ( 0nnnnnnnnnn )   which would be used by subsequent code (extensions, outside numbers, passed to 'wph-office', etc.)

Within your freeswitch.log file you should be able to see if that particular 'extension' is executed.
Do a search for    wph_out_work    and see if it is listed.   If so, they you should be able to see   PASS  or  FAIL  on the RegEx  that it received.
If it is not found by the Search, then something else might be 'intercepting' the number before it ever gets to that 'extension'

If all that looks good, then I'd look into whether or not you should be using  'bridge'  instead of  'transfer' 

Good Luck






--------------------------------------------
On Sat, 3/7/15, Richard Mace <richard.mace at gmail.com> wrote:

 Subject: [Freeswitch-users] Regex
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Saturday, March 7, 2015, 2:42 AM
 
 Hi,Can I just confirm that the
 following:
 ? ? <extension
 name="wph_out_work">? ? ? <condition
 field="destination_number"
 expression="^9(0\d{10})$">? ? ? ? <action
 application="bridge"
 data="sofia/gateway/wph-office/$1"/>? ? ?
 </condition>? ?
 </extension>
 
 Would route all calls that were 90 and then
 another 10 digits, to the gateway wph-office by just passing
 the number beginning with the 0
 Thanks
 Richard
 -----Inline Attachment Follows-----
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services: 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


From tfred31 at yahoo.com  Mon Mar  9 16:28:10 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Mon, 9 Mar 2015 06:28:10 -0700
Subject: [Freeswitch-users] SIP Trunk
In-Reply-To: <CAK+KT38AYtgyZ6gmMgG6L4hg4XqfFNcjatMk7PZN6PLdTbHXsA@mail.gmail.com>
Message-ID: <1425907690.55633.YahooMailBasic@web160205.mail.bf1.yahoo.com>

You indicate that your 'current attempt' is not working, but you do not say what that involves.

Have you defined the SIP trunk 'gateway' into its own XML file within the directory:   conf\sip_profiles\external    (it is easier to maintain that way)
And within the   conf\sip_profiles\extenal.xml    file have you made sure that the <gateways> will include that new XML file?




--------------------------------------------
On Sat, 3/7/15, Richard Mace <richard.mace at gmail.com> wrote:

 Subject: [Freeswitch-users] SIP Trunk
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Saturday, March 7, 2015, 3:57 AM
 
 Hi,I have a trunk that
 currently works with Asterisk, and I am trying to get it
 working with FreeSWITCH. The Asterisk config is:
 [out_trunk]disallow=allhost=sip.voip-unlimited.netusername=usernamefromuser=usernamesecret=passwordtype=peerdtmfmode=rfc2833nat=nocontext=incoming-sipinsecure=inviteallow=alawfromdomain=voip-unlimited.net
 Any idea how I
 would configure the same in within FreeSWITCH please, as my
 current attempt doesn't seem to be working?
 Thanks very
 much in advance
 Richard
 -----Inline Attachment Follows-----
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services: 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.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 Mar  9 16:41:44 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 09:41:44 -0400
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAAyX+KEksE4a==CYZ4oc9BtuKSezBx8fx9pVxqC8t4nwJaQ=9g@mail.gmail.com>
References: <D12246D2.98025%krice@freeswitch.org>
	<E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>
	<CAAyX+KEksE4a==CYZ4oc9BtuKSezBx8fx9pVxqC8t4nwJaQ=9g@mail.gmail.com>
Message-ID: <120E4BAE-F8A6-4404-97E7-B1D2FF3B771D@jerris.com>

We are not rewriting history to fix this so it doesn't really matter who is right or wrong.

Mike

> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
> 
> I'm curious about this one... If git fsck complains about the issue, what is the justification for saying that Github is broken? How were these commits created with a format that git itself complains about?
> 
> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com <mailto:cmrienzo at gmail.com>> wrote:
> I switched to bitbucket.org <http://bitbucket.org/> just for the FreeSWITCH repo to work around this. 
> 
> 
> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> 
>> This is a known issue with github and will not be fixed 
>> 
>> 
>> 
>> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com <http://aronp at guaranteedplus.com/>> wrote:
>> 
>> Hi,
>> 
>> I'm trying to push the freeswitch git repo to my github, but I get the following error
>> 
>> 
>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid format - expected 'committer' line
>> remote: fatal: Error in object
>> channel_by_id: 0: bad id: channel free
>> Received window adjust for non-open channel 0.
>> error: pack-objects died of signal 13
>> 
>> This is caused by having multiple authors on a commit (which in general is not allowed by git) and github verifies the commits and rejects it.
>> 
>> here is the output of git fsck
>> 
>> Checking object directories: 100% (256/256), done.
>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format - expected 'committer' line
>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format - expected 'committer' line
>> Checking objects: 100% (254227/254227), done.
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/53c031da/attachment.html 

From mike at jerris.com  Mon Mar  9 16:49:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 09:49:35 -0400
Subject: [Freeswitch-users] issue terminating outbound calls in
	freeswitch
In-Reply-To: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>
References: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>
Message-ID: <B31DB44E-8A50-4F9D-A4B8-508B1FCDB170@jerris.com>

What is GSM over IP?  To my knowledge we don't have any module that supports such a thing.


> On Mar 6, 2015, at 10:27 PM, David Montecillo <montecillodavid.spingine at gmail.com> wrote:
> 
> Hi Guys, 
> 
> Im using GOIP(GSM Over IP) to make outbound calls in freeswitch but I have a problem terminating the call. If the recipient ends the call from its end the call terminates normally but whenever I end a call from my end the GOIP thinks its still engage in a call so when I try to make another outbound call it fails. I need to reset the GOIP to make another call. 
> 
> Regards, 
> Dave Monte 



From mike at jerris.com  Mon Mar  9 16:51:13 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 09:51:13 -0400
Subject: [Freeswitch-users] Freeswitch --> Faxing issue
In-Reply-To: <004601d05928$a15b6630$e4123290$@gmx.de>
References: <004601d05928$a15b6630$e4123290$@gmx.de>
Message-ID: <28CF3ED7-970F-4843-908E-C95019D4E5E1@jerris.com>

Step 1.  Does your provider support T.38?

> On Mar 7, 2015, at 5:47 PM, Martin Sch?pfer (GMX) <martinschoepfer at gmx.de> wrote:
> 
> Hello,
>  
> can anyone help me by my config for faxing with freeswitch. I have testet al functions of spandsp also to set the software Modems active and receive faxes with hylafax but nothing works.
>  
> That?s the function I want to get
>  
> inbound
> ? Sip-provider ? Freeswitch ? Cisco SPA3102 Line 1(T.28 enabled) ? hp fax or another 
> For now i?m getting the message ?no fax detected? from my faxing maschine
> Outbound
> Hp fax or another ? Cisco SPA3102 Line 1 ? Freeswitch ? Sip-provider ?.
> There I get the message phone busy always ?I know the other faxing maschine isn busy that?s my second?
>  
> The softmodem are also always busy I can?t check the function by cu ?l FS because it tells me line in use.
>  
> Freeswitch Version is the latest master branch of 1.4

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/47680cb7/attachment.html 

From mike at jerris.com  Mon Mar  9 16:55:15 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 09:55:15 -0400
Subject: [Freeswitch-users] module dependency
In-Reply-To: <CAHtxdDcwcwRQeJXe9JQKM=x54ydcHq6qsKi0q7JcLH=VXvuxGg@mail.gmail.com>
References: <CAHtxdDeSGwJe2MEHQrxFFKtViOa_TKF=Di1f3yQWWh6_sUpT5Q@mail.gmail.com>
	<EAB26371-9A3F-4C1F-9448-00BD88F85AE5@jerris.com>
	<CAHtxdDe=UxLDt9DrUK_fCyB_OYEAFz=5owWJkRa16F3eGanu7A@mail.gmail.com>
	<0197C550-73DF-4EBC-8AEB-F78CBCCF81D1@jerris.com>
	<CAHtxdDcwcwRQeJXe9JQKM=x54ydcHq6qsKi0q7JcLH=VXvuxGg@mail.gmail.com>
Message-ID: <2785CFBF-271F-43A7-8F3D-839A05583C49@jerris.com>

Yes, i would use read/write locks.  The switch_thread ones are the right ones to use.

> On Mar 9, 2015, at 6:13 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Hi Michael
> I has created modified version of switch_load_timezones function in attached file.
> I has make first step - "swap out the pointers", but do know not how to make correctly thread synchronization. Can you give me reference to function and source file where located mutex protection of dialplan reload?
> 
> I has read apr_thread_mutex_lock and apr_thread_rwlock_rdlock functions description
> apr_thread_mutex_lock  http://apr.apache.org/docs/apr/1.3/group__apr__thread__mutex.html#g1430fd10d8d260c0e3832c959742a977 <http://apr.apache.org/docs/apr/1.3/group__apr__thread__mutex.html#g1430fd10d8d260c0e3832c959742a977>
> apr_thread_rwlock_rdlock  http://apr.apache.org/docs/apr/1.4/group__apr__thread__rwlock.html#gad44a106cd9a81eef362d31837ca7018f <http://apr.apache.org/docs/apr/1.4/group__apr__thread__rwlock.html#gad44a106cd9a81eef362d31837ca7018f>
> 
> May be usage switch_thread_rwlock_t datatype and switch_thread_rwlock_rdlock function will be more featured? This allow multiple threads make switch_lookup_timezone.
> 
> 
> On Tue, Mar 3, 2015 at 10:20 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> yes it will require code changes there.  I wouldn't make an idle loop tho.  I would do something to swap out the pointers with the new ones and protect it all with a mutex.  I think we do something similar with dialplan reload.
> 
> 
>> On Mar 3, 2015, at 1:35 PM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> 
>> Will it help addition of the configuration update flag of module CORE_SOFTTIMER_MODULE.
>> And to add idle loop 'into the function switch_lookup_timezone until 'update is complete?
>> 
>> On Tue, Mar 3, 2015 at 7:21 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> That is ALWAYS loaded before any other modules, so that not being loaded after.  Whats happening here, is the reload signal triggers the timezones to reload asynchronously.  This will require a code change to swap those out in some way that doesn't leave them empty for a short period, properly protected against race conditions.  This code is in switch_time.c.
>> 
>> 
>> > On Mar 3, 2015, at 10:41 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> >
>> > Please help me declare module dependency
>> > I has extended module radius_cdr by timezone support and from time to time is getting following error
>> >
>> > freeswitch at internal> reload mod_radius_cdr
>> > +OK Reloading XML
>> > +OK module unloaded
>> > +OK module loaded
>> >
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1935 Stopping: mod_radius_cdr
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1955 mod_radius_cdr unloaded.
>> > 2015-03-03 18:35:34.543407 [INFO] mod_enum.c:880 ENUM Reloaded
>> > 2015-03-03 18:35:34.543407 [ERR] switch_time.c:1324 Timezone 'Asia/Tokyo' not found!
>> > 2015-03-03 18:35:34.543407 [ERR] mod_radius_cdr.c:992 Cannot find timezone Asia/Tokyo
>> > , Setting timezone to GMT
>> > 2015-03-03 18:35:34.543407 [CONSOLE] switch_loadable_module.c:1465 Successfully Loaded [mod_radius_cdr]
>> > 2015-03-03 18:35:34.543407 [INFO] switch_time.c:1369 Timezone reloaded 1781 definitions
>> >
>> >
>> > Module currently depend of loaded configuradion of CORE_SOFTTIMER_MODULE but mod_radius_cdr loaded before loaded CORE_SOFTTIMER_MODULE configuration.
>> >
>> > How can I make sure that CORE_SOFTTIMER_MODULE configuration is loaded  before mod_radius_cdr?
>> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/c6a26c5e/attachment-0001.html 

From brian at freeswitch.org  Mon Mar  9 17:07:25 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 9 Mar 2015 09:07:25 -0500
Subject: [Freeswitch-users] Chat: invalid profile
In-Reply-To: <CAK9Ws2dTyUrKsDsHRnP1PNq2Vd+zPCH+k0yqRGbhE2h7zzNhQA@mail.gmail.com>
References: <CAK9Ws2dTyUrKsDsHRnP1PNq2Vd+zPCH+k0yqRGbhE2h7zzNhQA@mail.gmail.com>
Message-ID: <CAEJMVkBB6q_TU1kJKORcx51sVWANW8Fg+2pTgZBpRi-qNxcc+Q@mail.gmail.com>

alias the domain or IP to the profile so it can know where to route the
message.

On Mon, Mar 9, 2015 at 5:54 AM, David Wafula <davidwaf at gmail.com> wrote:

> Hello all,
> I used to have the following command working with default installation.
>
>
> chat sip|1000 at xxx.xxx.xxx.xxx|1001 at xxx.xxx.xxx.xxx|hello sopranos
>
>
> After binding the chatplan to xml curl, the command no longer works, i get
> :
>
> 2015-03-09 12:47:41.238102 [ERR] sofia_presence.c:200 Chat proto [global]
> from [1000 at xxx.xxx.xxx.xxx]
> to [1000 at xxx.xxx.xxx.xxx]
> hello sopranos
> Invalid Profile xxx.xxx.xxx.xxx
>
>  xxx.xxx.xxx.xxx is the same IP through.  Am not sure where to start
> looking. Help.
>
> --
> David Wafula
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/5ec906fc/attachment.html 

From rentmycoder at gmail.com  Mon Mar  9 17:08:37 2015
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 9 Mar 2015 15:08:37 +0100
Subject: [Freeswitch-users] incoming call from gateway - destination
	matching?
Message-ID: <CAEW218Jiwv2qVUu01TG6-y96cBNrVNWqNcBR-os0Fi9rk5Xd3A@mail.gmail.com>

I try to recive incoming calls from a gateway using FS without external
profile....
I would like to use just one profile, and route incoming calls based on
DID-s and context,
just like in asterisk...
But FS always logs:

sofia.c:8834 sofia/internal/13245678 at 1.2.3.4 receiving invite from
1.2.3.4:5060 version: 1.4.14 sofia.c:9001 IP 1.2.3.4 Rejected by acl
"domains". Falling back to Digest auth.

sofia_reg.c:2827 Can't find user [2000 at 192.168.101.44] from 1.2.3.4
You must define a domain called '192.168.101.44' in your directory and add
a user with the id="2000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

I do not need any users at all...

I just need to recive calls from a gateway and route them to specific
context/extension in XML dialplan...

How to do thet?
What is the context parameter of the gateway is used for, if not for that
purpuse?

Thanks,
John
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/d308eb61/attachment.html 

From vipkilla at gmail.com  Mon Mar  9 17:23:25 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 9 Mar 2015 10:23:25 -0400
Subject: [Freeswitch-users] callcenter does not exit if all agents are 'In a
	queue call'
Message-ID: <CAC-LwPMRDL+WPgNrNZxW49e8nQYP9y83O=oQvvGdZ2Qd0tPwRQ@mail.gmail.com>

Hello,
I'm trying to configured callcenter to exit app and return to dialplan IF
all agents are currently on a call.
I've tried many different strategies and timeouts to very low (like 1
second)
Still the call will sit in the queue.
I've tested this in callback mode with an external agent (outbound call)
The agent's state is 'In a queue call'

This seems like an obvious feature one would need from callcenter, yet I
cannot figure out how to configure callcenter this way.

Thanks in advance.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/96395f51/attachment.html 

From mike at jerris.com  Mon Mar  9 17:31:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 10:31:24 -0400
Subject: [Freeswitch-users] incoming call from gateway - destination
	matching?
In-Reply-To: <CAEW218Jiwv2qVUu01TG6-y96cBNrVNWqNcBR-os0Fi9rk5Xd3A@mail.gmail.com>
References: <CAEW218Jiwv2qVUu01TG6-y96cBNrVNWqNcBR-os0Fi9rk5Xd3A@mail.gmail.com>
Message-ID: <3D6CCDD5-40A5-4998-8792-D3E8CBD76E75@jerris.com>

All of this happens before it hits the dialplan.  The call is being received on a sip profile configured to require authentication and there is not matching user to authenticate against.  If your scenario never requires authentication, then don't configure it on your sip profile settings.



> On Mar 9, 2015, at 10:08 AM, rentmycoder rentmycoder <rentmycoder at gmail.com> wrote:
> 
> I try to recive incoming calls from a gateway using FS without external profile....
> I would like to use just one profile, and route incoming calls based on DID-s and context,
> just like in asterisk...
> But FS always logs:
> 
> sofia.c:8834 sofia/internal/13245678 at 1.2.3.4 <mailto:13245678 at 1.2.3.4> receiving invite from 1.2.3.4:5060 <http://1.2.3.4:5060/> version: 1.4.14 sofia.c:9001 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth.
> 
> sofia_reg.c:2827 Can't find user [2000 at 192.168.101.44 <mailto:2000 at 192.168.101.44>] from 1.2.3.4
> You must define a domain called '192.168.101.44' in your directory and add a user with the id="2000" attribute
> and you must configure your device to use the proper domain in it's authentication credentials.
> 
> I do not need any users at all...
> 
> I just need to recive calls from a gateway and route them to specific context/extension in XML dialplan...
> 
> How to do thet?
> What is the context parameter of the gateway is used for, if not for that purpuse?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/c425d711/attachment.html 

From Sharath.Kumar at meZocliq.com  Mon Mar  9 18:34:36 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 9 Mar 2015 15:34:36 +0000
Subject: [Freeswitch-users] mod_conference and outbound call status
Message-ID: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>

Hi,

I am using mod_conference to call out a bunch of users.  I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not.  Any ideas ?

Thanks
Sharath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/cefa6a69/attachment.html 

From mike at jerris.com  Mon Mar  9 19:03:39 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 12:03:39 -0400
Subject: [Freeswitch-users] mod_conference and outbound call status
In-Reply-To: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>

The conference has events and conference list commands to see these calls.

> On Mar 9, 2015, at 11:34 AM, Sharath Kumar <Sharath.Kumar at meZocliq.com> wrote:
> 
> Hi,
>  
> I am using mod_conference to call out a bunch of users.  I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not.  Any ideas ? 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/b71c057b/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Mon Mar  9 19:17:45 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 9 Mar 2015 16:17:45 +0000
Subject: [Freeswitch-users] mod_conference and outbound call status
In-Reply-To: <57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>
References: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>
Message-ID: <CY1PR0501MB109850AA98653B6F260489B1841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>


You mean this ?
Example 'advertise' Event via mod_event_multicast
Advertise event

So I need an XMPP server now!


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, March 09, 2015 12:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_conference and outbound call status

The conference has events and conference list commands to see these calls.

On Mar 9, 2015, at 11:34 AM, Sharath Kumar <Sharath.Kumar at meZocliq.com<mailto:Sharath.Kumar at meZocliq.com>> wrote:

Hi,

I am using mod_conference to call out a bunch of users.  I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not.  Any ideas ?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/1c660d86/attachment.html 

From victor.chukalovskiy at gmail.com  Mon Mar  9 19:29:57 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Mon, 09 Mar 2015 12:29:57 -0400
Subject: [Freeswitch-users] What exactly enable_soa = false changes?
Message-ID: <54FDCA85.3060508@gmail.com>

Good day,

I find documentation regarding enable_soa param very scarce, so looking 
for some clarifications here.

What exactly changes in FS behavior with enable_soa = false?

For example, I observe that with soa disabled, FS receiving "183" with 
SDP on one leg substitutes it with plain "180 Ringing" on another leg.
This is not correct, but I'm not sure if it's a bug or an expected 
outcome of disabling soa.

Thanks!!
-Victor


From fs-list at communicatefreely.net  Mon Mar  9 19:46:44 2015
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 09 Mar 2015 12:46:44 -0400
Subject: [Freeswitch-users] any handset recommendations that operate
 like polycom+cisco?
In-Reply-To: <CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>
References: <54B82BA0.20201@mst.edu>	<CAKzWOxUzsAk=EVDRw5+5Te4YQ3SwrQydGED4XiL3Y_VYbaPX0A@mail.gmail.com>	<54C40829.9060501@mst.edu>
	<CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>
Message-ID: <54FDCE74.60407@communicatefreely.net>

Hello,

Just saw this.

If you are still looking, the Aastra / Mitel 6800 series can do 
something like 20 SIP accounts, and you can put line keys on the 
expansion modules.  I'm not sure if that is enough, but if it can cover 
you for the transition, that might get you by.

-Tim



On 2015-01-25 05:13 PM, Michael Collins wrote:
>
>
> On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger <nneul at mst.edu
> <mailto:nneul at mst.edu>> wrote:
>
>     Personally I think it's nuts... but we have a number of
>     secretary/admin/receptionist users with Cisco expansion modules
>     that have the shared lines of all of the people in their department
>     on them. (i.e. a 7940/7960 plus the module). Usually
>     with some portion of the other lines set to just flash and not
>     audibly ring. While I'd expect that in most cases they
>     really would be sufficient with busy lamp, sometimes they do use it
>     to answer arbitrary calls for faculty that are out
>     of the office/etc.
>
>     With the transitioned cisco phones on FS/mod_skinny - it works the
>     same way, however we're wanting to position ourselves
>     with suitable replacements, particularly for any departments that
>     want more than bare bones functionality.
>
>     With the polycom phones, it appears to also work that way where you
>     can have a sip account for every line key if you
>     want - even including the expansion modules.
>
>     However, on the Yealink phones (got looking at them cause of the
>     T46G I won at ClueCon) we found the number of accounts
>     very limited.
>
>     It turns out that with the latest firmware (73.x) on the Yealink
>     units the count is increased on a number of the models
>     (to 16 on the T46 for example). The problem is that with the middle
>     tier ones that you'd add an expansion module to - it
>     doesn't really get you anything. If your base phone is limited to 6
>     accounts, adding the expansion module ONLY gets you
>     busy-lamp or speed dials.
>
>     We're working on getting the users "converted" to not using full
>     lines wherever possible, but still want options open.
>
>     -- Nathan
>
>
> Thanks for the explanation. I share your feelings about the T46. I love
> that phone but hate the fact that you only get 6 SIP accounts. (Glad to
> hear that they added more in a recent firmware - I'll test that out at
> some point...)
>
> If you find a solution other than the Cisco one I would be interested in
> hearing about it.
>
> Thanks,
> Michael
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Mon Mar  9 21:18:21 2015
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Mar 2015 11:18:21 -0700
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
References: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
Message-ID: <CAKzWOxXAnfh47aLdL0OWik2jYeRH6c7tO5zCSOwq3YZCrcQBDA@mail.gmail.com>

On Sat, Mar 7, 2015 at 9:03 AM, Paul Atreides <
paul.atreides83 at googlemail.com> wrote:

> When I do a blind transfer then I want freeswitch to call back the origin
> who initiated the call.
> But I am not able the capture the transfer event?
>

In other words, if user A did a blind x-fer of caller C to user B and user
B doesn't answer (for whatever failure reason) then caller C would start
ringing back to user A? Just making sure we understand the scope of the
feature you're implementing.

How does the GXP do the x-fer? Some kind of hook-flash and DTMF code? Can
you pastebin the dialplan that the transferor uses when sending the call?

-MC


> <condition field="destination_number" expression="^att_xfer$">
> <condition field="destination_number" expression="^is_transfer$">
>
> They seem do be ignored by the dialplan. Is there a list what kind of
> values destionation_number can have besides the called numbers?
>
>
> I am doing the transfer with a grandstream gxp2140
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/17ac43f4/attachment.html 

From msc at freeswitch.org  Mon Mar  9 21:26:08 2015
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Mar 2015 11:26:08 -0700
Subject: [Freeswitch-users] Incoming call - ring call group - then
	voicemail
In-Reply-To: <CAK+KT38GMxQ4N2-6e_aocAJo6+bhLyt9emvH1xEz3f-YyYjnEQ@mail.gmail.com>
References: <CAK+KT38GMxQ4N2-6e_aocAJo6+bhLyt9emvH1xEz3f-YyYjnEQ@mail.gmail.com>
Message-ID: <CAKzWOxUnYhNGhk56zQ4eeyyXiQ-HqBqy15iahos3hA8aaZjjbg@mail.gmail.com>

On Sun, Mar 8, 2015 at 11:49 PM, Richard Mace <richard.mace at gmail.com>
wrote:

> Hi All,
> I am trying to get an incoming call to ring the group 1000 in the domain
> wph.co.uk and then go into the voicemail for 200. Is the following code
> close because it doesn't seem to work?
>
>       <action application="set" data="call_timeout=20"/>
>       <action application="set" data="continue_on_fail=true"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="bridge" data="sofia/wph.co.uk/group/1000"/>
>       <action application="answer"/>
>       <action application="voicemail" data="default wph.co.uk 200?/>
>
> Thanks in advance
>
> Did you verify that the group "1000" is doing what you think it is? You
can verify the dialstring that will be generated by going to fs_cli and
doing:

group_call 1000

Verify that it really does produce a dialstring that calls everyone in the
group.

-MC


> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/ccbd6f03/attachment-0001.html 

From msc at freeswitch.org  Mon Mar  9 21:44:06 2015
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Mar 2015 11:44:06 -0700
Subject: [Freeswitch-users] mod_conference and outbound call status
In-Reply-To: <CY1PR0501MB109850AA98653B6F260489B1841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>
	<CY1PR0501MB109850AA98653B6F260489B1841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAKzWOxXg_+73PQb-XRxuhrA3p+B89ZpfpnTX2aBL_X+AmZ=rUQ@mail.gmail.com>

On Mon, Mar 9, 2015 at 9:17 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>
>
> You mean this ?
>
> *Example 'advertise' Event via mod_event_multicast*
>
> *Advertise event*
>
>
>
> So I need an XMPP server now!
>
Whether you need an XMPP server is really up to you. How will you plan to
alert the moderator who originally called the bridge? FreeSWITCH is
designed with all sorts of APIs, hooks, and events that let you find out
"what happened" or "is happening" for pretty much anything that transpires.

How elegant does this solution need to be? That will most likely determine
what other servers, etc. you will need to bring into production.

-MC


>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Monday, March 09, 2015 12:04 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_conference and outbound call status
>
>
>
> The conference has events and conference list commands to see these calls.
>
>
>
>  On Mar 9, 2015, at 11:34 AM, Sharath Kumar <Sharath.Kumar at meZocliq.com>
> wrote:
>
>
>
> Hi,
>
>
>
> I am using mod_conference to call out a bunch of users.  I would like to
> know how does the original caller(ie the moderator who called the bridge)
> come to know of the status of these calls, say I want to display the status
> of whether or not the users accepted the invitation or not.  Any ideas ?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/2b5e4b33/attachment.html 

From mike at jerris.com  Mon Mar  9 21:52:19 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Mar 2015 14:52:19 -0400
Subject: [Freeswitch-users] mod_conference and outbound call status
In-Reply-To: <CAKzWOxXg_+73PQb-XRxuhrA3p+B89ZpfpnTX2aBL_X+AmZ=rUQ@mail.gmail.com>
References: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>
	<CY1PR0501MB109850AA98653B6F260489B1841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKzWOxXg_+73PQb-XRxuhrA3p+B89ZpfpnTX2aBL_X+AmZ=rUQ@mail.gmail.com>
Message-ID: <F1616CC0-BC73-4A56-86AF-F73E1A784E52@jerris.com>

What Michael Said.. with notes.. multicast is one of the ways you can access events, but probably not the one you want, and it has nothing at all to do with xmpp.  How you want to interact with FreeSWITCH is a much bigger discussion.  How would you like to signal this information? There is a sip standard for this as well, rfc4579.  You can enable this in conference with the flag, and this would require you to have a sip device that supports this as well.


> On Mar 9, 2015, at 2:44 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> 
> 
> On Mon, Mar 9, 2015 at 9:17 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com <mailto:Sharath.Kumar at mezocliq.com>> wrote:
>  
> 
> You mean this ?
> 
> Example 'advertise' Event via mod_event_multicast
> 
> Advertise event
> 
>  
> 
> So I need an XMPP server now!
> 
> Whether you need an XMPP server is really up to you. How will you plan to alert the moderator who originally called the bridge? FreeSWITCH is designed with all sorts of APIs, hooks, and events that let you find out "what happened" or "is happening" for pretty much anything that transpires.
> 
> How elegant does this solution need to be? That will most likely determine what other servers, etc. you will need to bring into production.
> 
> -MC
>  
> 
>  
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Monday, March 09, 2015 12:04 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_conference and outbound call status
> 
>  
> 
> The conference has events and conference list commands to see these calls.
> 
>  
> 
> On Mar 9, 2015, at 11:34 AM, Sharath Kumar <Sharath.Kumar at meZocliq.com <mailto:Sharath.Kumar at meZocliq.com>> wrote:
> 
>  
> 
> Hi,
> 
>  
> 
> I am using mod_conference to call out a bunch of users.  I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not.  Any ideas ? 
> 
>  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/9ad1b947/attachment.html 

From Sharath.Kumar at meZocliq.com  Mon Mar  9 21:54:03 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 9 Mar 2015 18:54:03 +0000
Subject: [Freeswitch-users] mod_conference and outbound call status
In-Reply-To: <CAKzWOxXg_+73PQb-XRxuhrA3p+B89ZpfpnTX2aBL_X+AmZ=rUQ@mail.gmail.com>
References: <CY1PR0501MB10983BBA622D0CA6F5B6D88A841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<57918D04-EDD5-4DF3-AC9B-CAF4D25A5293@jerris.com>
	<CY1PR0501MB109850AA98653B6F260489B1841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKzWOxXg_+73PQb-XRxuhrA3p+B89ZpfpnTX2aBL_X+AmZ=rUQ@mail.gmail.com>
Message-ID: <CY1PR0501MB109851053ACBBDD805A93139841B0@CY1PR0501MB1098.namprd05.prod.outlook.com>

Okay thank you.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, March 09, 2015 2:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_conference and outbound call status



On Mon, Mar 9, 2015 at 9:17 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:

You mean this ?
Example 'advertise' Event via mod_event_multicast
Advertise event

So I need an XMPP server now!
Whether you need an XMPP server is really up to you. How will you plan to alert the moderator who originally called the bridge? FreeSWITCH is designed with all sorts of APIs, hooks, and events that let you find out "what happened" or "is happening" for pretty much anything that transpires.
How elegant does this solution need to be? That will most likely determine what other servers, etc. you will need to bring into production.

-MC



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
Sent: Monday, March 09, 2015 12:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_conference and outbound call status

The conference has events and conference list commands to see these calls.

On Mar 9, 2015, at 11:34 AM, Sharath Kumar <Sharath.Kumar at meZocliq.com<mailto:Sharath.Kumar at meZocliq.com>> wrote:

Hi,

I am using mod_conference to call out a bunch of users.  I would like to know how does the original caller(ie the moderator who called the bridge) come to know of the status of these calls, say I want to display the status of whether or not the users accepted the invitation or not.  Any ideas ?


_________________________________________________________________________
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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/6b427071/attachment-0001.html 

From martinschoepfer at gmx.de  Mon Mar  9 22:20:06 2015
From: martinschoepfer at gmx.de (=?iso-8859-1?Q?Martin_Sch=F6pfer_=28GMX=29?=)
Date: Mon, 9 Mar 2015 20:20:06 +0100
Subject: [Freeswitch-users] Freeswitch --> Faxing issue
Message-ID: <009d01d05a9e$0d8ef670$28ace350$@gmx.de>

Hello,

 

He told me so!

 

Thanks

 

 

>Step 1.  Does your provider support T.38?

 

>>On Mar 7, 2015, at 5:47 PM, Martin Sch?pfer (GMX) <
<mailto:martinschoepfer at gmx.de> martinschoepfer at gmx.de> wrote:

>> 

>>Hello,

 >>

>>can anyone help me by my config for faxing with freeswitch. I have testet
al functions of spandsp also to set the software Modems active and receive
faxes with hylafax but nothing works.

 >>

>>That?s the function I want to get

 >>

>>inbound

>>
 Sip-provider ? Freeswitch ? Cisco SPA3102 Line 1(T.28 enabled) ? hp fax
or another 

>>For now i?m getting the message ?no fax detected? from my faxing maschine

>>Outbound

>>Hp fax or another ? Cisco SPA3102 Line 1 ? Freeswitch ? Sip-provider 
.

>>There I get the message phone busy always ?I know the other faxing
maschine isn busy that?s my second?

>> 

>>The softmodem are also always busy I can?t check the function by cu ?l FS
because it tells me line in use.

>> 

>>Freeswitch Version is the latest master branch of 1.4

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/15d7f4ea/attachment.html 

From krice at freeswitch.org  Mon Mar  9 22:24:08 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 09 Mar 2015 19:24:08 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
	February 28th-March 6th
Message-ID: <54fdf35820c89_7f9712e733031048@resque-worker-ip-10-186-161-41.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/18vTmtt
FreeSWITCH Week in Review (Master Branch) February 28th-March 6th
Hello, again. This passed week in the FreeSWITCH master branch we had 18 commits. The features for this week include: updating Windows build to use flite-2.0.0-release and updating mod_mongo driver to 1.1.0 .

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7149 Update Windows build to use flite-2.0.0-release

FS-7346 Update mod_mongo driver to 1.1.0

The following bugs were squashed:

FS-7339 Move creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time

FS-7342 Fixed a crash regression in mod_conference caused by FS-7230

FS-7340 Remove json-c dependency in favor of our own json code

FS-7350 Add ?enable-address-sanitizer configure flag to enable clang address sanitizer

FS-6758 Revert a commit from FS-6758 fixing hold dropping calls on Skinny Cisco 7961G

FS-7305 Fix for making embedded versions of FS start up and shutdown faster like in the case of tone2wav

FS-5570 Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/373e75a3/attachment.html 

From brian at freeswitch.org  Mon Mar  9 22:43:51 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 9 Mar 2015 14:43:51 -0500
Subject: [Freeswitch-users] any handset recommendations that operate
	like polycom+cisco?
In-Reply-To: <54FDCE74.60407@communicatefreely.net>
References: <54B82BA0.20201@mst.edu>
	<CAKzWOxUzsAk=EVDRw5+5Te4YQ3SwrQydGED4XiL3Y_VYbaPX0A@mail.gmail.com>
	<54C40829.9060501@mst.edu>
	<CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>
	<54FDCE74.60407@communicatefreely.net>
Message-ID: <CAEJMVkCPY1i2k-yyri4MaziN_d=MrHW+HMrxqpVSAJ=HcwHh2A@mail.gmail.com>

The Fanvil phones are cheap and usable.

On Mon, Mar 9, 2015 at 11:46 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> Just saw this.
>
> If you are still looking, the Aastra / Mitel 6800 series can do
> something like 20 SIP accounts, and you can put line keys on the
> expansion modules.  I'm not sure if that is enough, but if it can cover
> you for the transition, that might get you by.
>
> -Tim
>
>
>
> On 2015-01-25 05:13 PM, Michael Collins wrote:
> >
> >
> > On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger <nneul at mst.edu
> > <mailto:nneul at mst.edu>> wrote:
> >
> >     Personally I think it's nuts... but we have a number of
> >     secretary/admin/receptionist users with Cisco expansion modules
> >     that have the shared lines of all of the people in their department
> >     on them. (i.e. a 7940/7960 plus the module). Usually
> >     with some portion of the other lines set to just flash and not
> >     audibly ring. While I'd expect that in most cases they
> >     really would be sufficient with busy lamp, sometimes they do use it
> >     to answer arbitrary calls for faculty that are out
> >     of the office/etc.
> >
> >     With the transitioned cisco phones on FS/mod_skinny - it works the
> >     same way, however we're wanting to position ourselves
> >     with suitable replacements, particularly for any departments that
> >     want more than bare bones functionality.
> >
> >     With the polycom phones, it appears to also work that way where you
> >     can have a sip account for every line key if you
> >     want - even including the expansion modules.
> >
> >     However, on the Yealink phones (got looking at them cause of the
> >     T46G I won at ClueCon) we found the number of accounts
> >     very limited.
> >
> >     It turns out that with the latest firmware (73.x) on the Yealink
> >     units the count is increased on a number of the models
> >     (to 16 on the T46 for example). The problem is that with the middle
> >     tier ones that you'd add an expansion module to - it
> >     doesn't really get you anything. If your base phone is limited to 6
> >     accounts, adding the expansion module ONLY gets you
> >     busy-lamp or speed dials.
> >
> >     We're working on getting the users "converted" to not using full
> >     lines wherever possible, but still want options open.
> >
> >     -- Nathan
> >
> >
> > Thanks for the explanation. I share your feelings about the T46. I love
> > that phone but hate the fact that you only get 6 SIP accounts. (Glad to
> > hear that they added more in a recent firmware - I'll test that out at
> > some point...)
> >
> > If you find a solution other than the Cisco one I would be interested in
> > hearing about it.
> >
> > Thanks,
> > Michael
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/02ebf0d1/attachment-0001.html 

From davidwaf at gmail.com  Tue Mar 10 00:12:59 2015
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 9 Mar 2015 23:12:59 +0200
Subject: [Freeswitch-users] Chat: invalid profile
In-Reply-To: <CAEJMVkBB6q_TU1kJKORcx51sVWANW8Fg+2pTgZBpRi-qNxcc+Q@mail.gmail.com>
References: <CAK9Ws2dTyUrKsDsHRnP1PNq2Vd+zPCH+k0yqRGbhE2h7zzNhQA@mail.gmail.com>
	<CAEJMVkBB6q_TU1kJKORcx51sVWANW8Fg+2pTgZBpRi-qNxcc+Q@mail.gmail.com>
Message-ID: <CAK9Ws2fOAOc-0nj6MmuZ1wUhM0+vN6yobmrd-OSGAbcVmRpmHQ@mail.gmail.com>

Thank you. That worked perfectly !

On Mon, Mar 9, 2015 at 4:07 PM, Brian West <brian at freeswitch.org> wrote:

> alias the domain or IP to the profile so it can know where to route the
> message.
>
> On Mon, Mar 9, 2015 at 5:54 AM, David Wafula <davidwaf at gmail.com> wrote:
>
>> Hello all,
>> I used to have the following command working with default installation.
>>
>>
>> chat sip|1000 at xxx.xxx.xxx.xxx|1001 at xxx.xxx.xxx.xxx|hello sopranos
>>
>>
>> After binding the chatplan to xml curl, the command no longer works, i
>> get :
>>
>> 2015-03-09 12:47:41.238102 [ERR] sofia_presence.c:200 Chat proto [global]
>> from [1000 at xxx.xxx.xxx.xxx]
>> to [1000 at xxx.xxx.xxx.xxx]
>> hello sopranos
>> Invalid Profile xxx.xxx.xxx.xxx
>>
>>  xxx.xxx.xxx.xxx is the same IP through.  Am not sure where to start
>> looking. Help.
>>
>> --
>> David Wafula
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/0243f9f8/attachment.html 

From nneul at mst.edu  Tue Mar 10 00:30:49 2015
From: nneul at mst.edu (Nathan Neulinger)
Date: Mon, 09 Mar 2015 16:30:49 -0500
Subject: [Freeswitch-users] any handset recommendations that operate
 like polycom+cisco?
In-Reply-To: <CAEJMVkCPY1i2k-yyri4MaziN_d=MrHW+HMrxqpVSAJ=HcwHh2A@mail.gmail.com>
References: <54B82BA0.20201@mst.edu>	<CAKzWOxUzsAk=EVDRw5+5Te4YQ3SwrQydGED4XiL3Y_VYbaPX0A@mail.gmail.com>	<54C40829.9060501@mst.edu>	<CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>	<54FDCE74.60407@communicatefreely.net>
	<CAEJMVkCPY1i2k-yyri4MaziN_d=MrHW+HMrxqpVSAJ=HcwHh2A@mail.gmail.com>
Message-ID: <54FE1109.2060405@mst.edu>

Looks like the Fanvil phones have the same underlying issue as the Yealinks unfortunately - # of sip accounts limited 
significantly below # of buttons.

-- Nathan

On 03/09/2015 02:43 PM, Brian West wrote:
> The Fanvil phones are cheap and usable.
>
> On Mon, Mar 9, 2015 at 11:46 AM, Tim St. Pierre <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> wrote:
>
>     Hello,
>
>     Just saw this.
>
>     If you are still looking, the Aastra / Mitel 6800 series can do
>     something like 20 SIP accounts, and you can put line keys on the
>     expansion modules.  I'm not sure if that is enough, but if it can cover
>     you for the transition, that might get you by.
>
>     -Tim
>
>
>
>     On 2015-01-25 05:13 PM, Michael Collins wrote:
>     >
>     >
>     > On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>
>      > <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>> wrote:
>      >
>      >     Personally I think it's nuts... but we have a number of
>      >     secretary/admin/receptionist users with Cisco expansion modules
>      >     that have the shared lines of all of the people in their department
>      >     on them. (i.e. a 7940/7960 plus the module). Usually
>      >     with some portion of the other lines set to just flash and not
>      >     audibly ring. While I'd expect that in most cases they
>      >     really would be sufficient with busy lamp, sometimes they do use it
>      >     to answer arbitrary calls for faculty that are out
>      >     of the office/etc.
>      >
>      >     With the transitioned cisco phones on FS/mod_skinny - it works the
>      >     same way, however we're wanting to position ourselves
>      >     with suitable replacements, particularly for any departments that
>      >     want more than bare bones functionality.
>      >
>      >     With the polycom phones, it appears to also work that way where you
>      >     can have a sip account for every line key if you
>      >     want - even including the expansion modules.
>      >
>      >     However, on the Yealink phones (got looking at them cause of the
>      >     T46G I won at ClueCon) we found the number of accounts
>      >     very limited.
>      >
>      >     It turns out that with the latest firmware (73.x) on the Yealink
>      >     units the count is increased on a number of the models
>      >     (to 16 on the T46 for example). The problem is that with the middle
>      >     tier ones that you'd add an expansion module to - it
>      >     doesn't really get you anything. If your base phone is limited to 6
>      >     accounts, adding the expansion module ONLY gets you
>      >     busy-lamp or speed dials.
>      >
>      >     We're working on getting the users "converted" to not using full
>      >     lines wherever possible, but still want options open.
>      >
>      >     -- Nathan
>      >
>      >
>      > Thanks for the explanation. I share your feelings about the T46. I love
>      > that phone but hate the fact that you only get 6 SIP accounts. (Glad to
>      > hear that they added more in a recent firmware - I'll test that out at
>      > some point...)
>      >
>      > If you find a solution other than the Cisco one I would be interested in
>      > hearing about it.
>      >
>      > Thanks,
>      > Michael
>      >
>      >
>      > _________________________________________________________________________
>      > 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://confluence.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://confluence.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
>
>
>
>
> --
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 steveayre at gmail.com  Tue Mar 10 02:12:01 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 Mar 2015 23:12:01 +0000
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <120E4BAE-F8A6-4404-97E7-B1D2FF3B771D@jerris.com>
References: <D12246D2.98025%krice@freeswitch.org>
	<E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>
	<CAAyX+KEksE4a==CYZ4oc9BtuKSezBx8fx9pVxqC8t4nwJaQ=9g@mail.gmail.com>
	<120E4BAE-F8A6-4404-97E7-B1D2FF3B771D@jerris.com>
Message-ID: <CAFiqYunnMKvwztvD5N3_npSrubwG2osOUQ-zoN0OBwv-S-Yoiw@mail.gmail.com>

Plus it's rather annoying to do so (rewrite history). The identifier of
each commit is a hash computed from the content of the commit plus the
metadata which includes the authors. Changing the author would change the
identifier of the commit. That then changes the identifier of every commit
afterwards. That then breaks every checkout / fork based off the tree as
they no longer know where they are forked from. And since identifiers have
all been rewritten we would no longer know what version you were running,
or what version bug reports were reported against.

(that's why git makes it easy to amend your latest uncommitted commit
message but rather difficult to edit any others)




On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:

> We are not rewriting history to fix this so it doesn't really matter who
> is right or wrong.
>
> Mike
>
> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
> I'm curious about this one... If git fsck complains about the issue, what
> is the justification for saying that Github is broken? How were these
> commits created with a format that git itself complains about?
>
> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>
>> I switched to bitbucket.org just for the FreeSWITCH repo to work around
>> this.
>>
>>
>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>
>> This is a known issue with github and will not be fixed
>>
>>
>>
>> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com> wrote:
>>
>> Hi,
>>
>> I'm trying to push the freeswitch git repo to my github, but I get the
>> following error
>>
>>
>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>> format - expected 'committer' line
>> remote: fatal: Error in object
>> channel_by_id: 0: bad id: channel free
>> Received window adjust for non-open channel 0.
>> error: pack-objects died of signal 13
>>
>> This is caused by having multiple authors on a commit (which in general
>> is not allowed by git) and github verifies the commits and rejects it.
>>
>> here is the output of git fsck
>>
>> Checking object directories: 100% (256/256), done.
>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>> - expected 'committer' line
>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>> - expected 'committer' line
>> Checking objects: 100% (254227/254227), done.
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/a8d7ff3c/attachment-0001.html 

From steveayre at gmail.com  Tue Mar 10 02:15:23 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 9 Mar 2015 23:15:23 +0000
Subject: [Freeswitch-users] What exactly enable_soa = false changes?
In-Reply-To: <54FDCA85.3060508@gmail.com>
References: <54FDCA85.3060508@gmail.com>
Message-ID: <CAFiqYumhpce6YVE7s4OU67BUtbq-pKrp=cw46FNRAK4raOdTfQ@mail.gmail.com>

The high level answer is it controls whether the soa portion of the sofia
stack is used (http://sofia-sip.sourceforge.net/refdocs/soa/), disabling it
would use an alternative implementation. Obviously there are differences
between the implementations from what you've observed. I can't tell you
exactly what though.

On 9 March 2015 at 16:29, Victor Chukalovskiy <victor.chukalovskiy at gmail.com
> wrote:

> Good day,
>
> I find documentation regarding enable_soa param very scarce, so looking
> for some clarifications here.
>
> What exactly changes in FS behavior with enable_soa = false?
>
> For example, I observe that with soa disabled, FS receiving "183" with
> SDP on one leg substitutes it with plain "180 Ringing" on another leg.
> This is not correct, but I'm not sure if it's a bug or an expected
> outcome of disabling soa.
>
> Thanks!!
> -Victor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/217be86f/attachment.html 

From ben at langfeld.co.uk  Tue Mar 10 03:57:39 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 9 Mar 2015 21:57:39 -0300
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAFiqYunnMKvwztvD5N3_npSrubwG2osOUQ-zoN0OBwv-S-Yoiw@mail.gmail.com>
References: <D12246D2.98025%krice@freeswitch.org>
	<E13D3034-AC5B-4DD9-BF18-8B3BF3681100@gmail.com>
	<CAAyX+KEksE4a==CYZ4oc9BtuKSezBx8fx9pVxqC8t4nwJaQ=9g@mail.gmail.com>
	<120E4BAE-F8A6-4404-97E7-B1D2FF3B771D@jerris.com>
	<CAFiqYunnMKvwztvD5N3_npSrubwG2osOUQ-zoN0OBwv-S-Yoiw@mail.gmail.com>
Message-ID: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>

I understand very well why rewriting history is undesirable and how git
works. What I wonder is what process was used to convince git to create a
commit which it would later say is invalid. As I said, I'm curious.

On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:

> Plus it's rather annoying to do so (rewrite history). The identifier of
> each commit is a hash computed from the content of the commit plus the
> metadata which includes the authors. Changing the author would change the
> identifier of the commit. That then changes the identifier of every commit
> afterwards. That then breaks every checkout / fork based off the tree as
> they no longer know where they are forked from. And since identifiers have
> all been rewritten we would no longer know what version you were running,
> or what version bug reports were reported against.
>
> (that's why git makes it easy to amend your latest uncommitted commit
> message but rather difficult to edit any others)
>
>
>
>
> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>
>> We are not rewriting history to fix this so it doesn't really matter who
>> is right or wrong.
>>
>> Mike
>>
>> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>> I'm curious about this one... If git fsck complains about the issue, what
>> is the justification for saying that Github is broken? How were these
>> commits created with a format that git itself complains about?
>>
>> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>>
>>> I switched to bitbucket.org just for the FreeSWITCH repo to work around
>>> this.
>>>
>>>
>>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>> This is a known issue with github and will not be fixed
>>>
>>>
>>>
>>> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com> wrote:
>>>
>>> Hi,
>>>
>>> I'm trying to push the freeswitch git repo to my github, but I get the
>>> following error
>>>
>>>
>>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>>> format - expected 'committer' line
>>> remote: fatal: Error in object
>>> channel_by_id: 0: bad id: channel free
>>> Received window adjust for non-open channel 0.
>>> error: pack-objects died of signal 13
>>>
>>> This is caused by having multiple authors on a commit (which in general
>>> is not allowed by git) and github verifies the commits and rejects it.
>>>
>>> here is the output of git fsck
>>>
>>> Checking object directories: 100% (256/256), done.
>>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>>> - expected 'committer' line
>>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>>> - expected 'committer' line
>>> Checking objects: 100% (254227/254227), done.
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/35181a54/attachment.html 

From krice at freeswitch.org  Tue Mar 10 05:21:11 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 09 Mar 2015 20:21:11 -0600
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>
Message-ID: <D123B137.9810D%krice@freeswitch.org>

Git itself doesn?t say this is an invalid commit (we use stash/git and its
perfectly fine with this)... There is a git-lint process that github uses
and it rejects this commit


On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:

> I understand very well why rewriting history is undesirable and how git works.
> What I wonder is what process was used to convince git to create a commit
> which it would later say is invalid. As I said, I'm curious.
> 
> On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:
>> Plus it's rather annoying to do so (rewrite history). The identifier of each
>> commit is a hash computed from the content of the commit plus the metadata
>> which includes the authors. Changing the author would change the identifier
>> of the commit. That then changes the identifier of every commit afterwards.
>> That then breaks every checkout / fork based off the tree as they no longer
>> know where they are forked from. And since identifiers have all been
>> rewritten we would no longer know what version you were running, or what
>> version bug reports were reported against.
>> 
>> (that's why git makes it easy to amend your latest uncommitted commit message
>> but rather difficult to edit any others)
>> 
>> 
>> 
>> 
>> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>>> We are not rewriting history to fix this so it doesn't really matter who is
>>> right or wrong.
>>> 
>>> Mike
>>> 
>>>> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>>> 
>>>> I'm curious about this one... If git fsck complains about the issue, what
>>>> is the justification for saying that Github is broken? How were these
>>>> commits created with a format that git itself complains about?
>>>> 
>>>> On 9 March 2015 at 09:01,??<cmrienzo at gmail.com>?wrote:
>>>>> I switched to?bitbucket.org <http://bitbucket.org/> ?just for the
>>>>> FreeSWITCH repo to work around this.?
>>>>> 
>>>>> 
>>>>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>>>> 
>>>>>> This is a known issue with github and will not be fixed?
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com
>>>>>> <http://aronp at guaranteedplus.com/> > wrote:
>>>>>> 
>>>>>>> Hi,
>>>>>>> 
>>>>>>> I'm trying to push the freeswitch git repo to my github, but I get the
>>>>>>> following error
>>>>>>> 
>>>>>>> 
>>>>>>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>>>>>>> format - expected 'committer' line
>>>>>>> remote: fatal: Error in object
>>>>>>> channel_by_id: 0: bad id: channel free
>>>>>>> Received window adjust for non-open channel 0.
>>>>>>> error: pack-objects died of signal 13
>>>>>>> 
>>>>>>> This is caused by having multiple authors on a commit (which in general
>>>>>>> is not allowed by git) and github verifies the commits and rejects it.
>>>>>>> 
>>>>>>> here is the output of git fsck
>>>>>>> 
>>>>>>> Checking object directories: 100% (256/256), done.
>>>>>>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>>>>>>> - expected 'committer' line
>>>>>>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>>>>>>> - expected 'committer' line
>>>>>>> Checking objects: 100% (254227/254227), done.
>>>>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/dab13a6a/attachment-0001.html 

From montecillodavid.spingine at gmail.com  Tue Mar 10 04:26:54 2015
From: montecillodavid.spingine at gmail.com (David Montecillo)
Date: Tue, 10 Mar 2015 09:26:54 +0800
Subject: [Freeswitch-users] issue terminating outbound calls in
	freeswitch
In-Reply-To: <bf6fe30133e2aced7071b7608d0e7d45@spingine.com>
References: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>
	<B31DB44E-8A50-4F9D-A4B8-508B1FCDB170@jerris.com>
	<bf6fe30133e2aced7071b7608d0e7d45@spingine.com>
Message-ID: <CAN3R+ShXKxOdj6WBD11foTxRFJ==Jv0ePp++bLT6S=7R5VkB3Q@mail.gmail.com>

It's a hardware where you can insert multiple sims but Im just using the
single version(http://www.dbltek.com/products/goip-1.html). We use it as a
gsm gateway for outbound calls and freeswitch setup instructions are
available in the freeswitch pagehttps://freeswitch.org/
confluence/display/FREESWITCH/Goip+HowTo

Do you have an idea whats causing the termination problem?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/25daa722/attachment.html 

From brian at freeswitch.org  Tue Mar 10 04:28:21 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 9 Mar 2015 20:28:21 -0500
Subject: [Freeswitch-users] any handset recommendations that operate
	like polycom+cisco?
In-Reply-To: <54FE1109.2060405@mst.edu>
References: <54B82BA0.20201@mst.edu>
	<CAKzWOxUzsAk=EVDRw5+5Te4YQ3SwrQydGED4XiL3Y_VYbaPX0A@mail.gmail.com>
	<54C40829.9060501@mst.edu>
	<CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>
	<54FDCE74.60407@communicatefreely.net>
	<CAEJMVkCPY1i2k-yyri4MaziN_d=MrHW+HMrxqpVSAJ=HcwHh2A@mail.gmail.com>
	<54FE1109.2060405@mst.edu>
Message-ID: <CAEJMVkDUw2vXRZG0UeurNiB1MTk7V894mJvJUegDTymRjx1Obg@mail.gmail.com>

Buttons, WHO needs buttons, you have two ears and one mouth, till that
condition changes and people start sprouting more mouths and ears I suspect
a low button count is sufficient.

On Mon, Mar 9, 2015 at 4:30 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> Looks like the Fanvil phones have the same underlying issue as the
> Yealinks unfortunately - # of sip accounts limited
> significantly below # of buttons.
>
> -- Nathan
>
> On 03/09/2015 02:43 PM, Brian West wrote:
> > The Fanvil phones are cheap and usable.
> >
> > On Mon, Mar 9, 2015 at 11:46 AM, Tim St. Pierre <
> fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> > wrote:
> >
> >     Hello,
> >
> >     Just saw this.
> >
> >     If you are still looking, the Aastra / Mitel 6800 series can do
> >     something like 20 SIP accounts, and you can put line keys on the
> >     expansion modules.  I'm not sure if that is enough, but if it can
> cover
> >     you for the transition, that might get you by.
> >
> >     -Tim
> >
> >
> >
> >     On 2015-01-25 05:13 PM, Michael Collins wrote:
> >     >
> >     >
> >     > On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger <nneul at mst.edu
> <mailto:nneul at mst.edu>
> >      > <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>> wrote:
> >      >
> >      >     Personally I think it's nuts... but we have a number of
> >      >     secretary/admin/receptionist users with Cisco expansion
> modules
> >      >     that have the shared lines of all of the people in their
> department
> >      >     on them. (i.e. a 7940/7960 plus the module). Usually
> >      >     with some portion of the other lines set to just flash and not
> >      >     audibly ring. While I'd expect that in most cases they
> >      >     really would be sufficient with busy lamp, sometimes they do
> use it
> >      >     to answer arbitrary calls for faculty that are out
> >      >     of the office/etc.
> >      >
> >      >     With the transitioned cisco phones on FS/mod_skinny - it
> works the
> >      >     same way, however we're wanting to position ourselves
> >      >     with suitable replacements, particularly for any departments
> that
> >      >     want more than bare bones functionality.
> >      >
> >      >     With the polycom phones, it appears to also work that way
> where you
> >      >     can have a sip account for every line key if you
> >      >     want - even including the expansion modules.
> >      >
> >      >     However, on the Yealink phones (got looking at them cause of
> the
> >      >     T46G I won at ClueCon) we found the number of accounts
> >      >     very limited.
> >      >
> >      >     It turns out that with the latest firmware (73.x) on the
> Yealink
> >      >     units the count is increased on a number of the models
> >      >     (to 16 on the T46 for example). The problem is that with the
> middle
> >      >     tier ones that you'd add an expansion module to - it
> >      >     doesn't really get you anything. If your base phone is
> limited to 6
> >      >     accounts, adding the expansion module ONLY gets you
> >      >     busy-lamp or speed dials.
> >      >
> >      >     We're working on getting the users "converted" to not using
> full
> >      >     lines wherever possible, but still want options open.
> >      >
> >      >     -- Nathan
> >      >
> >      >
> >      > Thanks for the explanation. I share your feelings about the T46.
> I love
> >      > that phone but hate the fact that you only get 6 SIP accounts.
> (Glad to
> >      > hear that they added more in a recent firmware - I'll test that
> out at
> >      > some point...)
> >      >
> >      > If you find a solution other than the Cisco one I would be
> interested in
> >      > hearing about it.
> >      >
> >      > Thanks,
> >      > Michael
> >      >
> >      >
> >      >
> _________________________________________________________________________
> >      > 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://confluence.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://confluence.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
> >
> >
> >
> >
> > --
> >
> > */Brian West/*
> > brian at freeswitch.org <mailto:brian at freeswitch.org>
> >
> >
> > */Twitter: @FreeSWITCH , @briankwest/*
> > http://www.freeswitchbook.com
> > http://www.freeswitchcookbook.com
> >
> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/5ecdce64/attachment.html 

From brian at freeswitch.org  Tue Mar 10 04:37:40 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 9 Mar 2015 20:37:40 -0500
Subject: [Freeswitch-users] issue terminating outbound calls in
	freeswitch
In-Reply-To: <CAN3R+ShXKxOdj6WBD11foTxRFJ==Jv0ePp++bLT6S=7R5VkB3Q@mail.gmail.com>
References: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>
	<B31DB44E-8A50-4F9D-A4B8-508B1FCDB170@jerris.com>
	<bf6fe30133e2aced7071b7608d0e7d45@spingine.com>
	<CAN3R+ShXKxOdj6WBD11foTxRFJ==Jv0ePp++bLT6S=7R5VkB3Q@mail.gmail.com>
Message-ID: <CAEJMVkAwE22H5zWHeQVWTyZtjKBWjwYcyTPfsouUoS4LTbNCjA@mail.gmail.com>

Can you get a sip trace?

On Mon, Mar 9, 2015 at 8:26 PM, David Montecillo <
montecillodavid.spingine at gmail.com> wrote:

> It's a hardware where you can insert multiple sims but Im just using the
> single version(http://www.dbltek.com/products/goip-1.html). We use it as
> a gsm gateway for outbound calls and freeswitch setup instructions are
> available in the freeswitch pagehttps://freeswitch.org/
> confluence/display/FREESWITCH/Goip+HowTo
>
> Do you have an idea whats causing the termination problem?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150309/7aa175da/attachment-0001.html 

From montecillodavid.spingine at gmail.com  Tue Mar 10 05:04:36 2015
From: montecillodavid.spingine at gmail.com (David Montecillo)
Date: Tue, 10 Mar 2015 10:04:36 +0800
Subject: [Freeswitch-users] issue terminating outbound calls in
	freeswitch
In-Reply-To: <CAEJMVkAwE22H5zWHeQVWTyZtjKBWjwYcyTPfsouUoS4LTbNCjA@mail.gmail.com>
References: <CAN3R+SirvqRH_cRzE7=GTg7YTSvAYm1ymmO2sTcAz9sDRo4Z6g@mail.gmail.com>
	<B31DB44E-8A50-4F9D-A4B8-508B1FCDB170@jerris.com>
	<bf6fe30133e2aced7071b7608d0e7d45@spingine.com>
	<CAN3R+ShXKxOdj6WBD11foTxRFJ==Jv0ePp++bLT6S=7R5VkB3Q@mail.gmail.com>
	<CAEJMVkAwE22H5zWHeQVWTyZtjKBWjwYcyTPfsouUoS4LTbNCjA@mail.gmail.com>
Message-ID: <CAN3R+Si_RkkDsqeQkeYizpHmJ4Yx1ZDHbZyDuxxz5xyU9eecCw@mail.gmail.com>

freeswitch ip is 55.255.43.35
goip is 122.107.515.356


call-log.txt is attached.

thanks,
Dave

On Tue, Mar 10, 2015 at 9:37 AM, Brian West <brian at freeswitch.org> wrote:

> Can you get a sip trace?
>
> On Mon, Mar 9, 2015 at 8:26 PM, David Montecillo <
> montecillodavid.spingine at gmail.com> wrote:
>
>> It's a hardware where you can insert multiple sims but Im just using the
>> single version(http://www.dbltek.com/products/goip-1.html). We use it as
>> a gsm gateway for outbound calls and freeswitch setup instructions are
>> available in the freeswitch pagehttps://freeswitch.org/
>> confluence/display/FREESWITCH/Goip+HowTo
>>
>> Do you have an idea whats causing the termination problem?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/83e00967/attachment-0001.html 
-------------- next part --------------

.=======================================================.
|            _____ ____     ____ _     ___              |
|           |  ___/ ___|   / ___| |   |_ _|             |
|           | |_  \___ \  | |   | |    | |              |
|           |  _|  ___) | | |___| |___ | |              |
|           |_|   |____/   \____|_____|___|             |
|                                                       |
.=======================================================.
| Anthony Minessale II, Ken Rice,                       |
| Michael Jerris, Travis Cross                          |
| FreeSWITCH (http://www.freeswitch.org)                |
| Paypal Donations Appreciated: paypal at freeswitch.org   |
| Brought to you by ClueCon http://www.cluecon.com/     |
.=======================================================.

.===============================================================.
|       _                                                       |
|   ___| |_   _  ___  ___ ___  _ __         ___ ___  _ __ ___   |
|  / __| | | | |/ _ \/ __/ _ \| '_ \       / __/ _ \| '_ ` _ \  |
| | (__| | |_| |  __/ (_| (_) | | | |  _  | (_| (_) | | | | | | |
|  \___|_|\__,_|\___|\___\___/|_| |_| (_)  \___\___/|_| |_| |_| |
|                                                               |
.===============================================================.

Type /help <enter> to see a list of commands



   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK306957307
   From: "1000" <sip:1000 at 55.255.43.35>;tag=29480845
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 40 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="6f1204c8-                                                                                c6c8-11e4-8f41-79798a807fc4", uri="sip:55.255.43.35", response="7d1e824bdb87b1e5                                                                                69e023e3ced7d667", algorithm=MD5, cnonce="54fe4ee8", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 649 bytes to udp/[122.107.515.356]:5060 at 09:55:18.975642:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK306957307;received=112.207.1                                                                                55.145;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=29480845
   To: "1000" <sip:1000 at 55.255.43.35>;tag=tZ715ZaFjXDDc
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 40 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE                                                                                FER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   WWW-Authenticate: Digest realm="55.255.43.35", nonce="811b16b4-c6c8-11e4-8f42                                                                                -79798a807fc4", stale=true, algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 588 bytes from udp/[122.107.515.356]:5060 at 09:55:19.098722:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK945832044
   From: "1000" <sip:1000 at 55.255.43.35>;tag=29480845
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 41 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="811b16b4-                                                                                c6c8-11e4-8f42-79798a807fc4", uri="sip:55.255.43.35", response="7e9353575e330165                                                                                cb9f4f8384afc611", algorithm=MD5, cnonce="54fe4f06", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 624 bytes to udp/[122.107.515.356]:5060 at 09:55:19.108458:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK945832044;received=112.207.1                                                                                55.145;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=29480845
   To: "1000" <sip:1000 at 55.255.43.35>;tag=U80t7tUjF63ZQ
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 41 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060;received=122.107.515.356:5060>;expires=6                                                                                0
   Date: Tue, 10 Mar 2015 01:55:18 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE                                                                                FER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 794 bytes from udp/[122.107.515.356]:15647 at 09:55:21.564252:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-b8661640cefa8b3b-1-                                                                                --d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:302 at 192.168.1.46:50204;rinstance=2535928de6f322f2>
   To: "agentYellow"<sip:302 at 55.255.43.35>
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=1cff7d45
   Call-ID: ODE5YTgxNTQzOTM2Nzk1MjI1ODQ0YWY2NmE1NzBkMjU
   CSeq: 3 REGISTER
   Expires: 180
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,                                                                                 INFO
   User-Agent: X-Lite 4.7.1 74247-b4cb457e-W6.1
   Authorization: Digest username="302",realm="55.255.43.35",nonce="21dd69d6-c6c                                                                                8-11e4-8f20-79798a807fc4",uri="sip:55.255.43.35",response="b1cbff3f102a1da53a2a1                                                                                dce2040cf23",cnonce="3fb9b38a414ca26558e5b51d9648443b",nc=00000002,qop=auth,algo                                                                                rithm=MD5
   Content-Length: 0

   ------------------------------------------------------------------------
send 712 bytes to udp/[122.107.515.356]:15647 at 09:55:21.575521:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-b8661640cefa8b3b-1-                                                                                --d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=1cff7d45
   To: "agentYellow" <sip:302 at 55.255.43.35>;tag=vHtK9NcpcFtjK
   Call-ID: ODE5YTgxNTQzOTM2Nzk1MjI1ODQ0YWY2NmE1NzBkMjU
   CSeq: 3 REGISTER
   Contact: <sip:302 at 192.168.1.46:50204;rinstance=2535928de6f322f2;received=112.                                                                                207.155.145:15647>;expires=180
   Date: Tue, 10 Mar 2015 01:55:20 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE                                                                                FER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 793 bytes from udp/[121.96.255.69]:10326 at 09:55:26.411124:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.146:10326;branch=z9hG4bK-d8754z-6b594b6ee050ce5b-1                                                                                ---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:702 at 192.168.1.146:10326;rinstance=b5851044abd68743>
   To: "agentBlue"<sip:702 at 55.255.43.35>
   From: "agentBlue"<sip:702 at 55.255.43.35>;tag=231ac24f
   Call-ID: N2U0YmM4MmNhNzM0MmEyM2Y2OWUwYjEwZjMxZTMwY2M
   CSeq: 18 REGISTER
   Expires: 180
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,                                                                                 INFO
   User-Agent: X-Lite 4.7.0 73589-708b141d-W6.1
   Authorization: Digest username="702",realm="55.255.43.35",nonce="7af4e644-c6c                                                                                2-11e4-8f02-79798a807fc4",uri="sip:55.255.43.35",response="3923340ed9e75f33f85de                                                                                7ab79efa269",cnonce="160382e5efddc2fa3da5ae371f016ac9",nc=00000011,qop=auth,algo                                                                                rithm=MD5
   Content-Length: 0

   ------------------------------------------------------------------------
send 707 bytes to udp/[121.96.255.69]:10326 at 09:55:26.422433:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.146:10326;branch=z9hG4bK-d8754z-6b594b6ee050ce5b-1                                                                                ---d8754z-;rport=10326;received=121.96.255.69
   From: "agentBlue"<sip:702 at 55.255.43.35>;tag=231ac24f
   To: "agentBlue" <sip:702 at 55.255.43.35>;tag=XtKcBHXS9Qg5e
   Call-ID: N2U0YmM4MmNhNzM0MmEyM2Y2OWUwYjEwZjMxZTMwY2M
   CSeq: 18 REGISTER
   Contact: <sip:702 at 192.168.1.146:10326;rinstance=b5851044abd68743;received=121                                                                                .96.255.69:10326>;expires=180
   Date: Tue, 10 Mar 2015 01:55:25 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE                                                                                FER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 789 bytes from udp/[121.96.255.69]:10326 at 09:55:26.577667:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.146:10326;branch=z9hG4bK-d8754z-40817d3badf2374b-1                                                                                ---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:702 at 192.168.1.146:10326;rinstance=b5851044abd68743>;expires=0
   To: "agentBlue"<sip:702 at 55.255.43.35>
   From: "agentBlue"<sip:702 at 55.255.43.35>;tag=231ac24f
   Call-ID: N2U0YmM4MmNhNzM0MmEyM2Y2OWUwYjEwZjMxZTMwY2M
   CSeq: 19 REGISTER
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,                                                                                 INFO
   User-Agent: X-Lite 4.7.0 73589-708b141d-W6.1
   Authorization: Digest username="702",realm="55.255.43.35",nonce="7af4e644-c6c                                                                                2-11e4-8f02-79798a807fc4",uri="sip:55.255.43.35",response="353c24147b466f8793428                                                                                85f85143b9d",cnonce="83b06e565600c883484d6cab57f49e47",nc=00000012,qop=auth,algo                                                                                rithm=MD5
   Content-Length: 0

   ------------------------------------------------------------------------
send 599 bytes to udp/[121.96.255.69]:10326 at 09:55:26.587177:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.146:10326;branch=z9hG4bK-d8754z-40817d3badf2374b-1                                                                                ---d8754z-;rport=10326;received=121.96.255.69
   From: "agentBlue"<sip:702 at 55.255.43.35>;tag=231ac24f
   To: "agentBlue" <sip:702 at 55.255.43.35>;tag=y3c5cceX606Qa
   Call-ID: N2U0YmM4MmNhNzM0MmEyM2Y2OWUwYjEwZjMxZTMwY2M
   CSeq: 19 REGISTER
   Date: Tue, 10 Mar 2015 01:55:25 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, RE                                                                                FER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 832 bytes from udp/[122.107.515.356]:15647 at 09:55:43.010783:
   ------------------------------------------------------------------------
   INVITE sip:09204630267 at 55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-774e3a0199a1c601-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:302 at 192.168.1.46:50204>
   To: <sip:09204630267 at 55.255.43.35>
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   Supported: replaces
   User-Agent: X-Lite 4.7.1 74247-b4cb457e-W6.1
   Content-Length: 270

   v=0
   o=- 13070426140613802 1 IN IP4 192.168.1.46
   s=X-Lite release 4.7.1 stamp 74247
   c=IN IP4 192.168.1.46
   t=0 0
   m=audio 63604 RTP/AVP 100 0 97 9 8 101
   a=rtpmap:100 speex/16000
   a=rtpmap:97 speex/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv
   ------------------------------------------------------------------------
send 407 bytes to udp/[122.107.515.356]:15647 at 09:55:43.011085:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-774e3a0199a1c601-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:42.400917 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/302 at 55.255.43.35 [8f6eb162-c6c8-11e4-8f43-79798a807fc4]
2015-03-10 09:55:42.400917 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.400917 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.400917 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_NEW
2015-03-10 09:55:42.400917 [DEBUG] sofia.c:8834 sofia/internal/302 at 55.255.43.35 receiving invite from 122.107.515.356:15647 version: 1.4.14 git ca1d990 2014-11-19 22:11:13Z 64bit
2015-03-10 09:55:42.400917 [DEBUG] sofia.c:9001 IP 122.107.515.356 Rejected by acl "domains". Falling back to Digest auth.
send 903 bytes to udp/[122.107.515.356]:15647 at 09:55:43.012241:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-774e3a0199a1c601-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>;tag=Zc6Xe7y039vap
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="55.255.43.35", nonce="8f6ec544-c6c8-11e4-8f44-79798a807fc4", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:42.400917 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.400917 [DEBUG] sofia.c:2067 detaching session 8f6eb162-c6c8-11e4-8f43-79798a807fc4
2015-03-10 09:55:42.400917 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/302 at 55.255.43.35) State NEW
recv 352 bytes from udp/[122.107.515.356]:15647 at 09:55:43.124937:
   ------------------------------------------------------------------------
   ACK sip:09204630267 at 55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-774e3a0199a1c601-1---d8754z-;rport
   Max-Forwards: 70
   To: <sip:09204630267 at 55.255.43.35>;tag=Zc6Xe7y039vap
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 1 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1098 bytes from udp/[122.107.515.356]:15647 at 09:55:43.227090:
   ------------------------------------------------------------------------
   INVITE sip:09204630267 at 55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-f0d3a8756b141934-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:302 at 192.168.1.46:50204>
   To: <sip:09204630267 at 55.255.43.35>
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 2 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   Proxy-Authorization: Digest username="302",realm="55.255.43.35",nonce="8f6ec544-c6c8-11e4-8f44-79798a807fc4",uri="sip:09204630267 at 55.255.43.35",response="ee908e82e35c39ac6554cdb4f06ee09b",cnonce="714dd3aeb53e97cc846d9898cef22a1b",nc=00000001,qop=auth,algorithm=MD5
   Supported: replaces
   User-Agent: X-Lite 4.7.1 74247-b4cb457e-W6.1
   Content-Length: 270

   v=0
   o=- 13070426140613802 1 IN IP4 192.168.1.46
   s=X-Lite release 4.7.1 stamp 74247
   c=IN IP4 192.168.1.46
   t=0 0
   m=audio 63604 RTP/AVP 100 0 97 9 8 101
   a=rtpmap:100 speex/16000
   a=rtpmap:97 speex/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv
   ------------------------------------------------------------------------
send 407 bytes to udp/[122.107.515.356]:15647 at 09:55:43.227338:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-f0d3a8756b141934-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:2175 Re-attaching to session 8f6eb162-c6c8-11e4-8f43-79798a807fc4
2015-03-10 09:55:42.620941 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:8834 sofia/internal/302 at 55.255.43.35 receiving invite from 122.107.515.356:15647 version: 1.4.14 git ca1d990 2014-11-19 22:11:13Z 64bit
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:9001 IP 122.107.515.356 Rejected by acl "domains". Falling back to Digest auth.
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:10100 Setting NAT mode based on via received
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:6614 Channel sofia/internal/302 at 55.255.43.35 entering state [received][100]
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=- 13070426140613802 1 IN IP4 192.168.1.46
s=X-Lite release 4.7.1 stamp 74247
c=IN IP4 192.168.1.46
t=0 0
m=audio 63604 RTP/AVP 100 0 97 9 8 101
a=rtpmap:100 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:100:16000:20:0:1]/[G722:9:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:100:16000:20:0:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:97:8000:20:0:1]/[G722:9:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:97:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:97:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [speex:97:8000:20:0:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3531 Set telephone-event payload to 101
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/302 at 55.255.43.35 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2015-03-10 09:55:42.620941 [DEBUG] switch_core_codec.c:111 sofia/internal/302 at 55.255.43.35 Original read codec set to PCMU:0
2015-03-10 09:55:42.620941 [DEBUG] switch_core_media.c:3861 Set 2833 dtmf send/recv payload to 101
2015-03-10 09:55:42.620941 [DEBUG] sofia.c:6910 (sofia/internal/302 at 55.255.43.35) State Change CS_NEW -> CS_INIT
2015-03-10 09:55:42.620941 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_INIT
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/302 at 55.255.43.35) State INIT
2015-03-10 09:55:42.620941 [DEBUG] mod_sofia.c:87 sofia/internal/302 at 55.255.43.35 SOFIA INIT
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:40 sofia/internal/302 at 55.255.43.35 Standard INIT
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/302 at 55.255.43.35) State Change CS_INIT -> CS_ROUTING
2015-03-10 09:55:42.620941 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/302 at 55.255.43.35) State INIT going to sleep
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_ROUTING
2015-03-10 09:55:42.620941 [DEBUG] switch_channel.c:2184 (sofia/internal/302 at 55.255.43.35) Callstate Change DOWN -> RINGING
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/302 at 55.255.43.35) State ROUTING
2015-03-10 09:55:42.620941 [DEBUG] mod_sofia.c:123 sofia/internal/302 at 55.255.43.35 SOFIA ROUTING
2015-03-10 09:55:42.620941 [DEBUG] switch_core_state_machine.c:166 sofia/internal/302 at 55.255.43.35 Standard ROUTING
2015-03-10 09:55:42.620941 [INFO] mod_dialplan_xml.c:635 Processing agentYellow <302>->09204630267 in context default
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->user_exists] continue=true
Dialplan: sofia/internal/302 at 55.255.43.35 Absolute Condition [user_exists]
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(user_exists=${user_exists id ${destination_number} ${domain_name}}) INLINE
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x7fb4740e4f30 Connected.
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:377 DBH handle 0x7fb4740e4f30 released.
EXECUTE sofia/internal/302 at 55.255.43.35 set(user_exists=false)
2015-03-10 09:55:42.640973 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [user_exists]=[false]
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_exists] ${user_exists}(false) =~ /^true$/ break=on-false
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->call-direction] continue=true
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 ANTI-Action set(call_direction=local)
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [call-direction] ${user_exists}(false) =~ /^false$/ break=on-false
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [call-direction] destination_number(09204630267) =~ /^\d{7,20}$/ break=on-false
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(call_direction=outbound)
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->variables] continue=true
Dialplan: sofia/internal/302 at 55.255.43.35 Absolute Condition [variables]
Dialplan: sofia/internal/302 at 55.255.43.35 Action export(origination_callee_id_name=${destination_number})
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->user_record] continue=true
Dialplan: sofia/internal/302 at 55.255.43.35 Absolute Condition [user_record]
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(user_record=${user_data ${destination_number}@${domain_name} var user_record}) INLINE
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x7fb4740e4f30 Connected.
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:377 DBH handle 0x7fb4740e4f30 released.
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x7fb4740e4f30 Connected.
2015-03-10 09:55:42.640973 [DEBUG] freeswitch_lua.cpp:377 DBH handle 0x7fb4740e4f30 released.
EXECUTE sofia/internal/302 at 55.255.43.35 set(user_record=)
2015-03-10 09:55:42.640973 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [user_record]=[UNDEF]
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(from_user_exists=${user_exists id ${sip_from_user} ${sip_from_host}}) INLINE
EXECUTE sofia/internal/302 at 55.255.43.35 set(from_user_exists=true)
2015-03-10 09:55:42.640973 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [from_user_exists]=[true]
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_record}() =~ /^all$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^inbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_record}() =~ /^inbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^outbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_record}() =~ /^outbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^local$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${user_record}() =~ /^local$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [user_record] ${from_user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(from_user_record=${user_data ${sip_from_user}@${sip_from_host} var user_record}) INLINE
EXECUTE sofia/internal/302 at 55.255.43.35 set(from_user_record=)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [from_user_record]=[UNDEF]
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [user_record] ${from_user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${from_user_record}() =~ /^all$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [user_record] ${from_user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^inbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${from_user_record}() =~ /^inbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [user_record] ${from_user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^outbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${from_user_record}() =~ /^outbound$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [user_record] ${from_user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${call_direction}() =~ /^local$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${from_user_record}() =~ /^local$/ break=never
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [user_record] ${record_session}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->redial] continue=true
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (FAIL) [redial] destination_number(09204630267) =~ /^(redial|\*870)$/ break=on-true
Dialplan: sofia/internal/302 at 55.255.43.35 Absolute Condition [redial]
Dialplan: sofia/internal/302 at 55.255.43.35 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/302 at 55.255.43.35 parsing [default->GoIP.d11] continue=false
Dialplan: sofia/internal/302 at 55.255.43.35 Regex (PASS) [GoIP.d11] destination_number(09204630267) =~ /^(\d{11})$/ break=on-false
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(sip_h_X-accountcode=${accountcode})
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(sip_h_X-Tag=)
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(call_direction=outbound)
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(inherit_codec=true)
Dialplan: sofia/internal/302 at 55.255.43.35 Action set(continue_on_fail=true)
Dialplan: sofia/internal/302 at 55.255.43.35 Action bridge(sofia/gateway/8b468805-04d4-4de4-9fd1-ad1b9f01a37d/09204630267)
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/302 at 55.255.43.35) State Change CS_ROUTING -> CS_EXECUTE
2015-03-10 09:55:42.660924 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/302 at 55.255.43.35) State ROUTING going to sleep
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_EXECUTE
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/302 at 55.255.43.35) State EXECUTE
2015-03-10 09:55:42.660924 [DEBUG] mod_sofia.c:178 sofia/internal/302 at 55.255.43.35 SOFIA EXECUTE
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:258 sofia/internal/302 at 55.255.43.35 Standard EXECUTE
EXECUTE sofia/internal/302 at 55.255.43.35 set(call_direction=local)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [call_direction]=[local]
EXECUTE sofia/internal/302 at 55.255.43.35 set(call_direction=outbound)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [call_direction]=[outbound]
EXECUTE sofia/internal/302 at 55.255.43.35 export(origination_callee_id_name=09204630267)
2015-03-10 09:55:42.660924 [DEBUG] switch_channel.c:1247 EXPORT (export_vars) [origination_callee_id_name]=[09204630267]
EXECUTE sofia/internal/302 at 55.255.43.35 set(RFC2822_DATE=Tue, 10 Mar 2015 09:55:42 +0800)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [RFC2822_DATE]=[Tue, 10 Mar 2015 09:55:42 +0800]
EXECUTE sofia/internal/302 at 55.255.43.35 hash(insert/55.255.43.35-last_dial/302/09204630267)
EXECUTE sofia/internal/302 at 55.255.43.35 set(sip_h_X-accountcode=55.255.43.35)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [sip_h_X-accountcode]=[55.255.43.35]
EXECUTE sofia/internal/302 at 55.255.43.35 set(sip_h_X-Tag=)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [sip_h_X-Tag]=[UNDEF]
EXECUTE sofia/internal/302 at 55.255.43.35 set(call_direction=outbound)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [call_direction]=[outbound]
EXECUTE sofia/internal/302 at 55.255.43.35 set(hangup_after_bridge=true)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/302 at 55.255.43.35 set(effective_caller_id_name=)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [effective_caller_id_name]=[UNDEF]
EXECUTE sofia/internal/302 at 55.255.43.35 set(effective_caller_id_number=)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [effective_caller_id_number]=[UNDEF]
EXECUTE sofia/internal/302 at 55.255.43.35 set(inherit_codec=true)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [inherit_codec]=[true]
EXECUTE sofia/internal/302 at 55.255.43.35 set(continue_on_fail=true)
2015-03-10 09:55:42.660924 [DEBUG] mod_dptools.c:1435 sofia/internal/302 at 55.255.43.35 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/302 at 55.255.43.35 bridge(sofia/gateway/8b468805-04d4-4de4-9fd1-ad1b9f01a37d/09204630267)
2015-03-10 09:55:42.660924 [DEBUG] switch_channel.c:1201 sofia/internal/302 at 55.255.43.35 EXPORTING[export_vars] [domain_name]=[55.255.43.35] to event
2015-03-10 09:55:42.660924 [DEBUG] switch_channel.c:1201 sofia/internal/302 at 55.255.43.35 EXPORTING[export_vars] [origination_callee_id_name]=[09204630267] to event
2015-03-10 09:55:42.660924 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables
2015-03-10 09:55:42.660924 [NOTICE] switch_channel.c:1055 New Channel sofia/external/09204630267 [8f95d710-c6c8-11e4-8f5a-79798a807fc4]
2015-03-10 09:55:42.660924 [DEBUG] mod_sofia.c:4615 (sofia/external/09204630267) State Change CS_NEW -> CS_INIT
2015-03-10 09:55:42.660924 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_INIT
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:512 (sofia/external/09204630267) State INIT
2015-03-10 09:55:42.660924 [DEBUG] mod_sofia.c:87 sofia/external/09204630267 SOFIA INIT
2015-03-10 09:55:42.660924 [DEBUG] sofia_glue.c:1232 sofia/external/09204630267 sending invite version: 1.4.14 git ca1d990 2014-11-19 22:11:13Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1425922064 1425922065 IN IP4 55.255.43.35
s=FreeSWITCH
c=IN IP4 55.255.43.35
t=0 0
m=audio 30478 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

send 1205 bytes to udp/[122.107.515.356]:5060 at 09:55:43.269019:
   ------------------------------------------------------------------------
   INVITE sip:09204630267 at 122.107.515.356 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   Max-Forwards: 69
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 270
   X-accountcode: 55.255.43.35
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "agentYellow" <sip:302 at 122.107.515.356>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1425922064 1425922065 IN IP4 55.255.43.35
   s=FreeSWITCH
   c=IN IP4 55.255.43.35
   t=0 0
   m=audio 30478 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2015-03-10 09:55:42.660924 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:40 sofia/external/09204630267 Standard INIT
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:48 (sofia/external/09204630267) State Change CS_INIT -> CS_ROUTING
2015-03-10 09:55:42.660924 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:512 (sofia/external/09204630267) State INIT going to sleep
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_ROUTING
2015-03-10 09:55:42.660924 [DEBUG] sofia.c:6614 Channel sofia/external/09204630267 entering state [calling][0]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:528 (sofia/external/09204630267) State ROUTING
2015-03-10 09:55:42.660924 [DEBUG] mod_sofia.c:123 sofia/external/09204630267 SOFIA ROUTING
2015-03-10 09:55:42.660924 [DEBUG] switch_ivr_originate.c:67 (sofia/external/09204630267) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-03-10 09:55:42.660924 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:528 (sofia/external/09204630267) State ROUTING going to sleep
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_CONSUME_MEDIA
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:547 (sofia/external/09204630267) State CONSUME_MEDIA
2015-03-10 09:55:42.660924 [DEBUG] switch_core_state_machine.c:547 (sofia/external/09204630267) State CONSUME_MEDIA going to sleep
recv 358 bytes from udp/[122.107.515.356]:5060 at 09:55:43.539658:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Length: 0

   ------------------------------------------------------------------------
recv 359 bytes from udp/[122.107.515.356]:5060 at 09:55:43.544714:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:42.941015 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.941015 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:42.941015 [DEBUG] sofia.c:6614 Channel sofia/external/09204630267 entering state [proceeding][180]
2015-03-10 09:55:42.941015 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/09204630267!
2015-03-10 09:55:42.941015 [DEBUG] switch_channel.c:3277 (sofia/external/09204630267) Callstate Change DOWN -> RINGING
2015-03-10 09:55:42.941015 [INFO] switch_ivr_originate.c:1192 Sending early media
2015-03-10 09:55:42.941015 [DEBUG] switch_core_media.c:5111 AUDIO RTP [sofia/internal/302 at 55.255.43.35] 10.142.74.23 port 23340 -> 192.168.1.46 port 63604 codec: 0 ms: 20
2015-03-10 09:55:42.941015 [DEBUG] switch_rtp.c:3521 Starting timer [soft] 160 bytes per 20ms
2015-03-10 09:55:42.941015 [DEBUG] switch_core_media.c:5409 Set 2833 dtmf send payload to 101
2015-03-10 09:55:42.941015 [DEBUG] switch_core_media.c:5415 Set 2833 dtmf receive payload to 101
2015-03-10 09:55:42.941015 [DEBUG] mod_sofia.c:2247 Ring SDP:
v=0
o=FreeSWITCH 1425929202 1425929203 IN IP4 55.255.43.35
s=FreeSWITCH
c=IN IP4 55.255.43.35
t=0 0
m=audio 23340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-03-10 09:55:42.941015 [NOTICE] mod_sofia.c:2250 Pre-Answer sofia/internal/302 at 55.255.43.35!
2015-03-10 09:55:42.941015 [DEBUG] switch_channel.c:3399 (sofia/internal/302 at 55.255.43.35) Callstate Change RINGING -> EARLY
send 1210 bytes to udp/[122.107.515.356]:15647 at 09:55:43.553391:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-f0d3a8756b141934-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>;tag=0NZpg2F40jKXH
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 2 INVITE
   Contact: <sip:09204630267 at 55.255.43.35:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   Remote-Party-ID: "09204630267" <sip:09204630267 at 55.255.43.35>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1425929202 1425929203 IN IP4 55.255.43.35
   s=FreeSWITCH
   c=IN IP4 55.255.43.35
   t=0 0
   m=audio 23340 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2015-03-10 09:55:42.941015 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.941015 [DEBUG] sofia.c:6614 Channel sofia/internal/302 at 55.255.43.35 entering state [early][183]
2015-03-10 09:55:42.941015 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:42.941015 [DEBUG] switch_ivr_originate.c:1249 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2015-03-10 09:55:42.941015 [DEBUG] switch_core_codec.c:221 sofia/internal/302 at 55.255.43.35 Push codec L16:70
2015-03-10 09:55:42.941015 [DEBUG] switch_ivr_originate.c:1317 Play Ringback Tone [%(2000, 4000, 440.0, 480.0)]
2015-03-10 09:55:43.340919 [INFO] switch_rtp.c:5799 Auto Changing port from 192.168.1.46:63604 to 122.107.515.356:16122
recv 590 bytes from udp/[122.107.515.356]:5060 at 09:55:49.234815:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK396683003
   From: "1000" <sip:1000 at 55.255.43.35>;tag=1913811103
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 42 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="811b16b4-c6c8-11e4-8f42-79798a807fc4", uri="sip:55.255.43.35", response="7e9353575e330165cb9f4f8384afc611", algorithm=MD5, cnonce="54fe4f06", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 651 bytes to udp/[122.107.515.356]:5060 at 09:55:49.236225:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK396683003;received=122.107.515.356;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=1913811103
   To: "1000" <sip:1000 at 55.255.43.35>;tag=1yrFjX07XU9FD
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 42 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   WWW-Authenticate: Digest realm="55.255.43.35", nonce="93247daa-c6c8-11e4-8f5e-79798a807fc4", stale=true, algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 591 bytes from udp/[122.107.515.356]:5060 at 09:55:49.364584:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1402019793
   From: "1000" <sip:1000 at 55.255.43.35>;tag=1913811103
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 43 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="93247daa-c6c8-11e4-8f5e-79798a807fc4", uri="sip:55.255.43.35", response="4890ba8ee8cf09cf63667c2b58dceda3", algorithm=MD5, cnonce="54fe4f25", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 627 bytes to udp/[122.107.515.356]:5060 at 09:55:49.375527:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1402019793;received=122.107.515.356;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=1913811103
   To: "1000" <sip:1000 at 55.255.43.35>;tag=27H8KrHBU4Z2r
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 43 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060;received=122.107.515.356:5060>;expires=60
   Date: Tue, 10 Mar 2015 01:55:48 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 617 bytes from udp/[122.107.515.356]:5060 at 09:55:49.528472:
   ------------------------------------------------------------------------
   SIP/2.0 183 Ringing
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
2015-03-10 09:55:48.920917 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:48.920917 [DEBUG] sofia.c:6614 Channel sofia/external/09204630267 entering state [proceeding][183]
2015-03-10 09:55:48.920917 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=dble 1425952543 1425952543 IN IP4 127.0.0.1
s=dble
c=IN IP4 127.0.0.1
t=0 0
m=audio 64 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=ptime:20

2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3531 Set telephone-event payload to 101
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:2473 Set Codec sofia/external/09204630267 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2015-03-10 09:55:48.920917 [DEBUG] switch_core_codec.c:111 sofia/external/09204630267 Original read codec set to PCMU:0
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:3852 Set 2833 dtmf send payload to 101
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:5111 AUDIO RTP [sofia/external/09204630267] 10.142.74.23 port 30478 -> 127.0.0.1 port 64 codec: 0 ms: 20
2015-03-10 09:55:48.920917 [DEBUG] switch_rtp.c:3521 Starting timer [soft] 160 bytes per 20ms
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:5409 Set 2833 dtmf send payload to 101
2015-03-10 09:55:48.920917 [DEBUG] switch_core_media.c:5415 Set 2833 dtmf receive payload to 101
2015-03-10 09:55:48.920917 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/09204630267!
2015-03-10 09:55:48.920917 [DEBUG] switch_channel.c:3395 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:48.920917 [DEBUG] switch_channel.c:3399 (sofia/external/09204630267) Callstate Change RINGING -> EARLY
2015-03-10 09:55:48.940916 [DEBUG] switch_core_codec.c:246 sofia/internal/302 at 55.255.43.35 Restore previous codec PCMU:0.
2015-03-10 09:55:48.940916 [DEBUG] switch_ivr_originate.c:3552 Originate Resulted in Success: [sofia/external/09204630267]
2015-03-10 09:55:48.940916 [DEBUG] switch_core_session.c:908 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:48.940916 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:48.940916 [DEBUG] switch_ivr_bridge.c:1465 (sofia/external/09204630267) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-03-10 09:55:48.940916 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:48.940916 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_EXCHANGE_MEDIA
2015-03-10 09:55:48.940916 [DEBUG] switch_core_state_machine.c:538 (sofia/external/09204630267) State EXCHANGE_MEDIA
2015-03-10 09:55:48.940916 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
2015-03-10 09:55:49.200912 [INFO] switch_rtp.c:5799 Auto Changing port from 127.0.0.1:64 to 122.107.515.356:16138
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:55.270439:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
2015-03-10 09:55:54.660918 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:54.660918 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:54.680926 [DEBUG] sofia.c:6614 Channel sofia/external/09204630267 entering state [completing][200]
2015-03-10 09:55:54.680926 [DEBUG] sofia.c:6621 Duplicate SDP
v=0
o=dble 1425952543 1425952543 IN IP4 127.0.0.1
s=dble
c=IN IP4 127.0.0.1
t=0 0
m=audio 64 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=ptime:20

send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:55.280611:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:54.680926 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_session.c:1053 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:54.680926 [DEBUG] sofia.c:6614 Channel sofia/external/09204630267 entering state [ready][200]
2015-03-10 09:55:54.680926 [DEBUG] sofia.c:6621 Duplicate SDP
v=0
o=dble 1425952543 1425952543 IN IP4 127.0.0.1
s=dble
c=IN IP4 127.0.0.1
t=0 0
m=audio 64 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=ptime:20

2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3670 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3615 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3531 Set telephone-event payload to 101
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:3852 Set 2833 dtmf send payload to 101
2015-03-10 09:55:54.680926 [DEBUG] sofia.c:7318 Processing updated SDP
2015-03-10 09:55:54.680926 [DEBUG] switch_core_media.c:5095 Audio params are unchanged for sofia/external/09204630267.
2015-03-10 09:55:54.680926 [DEBUG] switch_channel.c:3635 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.680926 [NOTICE] sofia.c:7416 Channel [sofia/external/09204630267] has been answered
2015-03-10 09:55:54.680926 [DEBUG] switch_channel.c:3689 (sofia/external/09204630267) Callstate Change EARLY -> ACTIVE
2015-03-10 09:55:54.700927 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/302 at 55.255.43.35:
v=0
o=FreeSWITCH 1425929202 1425929204 IN IP4 55.255.43.35
s=FreeSWITCH
c=IN IP4 55.255.43.35
t=0 0
m=audio 23340 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

send 1171 bytes to udp/[122.107.515.356]:15647 at 09:55:55.300459:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-f0d3a8756b141934-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>;tag=0NZpg2F40jKXH
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 2 INVITE
   Contact: <sip:09204630267 at 55.255.43.35:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   Remote-Party-ID: "09204630267" <sip:09204630267 at 55.255.43.35>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1425929202 1425929203 IN IP4 55.255.43.35
   s=FreeSWITCH
   c=IN IP4 55.255.43.35
   t=0 0
   m=audio 23340 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2015-03-10 09:55:54.700927 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.700927 [DEBUG] sofia.c:6614 Channel sofia/internal/302 at 55.255.43.35 entering state [completed][200]
2015-03-10 09:55:54.700927 [DEBUG] switch_core_session.c:908 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.700927 [NOTICE] switch_ivr_bridge.c:496 Channel [sofia/internal/302 at 55.255.43.35] has been answered
2015-03-10 09:55:54.700927 [DEBUG] switch_channel.c:3689 (sofia/internal/302 at 55.255.43.35) Callstate Change EARLY -> ACTIVE
recv 456 bytes from udp/[122.107.515.356]:15647 at 09:55:55.546122:
   ------------------------------------------------------------------------
   ACK sip:09204630267 at 55.255.43.35:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-056e9121881cc806-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:302 at 192.168.1.46:50204>
   To: <sip:09204630267 at 55.255.43.35>;tag=0NZpg2F40jKXH
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 2 ACK
   User-Agent: X-Lite 4.7.1 74247-b4cb457e-W6.1
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:55:54.940921 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.940921 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.940921 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:55:54.960927 [DEBUG] sofia.c:6614 Channel sofia/internal/302 at 55.255.43.35 entering state [ready][200]
2015-03-10 09:55:54.960927 [DEBUG] switch_core_session.c:970 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:55:54.960927 [DEBUG] switch_core_session.c:970 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:55.771022:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:55.771137:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:56.271378:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:56.271466:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:56.770084:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:56.770222:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:57.270390:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:57.270507:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:57.771018:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:57.771162:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:58.270213:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:58.270344:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:58.770275:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:58.770376:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:59.271140:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:59.271267:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:55:59.771103:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:55:59.771250:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:00.269915:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:00.270038:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:00.771198:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:00.771339:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:01.270185:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:01.270319:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 741 bytes from udp/[122.107.515.356]:15647 at 09:56:01.488948:
   ------------------------------------------------------------------------
   BYE sip:09204630267 at 55.255.43.35:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-b006662ad8579e0a-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:302 at 192.168.1.46:50204>
   To: <sip:09204630267 at 55.255.43.35>;tag=0NZpg2F40jKXH
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 3 BYE
   Proxy-Authorization: Digest username="302",realm="55.255.43.35",nonce="8f6ec544-c6c8-11e4-8f44-79798a807fc4",uri="sip:09204630267 at 55.255.43.35:5060;transport=udp",response="0a425931822241bac6801c7912981cb6",cnonce="85624e6a7988424bb3aee3fd485d2bef",nc=00000002,qop=auth,algorithm=MD5
   User-Agent: X-Lite 4.7.1 74247-b4cb457e-W6.1
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:56:00.880918 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:56:00.900932 [NOTICE] sofia.c:952 Hangup sofia/internal/302 at 55.255.43.35 [CS_EXECUTE] [NORMAL_CLEARING]
2015-03-10 09:56:00.900932 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/302 at 55.255.43.35 [KILL]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
send 555 bytes to udp/[122.107.515.356]:15647 at 09:56:01.501086:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.46:50204;branch=z9hG4bK-d8754z-b006662ad8579e0a-1---d8754z-;rport=15647;received=122.107.515.356
   From: "agentYellow"<sip:302 at 55.255.43.35>;tag=292c3a30
   To: <sip:09204630267 at 55.255.43.35>;tag=0NZpg2F40jKXH
   Call-ID: OWEzZmJiYWExOGRiY2E0MWY0YzhlMWUxMTRmMTdkNTU
   CSeq: 3 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/302 at 55.255.43.35]
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:579 sofia/internal/302 at 55.255.43.35 ending bridge by request from write function
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/external/09204630267]
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:56:00.900932 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/09204630267 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-03-10 09:56:00.900932 [DEBUG] switch_channel.c:3222 Send signal sofia/external/09204630267 [KILL]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:538 (sofia/external/09204630267) State EXCHANGE_MEDIA going to sleep
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_HANGUP
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:735 (sofia/external/09204630267) Callstate Change ACTIVE -> HANGUP
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:737 (sofia/external/09204630267) State HANGUP
2015-03-10 09:56:00.900932 [DEBUG] mod_sofia.c:407 sofia/external/09204630267 Overriding SIP cause 480 with 200 from the other leg
2015-03-10 09:56:00.900932 [DEBUG] mod_sofia.c:413 Channel sofia/external/09204630267 hanging up, cause: NORMAL_CLEARING
2015-03-10 09:56:00.900932 [DEBUG] mod_sofia.c:465 Sending BYE to sofia/external/09204630267
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:01.502694:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:60 sofia/external/09204630267 Standard HANGUP, cause: NORMAL_CLEARING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:737 (sofia/external/09204630267) State HANGUP going to sleep
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:504 (sofia/external/09204630267) State Change CS_HANGUP -> CS_REPORTING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:472 (sofia/external/09204630267) Running State Change CS_REPORTING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:823 (sofia/external/09204630267) State REPORTING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:104 sofia/external/09204630267 Standard REPORTING, cause: NORMAL_CLEARING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:823 (sofia/external/09204630267) State REPORTING going to sleep
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:498 (sofia/external/09204630267) State Change CS_REPORTING -> CS_DESTROY
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/09204630267 [BREAK]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1615 Session 4 (sofia/external/09204630267) Locked, Waiting on external entities
2015-03-10 09:56:00.900932 [DEBUG] switch_ivr_bridge.c:1566 sofia/internal/302 at 55.255.43.35 skip receive message [UNBRIDGE] (channel is hungup already)
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:2893 sofia/internal/302 at 55.255.43.35 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/302 at 55.255.43.35) State EXECUTE going to sleep
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_HANGUP
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/302 at 55.255.43.35) Callstate Change ACTIVE -> HANGUP
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/302 at 55.255.43.35) State HANGUP
2015-03-10 09:56:00.900932 [DEBUG] mod_sofia.c:413 Channel sofia/internal/302 at 55.255.43.35 hanging up, cause: NORMAL_CLEARING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:60 sofia/internal/302 at 55.255.43.35 Standard HANGUP, cause: NORMAL_CLEARING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/302 at 55.255.43.35) State HANGUP going to sleep
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/302 at 55.255.43.35) State Change CS_HANGUP -> CS_REPORTING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/302 at 55.255.43.35) Running State Change CS_REPORTING
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/302 at 55.255.43.35) State REPORTING
2015-03-10 09:56:00.900932 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: INSERT INTO cdr VALUES ("302","302","09204630267","default","2015-03-10 09:55:42","2015-03-10 09:55:54","2015-03-10 09:56:00",18,6,"NORMAL_CLEARING","8f6eb162-c6c8-11e4-8f43-79798a807fc4","8f95d710-c6c8-11e4-8f5a-79798a807fc4","55.255.43.35")
2015-03-10 09:56:00.900932 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/external/09204630267) Ended
2015-03-10 09:56:00.900932 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/09204630267 [CS_DESTROY]
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:626 (sofia/external/09204630267) Running State Change CS_DESTROY
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:636 (sofia/external/09204630267) State DESTROY
2015-03-10 09:56:00.900932 [DEBUG] mod_sofia.c:323 sofia/external/09204630267 SOFIA DESTROY
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:111 sofia/external/09204630267 Standard DESTROY
2015-03-10 09:56:00.900932 [DEBUG] switch_core_state_machine.c:636 (sofia/external/09204630267) State DESTROY going to sleep
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:104 sofia/internal/302 at 55.255.43.35 Standard REPORTING, cause: NORMAL_CLEARING
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/302 at 55.255.43.35) State REPORTING going to sleep
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/302 at 55.255.43.35) State Change CS_REPORTING -> CS_DESTROY
2015-03-10 09:56:00.940933 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/302 at 55.255.43.35 [BREAK]
2015-03-10 09:56:00.940933 [DEBUG] switch_core_session.c:1615 Session 3 (sofia/internal/302 at 55.255.43.35) Locked, Waiting on external entities
2015-03-10 09:56:00.940933 [NOTICE] switch_core_session.c:1633 Session 3 (sofia/internal/302 at 55.255.43.35) Ended
2015-03-10 09:56:00.940933 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/302 at 55.255.43.35 [CS_DESTROY]
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/302 at 55.255.43.35) Running State Change CS_DESTROY
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/302 at 55.255.43.35) State DESTROY
2015-03-10 09:56:00.940933 [DEBUG] mod_sofia.c:323 sofia/internal/302 at 55.255.43.35 SOFIA DESTROY
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:111 sofia/internal/302 at 55.255.43.35 Standard DESTROY
2015-03-10 09:56:00.940933 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/302 at 55.255.43.35) State DESTROY going to sleep
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:01.769934:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:01.770096:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:02.270931:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:02.271105:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:02.502809:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:02.769996:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:02.770114:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:03.270015:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:03.270153:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:03.770813:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:03.770925:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:04.269810:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:04.269969:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:04.502765:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 612 bytes from udp/[122.107.515.356]:5060 at 09:56:04.770001:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKK1yj09K1pe9Bc
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 INVITE
   Contact: <sip:1000 at 192.168.1.39:5060>
   User-Agent: dble
   Content-Type: application/sdp
   Content-Length:   223

   v=0
   o=dble 1425952543 1425952543 IN IP4 127.0.0.1
   s=dble
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 64 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=ptime:20
   ------------------------------------------------------------------------
send 464 bytes to udp/[192.168.1.39]:5060 at 09:56:04.770104:
   ------------------------------------------------------------------------
   ACK sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKmarB2444KQZyQ
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640207 ACK
   Contact: <sip:gw+8b468805-04d4-4de4-9fd1-ad1b9f01a37d at 55.255.43.35:5080;transport=udp;gw=8b468805-04d4-4de4-9fd1-ad1b9f01a37d>
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:08.502835:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:12.502937:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:16.503035:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 590 bytes from udp/[122.107.515.356]:5060 at 09:56:19.506942:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1628750260
   From: "1000" <sip:1000 at 55.255.43.35>;tag=786276451
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 44 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="93247daa-c6c8-11e4-8f5e-79798a807fc4", uri="sip:55.255.43.35", response="4890ba8ee8cf09cf63667c2b58dceda3", algorithm=MD5, cnonce="54fe4f25", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 651 bytes to udp/[122.107.515.356]:5060 at 09:56:19.508389:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1628750260;received=122.107.515.356;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=786276451
   To: "1000" <sip:1000 at 55.255.43.35>;tag=3gB1NK2erDpNm
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 44 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   WWW-Authenticate: Digest realm="55.255.43.35", nonce="a52fa79a-c6c8-11e4-8f60-79798a807fc4", stale=true, algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 590 bytes from udp/[122.107.515.356]:5060 at 09:56:19.636182:
   ------------------------------------------------------------------------
   REGISTER sip:55.255.43.35 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1830034809
   From: "1000" <sip:1000 at 55.255.43.35>;tag=786276451
   To: "1000" <sip:1000 at 55.255.43.35>
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 45 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060>;expires=60
   Authorization: Digest username="1000", realm="55.255.43.35", nonce="a52fa79a-c6c8-11e4-8f60-79798a807fc4", uri="sip:55.255.43.35", response="c50871e8fcf1291ce0a3a89d97d9d2b3", algorithm=MD5, cnonce="54fe4f43", qop=auth, nc=00000001
   Max-Forwards: 30
   User-Agent: dble
   Expires: 60
   Content-Length: 0

   ------------------------------------------------------------------------
send 626 bytes to udp/[122.107.515.356]:5060 at 09:56:19.647273:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.39:5060;branch=z9hG4bK1830034809;received=122.107.515.356;rport=5060
   From: "1000" <sip:1000 at 55.255.43.35>;tag=786276451
   To: "1000" <sip:1000 at 55.255.43.35>;tag=4S4SQeKjNpc8F
   Call-ID: 1308467922 at 192.168.1.39
   CSeq: 45 REGISTER
   Contact: <sip:1000 at 192.168.1.39:5060;received=122.107.515.356:5060>;expires=60
   Date: Tue, 10 Mar 2015 01:56:19 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
send 586 bytes to udp/[192.168.1.39]:5060 at 09:56:20.503136:
   ------------------------------------------------------------------------
   BYE sip:1000 at 192.168.1.39:5060 SIP/2.0
   Via: SIP/2.0/UDP 55.255.43.35:5080;rport;branch=z9hG4bKNKH43ZN8g0NHK
   Max-Forwards: 70
   From: "agentYellow" <sip:1000 at 122.107.515.356>;tag=yeD415gycDN6a
   To: <sip:09204630267 at 122.107.515.356>;tag=1299381679
   Call-ID: 66f44245-416b-1233-edbf-1231410449e9
   CSeq: 72640208 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.14+git~20141119T221113Z~ca1d990cfc~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0

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

From ssinyagin at gmail.com  Tue Mar 10 05:42:40 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 03:42:40 +0100
Subject: [Freeswitch-users] any handset recommendations that operate
	like polycom+cisco?
In-Reply-To: <CAEJMVkDUw2vXRZG0UeurNiB1MTk7V894mJvJUegDTymRjx1Obg@mail.gmail.com>
References: <54B82BA0.20201@mst.edu>
	<CAKzWOxUzsAk=EVDRw5+5Te4YQ3SwrQydGED4XiL3Y_VYbaPX0A@mail.gmail.com>
	<54C40829.9060501@mst.edu>
	<CAKzWOxW0miKC9CMfQ84CnY2tk1CooJucqnH6Tj8zB8J7uwWKow@mail.gmail.com>
	<54FDCE74.60407@communicatefreely.net>
	<CAEJMVkCPY1i2k-yyri4MaziN_d=MrHW+HMrxqpVSAJ=HcwHh2A@mail.gmail.com>
	<54FE1109.2060405@mst.edu>
	<CAEJMVkDUw2vXRZG0UeurNiB1MTk7V894mJvJUegDTymRjx1Obg@mail.gmail.com>
Message-ID: <CACOYK=gGXstUqNiqM2KUMFJt88RJ_72TrNxCJJCM_pndg-BtXA@mail.gmail.com>

I guess for the use case that Nathan has described, a software-based
switchboard would be sufficient. The receptionist needs to know the line
for which the inbound call is ringing, and busy/available status of other
lines. So, a tablet with big enough touch screen would do the job, and the
SIP handset would be just any standard phone.

On Tue, Mar 10, 2015 at 2:28 AM, Brian West <brian at freeswitch.org> wrote:

> Buttons, WHO needs buttons, you have two ears and one mouth, till that
> condition changes and people start sprouting more mouths and ears I suspect
> a low button count is sufficient.
>
> On Mon, Mar 9, 2015 at 4:30 PM, Nathan Neulinger <nneul at mst.edu> wrote:
>
>> Looks like the Fanvil phones have the same underlying issue as the
>> Yealinks unfortunately - # of sip accounts limited
>> significantly below # of buttons.
>>
>> -- Nathan
>>
>> On 03/09/2015 02:43 PM, Brian West wrote:
>> > The Fanvil phones are cheap and usable.
>> >
>> > On Mon, Mar 9, 2015 at 11:46 AM, Tim St. Pierre <
>> fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
>> > wrote:
>> >
>> >     Hello,
>> >
>> >     Just saw this.
>> >
>> >     If you are still looking, the Aastra / Mitel 6800 series can do
>> >     something like 20 SIP accounts, and you can put line keys on the
>> >     expansion modules.  I'm not sure if that is enough, but if it can
>> cover
>> >     you for the transition, that might get you by.
>> >
>> >     -Tim
>> >
>> >
>> >
>> >     On 2015-01-25 05:13 PM, Michael Collins wrote:
>> >     >
>> >     >
>> >     > On Sat, Jan 24, 2015 at 1:01 PM, Nathan Neulinger <nneul at mst.edu
>> <mailto:nneul at mst.edu>
>> >      > <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>> wrote:
>> >      >
>> >      >     Personally I think it's nuts... but we have a number of
>> >      >     secretary/admin/receptionist users with Cisco expansion
>> modules
>> >      >     that have the shared lines of all of the people in their
>> department
>> >      >     on them. (i.e. a 7940/7960 plus the module). Usually
>> >      >     with some portion of the other lines set to just flash and
>> not
>> >      >     audibly ring. While I'd expect that in most cases they
>> >      >     really would be sufficient with busy lamp, sometimes they do
>> use it
>> >      >     to answer arbitrary calls for faculty that are out
>> >      >     of the office/etc.
>> >      >
>> >      >     With the transitioned cisco phones on FS/mod_skinny - it
>> works the
>> >      >     same way, however we're wanting to position ourselves
>> >      >     with suitable replacements, particularly for any departments
>> that
>> >      >     want more than bare bones functionality.
>> >      >
>> >      >     With the polycom phones, it appears to also work that way
>> where you
>> >      >     can have a sip account for every line key if you
>> >      >     want - even including the expansion modules.
>> >      >
>> >      >     However, on the Yealink phones (got looking at them cause of
>> the
>> >      >     T46G I won at ClueCon) we found the number of accounts
>> >      >     very limited.
>> >      >
>> >      >     It turns out that with the latest firmware (73.x) on the
>> Yealink
>> >      >     units the count is increased on a number of the models
>> >      >     (to 16 on the T46 for example). The problem is that with the
>> middle
>> >      >     tier ones that you'd add an expansion module to - it
>> >      >     doesn't really get you anything. If your base phone is
>> limited to 6
>> >      >     accounts, adding the expansion module ONLY gets you
>> >      >     busy-lamp or speed dials.
>> >      >
>> >      >     We're working on getting the users "converted" to not using
>> full
>> >      >     lines wherever possible, but still want options open.
>> >      >
>> >      >     -- Nathan
>> >      >
>> >      >
>> >      > Thanks for the explanation. I share your feelings about the T46.
>> I love
>> >      > that phone but hate the fact that you only get 6 SIP accounts.
>> (Glad to
>> >      > hear that they added more in a recent firmware - I'll test that
>> out at
>> >      > some point...)
>> >      >
>> >      > If you find a solution other than the Cisco one I would be
>> interested in
>> >      > hearing about it.
>> >      >
>> >      > Thanks,
>> >      > Michael
>> >      >
>> >      >
>> >      >
>> _________________________________________________________________________
>> >      > 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://confluence.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://confluence.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
>> >
>> >
>> >
>> >
>> > --
>> >
>> > */Brian West/*
>> > brian at freeswitch.org <mailto:brian at freeswitch.org>
>> >
>> >
>> > */Twitter: @FreeSWITCH , @briankwest/*
>> > http://www.freeswitchbook.com
>> > http://www.freeswitchcookbook.com
>> >
>> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/d349be50/attachment.html 

From jason at dickson.st  Tue Mar 10 06:19:22 2015
From: jason at dickson.st (Jason Lewis)
Date: Tue, 10 Mar 2015 14:19:22 +1100
Subject: [Freeswitch-users] How to configure G722 for internal and G729 for
	external calls
Message-ID: <54FE62BA.4070601@dickson.st>

hi,

I'm trying to get freeswitch to use G722 for internal calls and G729 for
external calls.

I'm using vanilla.

Am I missing something obvious here?


in vars.xml I have:

  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=OPUS,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>

when I place a call to an external number, it gets encoded as G722,
according to the CDR:

"1001","1001","938xxxxxx","default","2015-03-09 16:52:45","2015-03-09
16:52:48","2015-03-09
16:54:03","78","75","NORMAL_CLEARING","1ee1cfd1-f35c-4829-803a-771678574d07","cb8ee2cc-46bb-45d2-b302-effe038f195e","1001","G722","G722"

Dialplan:
  <extension name="pennytel-local_fixed_line">
    <condition field="destination_number" expression="^([2-9]{1}[0-9]{7})$">
      <action application="set" data="sip_h_X-accountcode=${accountcode}"/>
      <action application="set" data="call_direction=outbound"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="bridge"
data="{absolute_codec_string='G729'}sofia/gateway/pennytel/612$1"/>
    </condition>
  </extension>




restarting the external gateway shows the outbound codec preference is
there:

freeswitch at internal> sofia profile external restart reloadxml
Reload XML [Success]
restarting: external
freeswitch at internal> 2015-03-10 12:33:04.175760 [INFO] mod_enum.c:880
ENUM Reloaded
2015-03-10 12:33:04.175760 [INFO] switch_time.c:1411 Timezone reloaded
1781 definitions
2015-03-10 12:33:04.315790 [NOTICE] sofia_reg.c:135 UN-Registering sip2sip
2015-03-10 12:33:04.315790 [NOTICE] sofia_reg.c:135 UN-Registering pennytel
2015-03-10 12:33:05.315785 [NOTICE] sofia.c:3044 Waiting for worker thread
2015-03-10 12:33:05.315785 [INFO] switch_core_sqldb.c:1701
sofia:external Destroying SQL queue.
2015-03-10 12:33:05.515828 [INFO] switch_core_sqldb.c:1664
sofia:external Stopping SQL thread.
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:3099 Write lock external
2015-03-10 12:33:05.515828 [NOTICE] sofia_glue.c:1814 deleted gateway
example.com from profile external
2015-03-10 12:33:05.515828 [NOTICE] sofia_glue.c:1814 deleted gateway
sip2sip from profile external
2015-03-10 12:33:05.515828 [NOTICE] sofia_glue.c:1814 deleted gateway
pennytel from profile external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:3112 Write unlock external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 debug [0]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 sip-trace [no]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 sip-capture [no]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 rfc2833-pt [101]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 sip-port [5080]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 dialplan [XML]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 context [public]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 dtmf-duration [2000]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 inbound-codec-prefs
[OPUS,G722,PCMU,PCMA,GSM]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 outbound-codec-prefs [G729]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 hold-music
[local_stream://moh/8000]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 rtp-timer-name [soft]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 local-network-acl
[localnet.auto]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 manage-presence [false]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146
inbound-codec-negotiation [generous]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 nonce-ttl [60]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 auth-calls [false]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 inbound-late-negotiation
[true]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 inbound-zrtp-passthru [true]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 rtp-ip [10.0.2.145]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 sip-ip [10.0.2.145]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 ext-rtp-ip [aa.bb.cc.dd]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 ext-sip-ip [aa.bb.cc.dd]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 rtp-timeout-sec [300]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 rtp-hold-timeout-sec [1800]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls [false]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-only [false]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-bind-params
[transport=tls]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-sip-port [5081]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-passphrase []
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-verify-date [true]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-verify-policy [none]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-verify-depth [2]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-verify-in-subjects []
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:4146 tls-version
[tlsv1,tlsv1.1,tlsv1.2]
2015-03-10 12:33:05.515828 [NOTICE] sofia.c:5557 Started Profile
external [sofia_reg_external]
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:2755 Creating agent for external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:2873 Created agent for external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:2918 Set params for external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:2962 Activated db for external
2015-03-10 12:33:05.515828 [INFO] switch_core_sqldb.c:1679
sofia:external Starting SQL thread.
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:3000 Starting thread for external
2015-03-10 12:33:05.515828 [DEBUG] sofia.c:2655 Launching worker thread
for external
2015-03-10 12:33:05.515828 [NOTICE] sofia_reg.c:3329 Added gateway
'pennytel' to profile 'external'
2015-03-10 12:33:05.515828 [NOTICE] sofia_reg.c:3329 Added gateway
'sip2sip' to profile 'external'
2015-03-10 12:33:05.515828 [NOTICE] sofia_reg.c:3329 Added gateway
'example.com' to profile 'external'
2015-03-10 12:33:05.515828 [NOTICE] sofia_reg.c:448 Registering sip2sip
2015-03-10 12:33:05.515828 [NOTICE] sofia_reg.c:448 Registering pennytel




Dialing out, I get these logs:

2015-03-10 12:37:37.615813 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.615813 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.615813 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_NEW
2015-03-10 12:37:37.635769 [DEBUG] sofia.c:8834
sofia/internal/1001 at freeswitch.xyz.com.au receiving invite from
10.0.2.129:5062 version: 1.4.15 -1 64bit
2015-03-10 12:37:37.635769 [DEBUG] sofia.c:9001 IP 10.0.2.129 Rejected
by acl "domains". Falling back to Digest auth.
2015-03-10 12:37:37.635769 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1001 at freeswitch.xyz.com.au) State NEW
2015-03-10 12:37:37.635769 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.635769 [DEBUG] sofia.c:2067 detaching session
38291425-a146-4aa2-a129-854757744bc1
2015-03-10 12:37:37.755808 [DEBUG] sofia.c:2175 Re-attaching to session
38291425-a146-4aa2-a129-854757744bc1
2015-03-10 12:37:37.755808 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.755808 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] sofia.c:8834
sofia/internal/1001 at freeswitch.xyz.com.au receiving invite from
10.0.2.129:5062 version: 1.4.15 -1 64bit
2015-03-10 12:37:37.775786 [DEBUG] sofia.c:9001 IP 10.0.2.129 Rejected
by acl "domains". Falling back to Digest auth.
2015-03-10 12:37:37.775786 [DEBUG] sofia.c:6614 Channel
sofia/internal/1001 at freeswitch.xyz.com.au entering state [received][100]
2015-03-10 12:37:37.775786 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=- 20183 20183 IN IP4 10.0.2.129
s=SDP data
c=IN IP4 10.0.2.129
t=0 0
m=audio 11794 RTP/AVP 18 9 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2015-03-10 12:37:37.775786 [DEBUG] sofia.c:6890
(sofia/internal/1001 at freeswitch.xyz.com.au) State Change CS_NEW -> CS_INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:512
(sofia/internal/1001 at freeswitch.xyz.com.au) State INIT
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:87
sofia/internal/1001 at freeswitch.xyz.com.au SOFIA INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:40
sofia/internal/1001 at freeswitch.xyz.com.au Standard INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:48
(sofia/internal/1001 at freeswitch.xyz.com.au) State Change CS_INIT ->
CS_ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:512
(sofia/internal/1001 at freeswitch.xyz.com.au) State INIT going to sleep
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_channel.c:2184
(sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change DOWN -> RINGING
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:528
(sofia/internal/1001 at freeswitch.xyz.com.au) State ROUTING
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:123
sofia/internal/1001 at freeswitch.xyz.com.au SOFIA ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:166
sofia/internal/1001 at freeswitch.xyz.com.au Standard ROUTING
2015-03-10 12:37:37.775786 [INFO] mod_dialplan_xml.c:635 Processing 1001
<1001>->775 in context default
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->unloop] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS)
[unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->tod_example] continue=true
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Date/Time Match
(PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action set(open=true)
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->holiday_example] continue=true
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Date/TimeMatch
(FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[global-intercept] destination_number(775) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[group-intercept] destination_number(775) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[intercept-ext] destination_number(775) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->redial] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[redial] destination_number(775) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->global] continue=true
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[global] ${default_password}(18651) =~ /^1234$/ break=never
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[global] ${rtp_has_crypto}() =~
/^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
break=never
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS)
[global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED
NEGOTIATION)/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[global] ${switch_r_sdp}(v=0
o=- 20183 20183 IN IP4 10.0.2.129
s=SDP data
c=IN IP4 10.0.2.129
t=0 0
m=audio 11794 RTP/AVP 18 9 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Absolute Condition
[global]
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->snom-demo-2] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[snom-demo-2] destination_number(775) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->snom-demo-1] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[snom-demo-1] destination_number(775) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[eavesdrop] destination_number(775) =~ /^88(\d{4})$|^\*0(.*)$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[eavesdrop] destination_number(775) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->call_return] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[call_return] destination_number(775) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->del-group] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[del-group] destination_number(775) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->add-group] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[add-group] destination_number(775) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[call-group-simo] destination_number(775) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[call-group-order] destination_number(775) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[extension-intercom] destination_number(775) =~ /^8(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->SIPTAPI-AutoAnswer] continue=true
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[SIPTAPI-AutoAnswer] ${sip_user_agent}(Yealink SIP-T42G 29.73.0.45) =~
/.*SIPTAPI.*/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[Local_Extension] destination_number(775) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[Local_Extension_Skinny] destination_number(775) =~ /^(11[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[group_dial_sales] destination_number(775) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[group_dial_support] destination_number(775) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[group_dial_billing] destination_number(775) =~ /^2002$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->group_dial_custom] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[group_dial_custom] destination_number(775) =~ /^2003$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->operator] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[operator] destination_number(775) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->vmain] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [vmain]
destination_number(775) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->sip_uri] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[sip_uri] destination_number(775) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[nb_conferences] destination_number(775) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[wb_conferences] destination_number(775) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[uwb_conferences] destination_number(775) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[cdquality_conferences] destination_number(775) =~ /^(33\d{2})$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(775) =~
/^9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[mad_boss_intercom] destination_number(775) =~ /^0911$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->mad_boss_intercom] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[mad_boss_intercom] destination_number(775) =~ /^0912$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->mad_boss] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[mad_boss] destination_number(775) =~ /^0913$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ivr_demo] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ivr_demo] destination_number(775) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[dynamic_conference] destination_number(775) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[rtp_multicast_page] destination_number(775) =~ /^pagegroup$|^7243$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->park] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [park]
destination_number(775) =~ /^5900$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->unpark] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[unpark] destination_number(775) =~ /^5901$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->valet_park] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[valet_park] destination_number(775) =~ /^(6000)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->valet_park] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[valet_park] destination_number(775) =~ /^((?!6000)60\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->park] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [park]
destination_number(775) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->unpark] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS)
[unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[unpark] destination_number(775) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->park] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [park]
destination_number(775) =~ /callpark/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->unpark] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS)
[unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[unpark] destination_number(775) =~ /pickup/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->wait] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [wait]
destination_number(775) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->fax_receive] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[fax_receive] destination_number(775) =~ /^9178$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->fax_transmit] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[fax_transmit] destination_number(775) =~ /^9179$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ringback_180] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ringback_180] destination_number(775) =~ /^9180$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ringback_183_uk_ring] destination_number(775) =~ /^9181$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ringback_183_music_ring] destination_number(775) =~ /^9182$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(775) =~ /^9183$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ringback_post_answer_music] destination_number(775) =~ /^9184$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->ClueCon] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[ClueCon] destination_number(775) =~ /^9191$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->show_info] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[show_info] destination_number(775) =~ /^9192$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->video_record] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[video_record] destination_number(775) =~ /^9193$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->video_playback] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[video_playback] destination_number(775) =~ /^9194$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->delay_echo] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[delay_echo] destination_number(775) =~ /^9195$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->echo] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [echo]
destination_number(775) =~ /^9196$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->milliwatt] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[milliwatt] destination_number(775) =~ /^9197$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->tone_stream] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[tone_stream] destination_number(775) =~ /^9198$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[zrtp_enrollement] destination_number(775) =~ /^9787$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->hold_music] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[hold_music] destination_number(775) =~ /^9664$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->laugh break] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [laugh
break] destination_number(775) =~ /^9386$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->101] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL) [101]
destination_number(775) =~ /^101$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pizza_demo] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pizza_demo] destination_number(775) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->Talking Clock Time] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[Talking Clock Time] destination_number(775) =~ /^9170$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->Talking Clock Date] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[Talking Clock Date] destination_number(775) =~ /^9171$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->Talking Clock Date and Time] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[Talking Clock Date and Time] destination_number(775) =~ /^9172$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-local_fixed_line] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pennytel-local_fixed_line] destination_number(775) =~
/^([2-9]{1}[0-9]{7})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-national_fixed_line] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pennytel-national_fixed_line] destination_number(775) =~
/^(0|61|\+61)?([2?|3|5-9]{1}[0-9]{8})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-national_mobiles] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pennytel-national_mobiles] destination_number(775) =~
/^(0|61|\+61)?(4{1}[0-9]{8})$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-13] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pennytel-13] destination_number(775) =~ /^13(\d{4}|00\d{6})$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-18] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (FAIL)
[pennytel-18] destination_number(775) =~ /^180(\d{4}|0\d{6})$/
break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au parsing
[default->pennytel-test] continue=false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Regex (PASS)
[pennytel-test] destination_number(775) =~ /^775$/ break=on-false
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
set(sip_h_X-accountcode=${accountcode})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
set(call_direction=outbound)
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/internal/1001 at freeswitch.xyz.com.au Action
bridge({absolute_codec_string='G729'}sofia/gateway/pennytel/775)
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:216
(sofia/internal/1001 at freeswitch.xyz.com.au) State Change CS_ROUTING ->
CS_EXECUTE
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:528
(sofia/internal/1001 at freeswitch.xyz.com.au) State ROUTING going to sleep
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_EXECUTE
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:535
(sofia/internal/1001 at freeswitch.xyz.com.au) State EXECUTE
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:178
sofia/internal/1001 at freeswitch.xyz.com.au SOFIA EXECUTE
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:258
sofia/internal/1001 at freeswitch.xyz.com.au Standard EXECUTE
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au set(open=true)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET [open]=[true]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
hash(insert/freeswitch.xyz.com.au-spymap/1001/38291425-a146-4aa2-a129-854757744bc1)
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
hash(insert/freeswitch.xyz.com.au-last_dial/1001/775)
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
hash(insert/freeswitch.xyz.com.au-last_dial/global/38291425-a146-4aa2-a129-854757744bc1)
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
export(RFC2822_DATE=Tue, 10 Mar 2015 12:37:37 +1100)
2015-03-10 12:37:37.775786 [DEBUG] switch_channel.c:1247 EXPORT
(export_vars) [RFC2822_DATE]=[Tue, 10 Mar 2015 12:37:37 +1100]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
set(sip_h_X-accountcode=1001)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET [sip_h_X-accountcode]=[1001]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
set(call_direction=outbound)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET [call_direction]=[outbound]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
set(hangup_after_bridge=true)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
set(effective_caller_id_name=FreeSWITCH)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET
[effective_caller_id_name]=[FreeSWITCH]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
set(effective_caller_id_number=0000000000)
2015-03-10 12:37:37.775786 [DEBUG] mod_dptools.c:1435
sofia/internal/1001 at freeswitch.xyz.com.au SET
[effective_caller_id_number]=[0000000000]
EXECUTE sofia/internal/1001 at freeswitch.xyz.com.au
bridge({absolute_codec_string='G729'}sofia/gateway/pennytel/775)
2015-03-10 12:37:37.775786 [DEBUG] switch_channel.c:1201
sofia/internal/1001 at freeswitch.xyz.com.au EXPORTING[export_vars]
[RFC2822_DATE]=[Tue, 10 Mar 2015 12:37:37 +1100] to event
2015-03-10 12:37:37.775786 [DEBUG] switch_ivr_originate.c:2103 Parsing
global variables
2015-03-10 12:37:37.775786 [DEBUG] switch_event.c:1688 Parsing variable
[absolute_codec_string]=[G729]
2015-03-10 12:37:37.775786 [NOTICE] switch_channel.c:1055 New Channel
sofia/external/775 [a6bc5be8-d538-4b72-892f-6e7c26f6dea6]
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4627 (sofia/external/775)
State Change CS_NEW -> CS_INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4697 [zrtp_passthru]
Setting a-leg inherit_codec=true
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4700 [zrtp_passthru]
Setting b-leg
absolute_codec_string='G722 at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:512
(sofia/external/775) State INIT
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:87 sofia/external/775
SOFIA INIT
2015-03-10 12:37:37.775786 [DEBUG] sofia_glue.c:1232 sofia/external/775
sending invite version: 1.4.15 -1 64bit
Local SDP:
v=0
o=FreeSWITCH 1425927473 1425927474 IN IP4 aa.bbb.ccc.ddd
s=FreeSWITCH
c=IN IP4 aa.bbb.ccc.ddd
t=0 0
m=audio 23984 RTP/AVP 18 101 13
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:40
sofia/external/775 Standard INIT
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:48
(sofia/external/775) State Change CS_INIT -> CS_ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:512
(sofia/external/775) State INIT going to sleep
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] sofia.c:6614 Channel
sofia/external/775 entering state [calling][0]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:528
(sofia/external/775) State ROUTING
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:123 sofia/external/775
SOFIA ROUTING
2015-03-10 12:37:37.775786 [DEBUG] switch_ivr_originate.c:67
(sofia/external/775) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-03-10 12:37:37.775786 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:528
(sofia/external/775) State ROUTING going to sleep
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_CONSUME_MEDIA
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:547
(sofia/external/775) State CONSUME_MEDIA
2015-03-10 12:37:37.775786 [DEBUG] switch_core_state_machine.c:547
(sofia/external/775) State CONSUME_MEDIA going to sleep
2015-03-10 12:37:38.315813 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.315813 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.315813 [DEBUG] sofia.c:6614 Channel
sofia/external/775 entering state [calling][0]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6614 Channel
sofia/external/775 entering state [completing][200]
2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=Sippy 2433467651418324771 2 IN IP4 202.85.243.105
s=session
t=0 0
m=audio 10948 RTP/AVP 18 101
c=IN IP4 202.85.243.53
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

2015-03-10 12:37:38.735794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6614 Channel
sofia/external/775 entering state [ready][200]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec
Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3543 Set
telephone-event payload to 101
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:2473 Set Codec
sofia/external/775 G729/8000 20 ms 160 samples 8000 bits 1 channels
2015-03-10 12:37:38.735794 [DEBUG] switch_core_codec.c:111
sofia/external/775 Original read codec set to G729:18
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3881 Set 2833
dtmf send payload to 101
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5141 AUDIO RTP
[sofia/external/775] 10.0.2.145 port 23984 -> 202.85.243.53 port 10948
codec: 18 ms: 20
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:3548 Starting timer
[soft] 160 bytes per 20ms
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
START SESSION INITIALIZATION. sID=61.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ZID=306130303032393164366363.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
Loading User's profile:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
allowclear: OFF
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
autosecure: ON
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]: 
disclose_bit: OFF
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]: 
signal. role: Initiator
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:           TTL: 4294967295
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
SAS schemes:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 B256
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 B32
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:      Ciphers:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 AES3
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 AES1
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
PK schemes:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 EC25
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 DH3k
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 DH2k
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 Mult
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:           ATL:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 HS32
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:       Hashes:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357 S256
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
Session initialization - DONE. sID=61.

2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ATTACH NEW STREAM to sID=61:
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
Empty slot was found - initializing new stream with ID=61.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
Preparing ZRTP Hello according to the Session profile.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ATTACH NEW STREAM - DONE.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
START STREAM ID=61 mode=CLEAR state=ACTIVE.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=61 CLEAR switching <ACTIVE> ---> <START>.
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6662 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5439 Set 2833
dtmf send payload to 101
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5445 Set 2833
dtmf receive payload to 101
2015-03-10 12:37:38.735794 [DEBUG] switch_channel.c:3635 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.735794 [NOTICE] sofia.c:7475 Channel
[sofia/external/775] has been answered
2015-03-10 12:37:38.735794 [DEBUG] switch_channel.c:3689
(sofia/external/775) Callstate Change DOWN -> ACTIVE
2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec
string G729 at 8000h@20i not supported on
sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec
Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec
Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3543 Set
telephone-event payload to 101
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:2473 Set Codec
sofia/internal/1001 at freeswitch.xyz.com.au G722/8000 20 ms 160 samples
64000 bits 1 channels
2015-03-10 12:37:38.755785 [DEBUG] switch_core_codec.c:111
sofia/internal/1001 at freeswitch.xyz.com.au Original read codec set to G722:9
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:3890 Set 2833
dtmf send/recv payload to 101
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5141 AUDIO RTP
[sofia/internal/1001 at freeswitch.xyz.com.au] 10.0.2.145 port 20762 ->
10.0.2.129 port 11794 codec: 9 ms: 20
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:3548 Starting timer
[soft] 160 bytes per 20ms
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
START SESSION INITIALIZATION. sID=62.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ZID=306130303032393164366363.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
Loading User's profile:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
allowclear: OFF
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
autosecure: ON
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]: 
disclose_bit: OFF
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]: 
signal. role: Unknown
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:           TTL: 4294967295
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
SAS schemes:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 B256
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 B32
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:      Ciphers:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 AES3
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 AES1
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:   
PK schemes:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 EC25
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 DH3k
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 DH2k
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 Mult
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:           ATL:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 HS32
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp
main]:       Hashes:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357 S256
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
Session initialization - DONE. sID=62.

2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ATTACH NEW STREAM to sID=62:
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
Empty slot was found - initializing new stream with ID=62.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:  
Preparing ZRTP Hello according to the Session profile.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [   zrtp main]:
ATTACH NEW STREAM - DONE.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
START STREAM ID=62 mode=CLEAR state=ACTIVE.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=62 CLEAR switching <ACTIVE> ---> <START>.
2015-03-10 12:37:38.755785 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16177 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5439 Set 2833
dtmf send payload to 101
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5445 Set 2833
dtmf receive payload to 101
2015-03-10 12:37:38.755785 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1001 at freeswitch.xyz.com.au!
2015-03-10 12:37:38.755785 [DEBUG] switch_channel.c:3399
(sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change RINGING ->
EARLY
2015-03-10 12:37:38.755785 [DEBUG] mod_sofia.c:780 Local SDP
sofia/internal/1001 at freeswitch.xyz.com.au:
v=0
o=FreeSWITCH 1425930696 1425930697 IN IP4 10.0.2.145
s=FreeSWITCH
c=IN IP4 10.0.2.145
t=0 0
m=audio 20762 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-03-10 12:37:38.755785 [DEBUG] switch_core_session.c:908 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.755785 [NOTICE] switch_ivr_originate.c:3522 Channel
[sofia/internal/1001 at freeswitch.xyz.com.au] has been answered
2015-03-10 12:37:38.755785 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.755785 [DEBUG] switch_channel.c:3689
(sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change EARLY -> ACTIVE
2015-03-10 12:37:38.755785 [DEBUG] sofia.c:6614 Channel
sofia/internal/1001 at freeswitch.xyz.com.au entering state [completed][200]
2015-03-10 12:37:38.755785 [DEBUG] switch_ivr_originate.c:3580 Originate
Resulted in Success: [sofia/external/775]
2015-03-10 12:37:38.755785 [DEBUG] switch_core_session.c:908 Send signal
sofia/external/775 [BREAK]
2015-03-10 12:37:38.755785 [DEBUG] switch_core_session.c:908 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.755785 [DEBUG] switch_ivr_bridge.c:1465
(sofia/external/775) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-03-10 12:37:38.755785 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:38.755785 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_EXCHANGE_MEDIA
2015-03-10 12:37:38.755785 [DEBUG] switch_core_state_machine.c:538
(sofia/external/775) State EXCHANGE_MEDIA
2015-03-10 12:37:38.755785 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
2015-03-10 12:37:38.795788 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.795788 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.795788 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.795788 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6663 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:38.815813 [DEBUG] sofia.c:6614 Channel
sofia/internal/1001 at freeswitch.xyz.com.au entering state [ready][200]
2015-03-10 12:37:38.815813 [DEBUG] switch_core_session.c:970 Send signal
sofia/external/775 [BREAK]
2015-03-10 12:37:38.815813 [DEBUG] switch_core_session.c:970 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.815813 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16178 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:38.835797 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.835797 [DEBUG] sofia.c:6614 Channel
sofia/internal/1001 at freeswitch.xyz.com.au entering state [calling][0]
2015-03-10 12:37:38.895793 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6664 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:38.915789 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16179 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:38.935794 [DEBUG] switch_rtp.c:5853 Correct ip/port
confirmed.
2015-03-10 12:37:38.935794 [NOTICE] switch_core_io.c:1261 Activating
write resampler
2015-03-10 12:37:38.935794 [INFO] mod_com_g729.c:126 ENCODER LICENSE
ALLOCATED--->0x7fc5640abbf0 0x7fc5640abbf0
2015-03-10 12:37:38.935794 [INFO] mod_com_g729.c:133 ENCODER
CREATED------------->0x7fc5640abbf0 0x7fc5640abbf0
2015-03-10 12:37:38.935794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.935794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.935794 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:38.955811 [DEBUG] sofia.c:6614 Channel
sofia/internal/1001 at freeswitch.xyz.com.au entering state [ready][200]
2015-03-10 12:37:38.955811 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=- 20183 20184 IN IP4 10.0.2.129
s=SDP data
c=IN IP4 10.0.2.129
t=0 0
a=sendrecv
m=audio 11794 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3682 Audio Codec
Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3627 Audio Codec
Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3543 Set
telephone-event payload to 101
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:3890 Set 2833
dtmf send/recv payload to 101
2015-03-10 12:37:38.955811 [DEBUG] sofia.c:7318 Processing updated SDP
2015-03-10 12:37:38.955811 [DEBUG] switch_core_media.c:5124 Audio params
are unchanged for sofia/internal/1001 at freeswitch.xyz.com.au.
2015-03-10 12:37:39.095797 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6665 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:39.115795 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16180 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:39.295792 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6666 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:39.315799 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16181 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:39.495820 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
WARNING! HELLO have been resent 5 times without a response. Raising
ZRTP_EVENT_NO_ZRTP_QUICK event. ID=61
2015-03-10 12:37:39.495820 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6667 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:39.515800 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
WARNING! HELLO have been resent 5 times without a response. Raising
ZRTP_EVENT_NO_ZRTP_QUICK event. ID=62
2015-03-10 12:37:39.515800 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16182 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:39.635793 [DEBUG] switch_rtp.c:5853 Correct ip/port
confirmed.
2015-03-10 12:37:39.635793 [INFO] mod_com_g729.c:164 DECODER LICENSE
ALLOCATED--->0x7fc5640abb70 0x7fc5640abb78
2015-03-10 12:37:39.635793 [INFO] mod_com_g729.c:171 DECODER
CREATED------------->0x7fc5640abb70 0x7fc5640abb78
2015-03-10 12:37:39.635793 [NOTICE] switch_core_io.c:1261 Activating
write resampler
2015-03-10 12:37:39.695794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6668 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:39.715818 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16183 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:39.895794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6669 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:39.915788 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16184 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:40.095795 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6670 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:40.115797 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16185 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:40.295795 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6671 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:40.315799 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16186 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:40.495794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6672 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:40.515796 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16187 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:40.695794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6673 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:40.715793 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16188 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:40.895795 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6674 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:40.915814 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16189 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:41.095793 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6675 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:41.115793 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16190 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:41.295810 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6676 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:41.315792 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16191 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:41.495802 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6677 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:41.515827 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16192 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:41.695792 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6678 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:41.715791 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16193 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:41.895794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6679 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:41.915786 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16194 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:42.095794 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6680 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:42.115792 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16195 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:42.295796 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=2970911498 seq=6681 size=144. Stream 61:CLEAR:START
2015-03-10 12:37:42.315793 [DEBUG] switch_rtp.c:1357  [  zrtp utils]:  
Send <HELLO> ssrc=1447443834 seq=16196 size=144. Stream 62:CLEAR:START
2015-03-10 12:37:42.495796 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
WARNING! HELLO Max retransmissions count reached (20 retries). ID=61
2015-03-10 12:37:42.495796 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=61 CLEAR switching <START> ---> <NOZRTP>.
2015-03-10 12:37:42.515800 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
WARNING! HELLO Max retransmissions count reached (20 retries). ID=62
2015-03-10 12:37:42.515800 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=62 CLEAR switching <START> ---> <NOZRTP>.
2015-03-10 12:37:43.675790 [DEBUG] switch_core_session.c:1053 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:43.695793 [NOTICE] sofia.c:952 Hangup
sofia/internal/1001 at freeswitch.xyz.com.au [CS_EXECUTE] [NORMAL_CLEARING]
2015-03-10 12:37:43.695793 [DEBUG] switch_channel.c:3222 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [KILL]
2015-03-10 12:37:43.695793 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:43.695793 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD
DONE [sofia/internal/1001 at freeswitch.xyz.com.au]
2015-03-10 12:37:43.695793 [DEBUG] switch_ivr_bridge.c:690 Send signal
sofia/external/775 [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_ivr_bridge.c:579
sofia/internal/1001 at freeswitch.xyz.com.au ending bridge by request from
write function
2015-03-10 12:37:43.715797 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD
DONE [sofia/external/775]
2015-03-10 12:37:43.715797 [DEBUG] switch_ivr_bridge.c:690 Send signal
sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:43.715797 [NOTICE] switch_ivr_bridge.c:754 Hangup
sofia/external/775 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-03-10 12:37:43.715797 [DEBUG] switch_channel.c:3222 Send signal
sofia/external/775 [KILL]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:538
(sofia/external/775) State EXCHANGE_MEDIA going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_HANGUP
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:735
(sofia/external/775) Callstate Change ACTIVE -> HANGUP
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:737
(sofia/external/775) State HANGUP
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:407 sofia/external/775
Overriding SIP cause 480 with 200 from the other leg
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:413 Channel
sofia/external/775 hanging up, cause: NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:465 Sending BYE to
sofia/external/775
2015-03-10 12:37:43.715797 [DEBUG] switch_ivr_bridge.c:1563
sofia/external/775 skip receive message [UNBRIDGE] (channel is hungup
already)
2015-03-10 12:37:43.715797 [DEBUG] switch_ivr_bridge.c:1566
sofia/internal/1001 at freeswitch.xyz.com.au skip receive message
[UNBRIDGE] (channel is hungup already)
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:60
sofia/external/775 Standard HANGUP, cause: NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:737
(sofia/external/775) State HANGUP going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:504
(sofia/external/775) State Change CS_HANGUP -> CS_REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:2893
sofia/internal/1001 at freeswitch.xyz.com.au skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:472
(sofia/external/775) Running State Change CS_REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:535
(sofia/internal/1001 at freeswitch.xyz.com.au) State EXECUTE going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_HANGUP
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:823
(sofia/external/775) State REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:104
sofia/external/775 Standard REPORTING, cause: NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:823
(sofia/external/775) State REPORTING going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:735
(sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change ACTIVE ->
HANGUP
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:737
(sofia/internal/1001 at freeswitch.xyz.com.au) State HANGUP
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:498
(sofia/external/775) State Change CS_REPORTING -> CS_DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/775 [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1615 Session 66
(sofia/external/775) Locked, Waiting on external entities
2015-03-10 12:37:43.715797 [NOTICE] switch_core_session.c:1633 Session
66 (sofia/external/775) Ended
2015-03-10 12:37:43.715797 [NOTICE] switch_core_session.c:1637 Close
Channel sofia/external/775 [CS_DESTROY]
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:413 Channel
sofia/internal/1001 at freeswitch.xyz.com.au hanging up, cause: NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:626
(sofia/external/775) Running State Change CS_DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:636
(sofia/external/775) State DESTROY
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:323 sofia/external/775
SOFIA DESTROY
2015-03-10 12:37:43.715797 [INFO] mod_com_g729.c:95 DECODER
DESTROYED----------->0x7fc5640abb70 0x7fc5640abb78
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at freeswitch.xyz.com.au Standard HANGUP, cause:
NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:737
(sofia/internal/1001 at freeswitch.xyz.com.au) State HANGUP going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/1001 at freeswitch.xyz.com.au) State Change CS_HANGUP ->
CS_REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change
CS_REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:823
(sofia/internal/1001 at freeswitch.xyz.com.au) State REPORTING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:104
sofia/internal/1001 at freeswitch.xyz.com.au Standard REPORTING, cause:
NORMAL_CLEARING
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:823
(sofia/internal/1001 at freeswitch.xyz.com.au) State REPORTING going to sleep
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:498
(sofia/internal/1001 at freeswitch.xyz.com.au) State Change CS_REPORTING ->
CS_DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1388 Send
signal sofia/internal/1001 at freeswitch.xyz.com.au [BREAK]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_session.c:1615 Session 65
(sofia/internal/1001 at freeswitch.xyz.com.au) Locked, Waiting on external
entities
2015-03-10 12:37:43.715797 [NOTICE] switch_core_session.c:1633 Session
65 (sofia/internal/1001 at freeswitch.xyz.com.au) Ended
2015-03-10 12:37:43.715797 [NOTICE] switch_core_session.c:1637 Close
Channel sofia/internal/1001 at freeswitch.xyz.com.au [CS_DESTROY]
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:626
(sofia/internal/1001 at freeswitch.xyz.com.au) Running State Change CS_DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:636
(sofia/internal/1001 at freeswitch.xyz.com.au) State DESTROY
2015-03-10 12:37:43.715797 [DEBUG] mod_sofia.c:323
sofia/internal/1001 at freeswitch.xyz.com.au SOFIA DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=62 mode=CLEAR state=NOZRTP.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=0 UNKNOWN switching <NONE> ---> <NONE>.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:111
sofia/internal/1001 at freeswitch.xyz.com.au Standard DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:636
(sofia/internal/1001 at freeswitch.xyz.com.au) State DESTROY going to sleep
2015-03-10 12:37:43.715797 [INFO] mod_com_g729.c:98 DECODER LICENSE
DEALLOCATED->0x7fc5640abb70 0x7fc5640abb78
2015-03-10 12:37:43.715797 [INFO] mod_com_g729.c:84 ENCODER
DESTROYED----------->0x7fc5640abbf0 0x7fc5640abbf0
2015-03-10 12:37:43.715797 [INFO] mod_com_g729.c:87 ENCODER LICENSE
DEALLOCATED->0x7fc5640abbf0 0x7fc5640abbf0
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=61 mode=CLEAR state=NOZRTP.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [        zrtp]:  
Stream ID=0 UNKNOWN switching <NONE> ---> <NONE>.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-03-10 12:37:43.715797 [DEBUG] switch_rtp.c:1357  [ zrtp engine]:
STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:111
sofia/external/775 Standard DESTROY
2015-03-10 12:37:43.715797 [DEBUG] switch_core_state_machine.c:636
(sofia/external/775) State DESTROY going to sleep



-- 
Jason Lewis
http://emacstragic.net


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 834 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/599123d0/attachment-0001.bin 

From bote_radio at botecomm.com  Tue Mar 10 08:46:43 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 10 Mar 2015 01:46:43 -0400
Subject: [Freeswitch-users] How to configure G722 for internal and G729
	for	external calls
In-Reply-To: <54FE62BA.4070601@dickson.st>
References: <54FE62BA.4070601@dickson.st>
Message-ID: <049801d05af5$96f684d0$c4e38e70$@botecomm.com>

1) Please don't post extensive log output to the mailing list. The developers much prefer that you use the FreeSWITCH pastebin and choose the FreeSWITCH log syntax highlighting:
https://pastebin.freeswitch.org/
and pay close attention to the instructions in the prompt for credentials that pops up.

2) I see:
2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance

Did you license G.729 codec? Because FS will need to transcode from the G.722 used internally to G.729 towards your carrier. My guess is that this is where your problem lies.

With the vanilla configuration I believe that FS can agree on G.729 with the far end as long as it is merely passing the RTP stream through the switch untouched. Once FS needs to transcode it between G.729 and G.722 you need to fork over money for the number of simultaneous G.729 calls that you expect since it is a commercially restricted codec.

Details on the Confluence wiki at:
https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729

Of course, I could be totally wrong about this, but if you are down under then you'll be asleep when the everybody else wakes up so I figure I'd give it a stab to give you a head-start.

Bote

 


-----Original Message-----
From: Jason Lewis
Sent: Monday, 09 March, 2015 23:19
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How to configure G722 for internal and G729 for external calls

hi,

I'm trying to get freeswitch to use G722 for internal calls and G729 for
external calls.

I'm using vanilla.

Am I missing something obvious here?
.
.
.
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4697 [zrtp_passthru] Setting a-leg inherit_codec=true
2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4700 [zrtp_passthru] Setting b-leg absolute_codec_string='G722 at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
...
2015-03-10 12:37:37.775786 [DEBUG] sofia_glue.c:1232 sofia/external/775 sending invite version: 1.4.15 -1 64bit
Local SDP:
v=0
o=FreeSWITCH 1425927473 1425927474 IN IP4 aa.bbb.ccc.ddd
s=FreeSWITCH
c=IN IP4 aa.bbb.ccc.ddd
t=0 0
m=audio 23984 RTP/AVP 18 101 13
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6624 Remote SDP:
v=0
o=Sippy 2433467651418324771 2 IN IP4 202.85.243.105
s=session
t=0 0
m=audio 10948 RTP/AVP 18 101
c=IN IP4 202.85.243.53
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:2473 Set Codec sofia/external/775 G729/8000 20 ms 160 samples 8000 bits 1 channels
2015-03-10 12:37:38.735794 [DEBUG] switch_core_codec.c:111 sofia/external/775 Original read codec set to G729:18
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/external/775] 10.0.2.145 port 23984 -> 202.85.243.53 port 10948 codec: 18 ms: 20
2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 160 bytes per 20ms
...
2015-03-10 12:37:38.735794 [NOTICE] sofia.c:7475 Channel [sofia/external/775] has been answered
2015-03-10 12:37:38.735794 [DEBUG] switch_channel.c:3689 (sofia/external/775) Callstate Change DOWN -> ACTIVE
2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1001 at freeswitch.xyz.com.au G722/8000 20 ms 160 samples 64000 bits 1 channels
2015-03-10 12:37:38.755785 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at freeswitch.xyz.com.au Original read codec set to G722:9
2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1001 at freeswitch.xyz.com.au] 10.0.2.145 port 20762 -> 10.0.2.129 port 11794 codec: 9 ms: 20
2015-03-10 12:37:38.755785 [DEBUG] switch_channel.c:3399 (sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change RINGING -> EARLY
2015-03-10 12:37:38.755785 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1001 at freeswitch.xyz.com.au:
v=0
o=FreeSWITCH 1425930696 1425930697 IN IP4 10.0.2.145
s=FreeSWITCH
c=IN IP4 10.0.2.145
t=0 0
m=audio 20762 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



-- 
Jason Lewis
http://emacstragic.net





From richard.mace at gmail.com  Tue Mar 10 09:17:29 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 10 Mar 2015 06:17:29 +0000
Subject: [Freeswitch-users] Realtime sip registrations
Message-ID: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>

Hi All,
Is it possible to see when sip registrations happen in real time?

Thanks

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/2e679aac/attachment.html 

From richard.mace at gmail.com  Tue Mar 10 09:19:46 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 10 Mar 2015 06:19:46 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
Message-ID: <CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>

Hi,
Is there an fs_cli command that I can use that will get me round the
current bug?
Its strange as it's only happened within the last month, as I built a
system recently that worked fine out of the box.

Thanks

Richard

On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:

> Hi Brian,
> Removed the line, and rebooted, but still getting:
>
> root at FreeSWITCH:~# fs_cli
> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>
> Richard
>
> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>
>> remove
>>
>> ::1     localhost ip6-localhost ip6-loopback
>>
>>
>> from /etc/hosts
>>
>>
>> its a bug in debian.
>>
>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> Sorry, I should have clarified that this is running locally on the
>>> machine running FreeSWITCH.
>>>
>>> Richard
>>>
>>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>>
>>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>>
>>>>
>>>>
>>>> To connect from a remote machine put a valid routable interface address
>>>> (although I have 0.0.0.0 in mine) in
>>>>
>>>> conf/autoload_configs/event_socket.conf.xml
>>>>
>>>>
>>>>
>>>> and change the password and maybe even the port depending on the
>>>> crackability of your network.
>>>>
>>>>
>>>>
>>>> Then you?ll probably want to configure a profile configuration file
>>>> with tight permissions to avoid having to type the parameters on the
>>>> command line every time you start fs_cli.
>>>>
>>>>
>>>>
>>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>>> details.
>>>>
>>>>
>>>>
>>>> Bote
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *From:* Richard Mace
>>>> *Sent:* Friday, 06 March, 2015 14:34
>>>> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install
>>>> on Debian
>>>>
>>>>
>>>>
>>>> Hi All,
>>>>
>>>> I did a fresh install of both Debian and FreeSWITCH today, following
>>>> the article here:
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>>
>>>>
>>>>
>>>> However, after installation, fs_cli will not connect. Any ideas?
>>>>
>>>>
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> Richard
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/0173d15d/attachment-0001.html 

From soulofmischief87 at gmail.com  Tue Mar 10 09:23:43 2015
From: soulofmischief87 at gmail.com (Tito Cumpen)
Date: Tue, 10 Mar 2015 02:23:43 -0400
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
	<CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
Message-ID: <CAPJLwtVkBwpuWX_CpKXUq9Ebbh8jY0qtbVhaHAufwibQeV_tLw@mail.gmail.com>

If you have issues getting to the console. Use ngrep or sipgrep check this
wiki out https://wiki.freeswitch.org/wiki/Packet_Capture.
On Mar 10, 2015 2:21 AM, "Tito Cumpen" <soulofmischief87 at gmail.com> wrote:

> Richard,
>
> You may view registrations through the fs_cli console. You can get very
> insightful debug ibformation through Sofia.check out Sofia debug
> http://wiki.freeswitch.org/wiki/Sofia-SIP
> On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:
>
>> Hi All,
>> Is it possible to see when sip registrations happen in real time?
>>
>> Thanks
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/170e2a8b/attachment.html 

From soulofmischief87 at gmail.com  Tue Mar 10 09:21:14 2015
From: soulofmischief87 at gmail.com (Tito Cumpen)
Date: Tue, 10 Mar 2015 02:21:14 -0400
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
Message-ID: <CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>

Richard,

You may view registrations through the fs_cli console. You can get very
insightful debug ibformation through Sofia.check out Sofia debug
http://wiki.freeswitch.org/wiki/Sofia-SIP
On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:

> Hi All,
> Is it possible to see when sip registrations happen in real time?
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/768a6a95/attachment.html 

From bote_radio at botecomm.com  Tue Mar 10 10:11:21 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 10 Mar 2015 03:11:21 -0400
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
	on	Debian
In-Reply-To: <CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>	<00c301d05848$833f55c0$89be0140$@botecomm.com>	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
Message-ID: <04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>

You could explicitly direct fs_cli to a particular i.p. address either on the command line or using a profile definition.

 

Bote

 

 

From: Richard Mace
Sent: Tuesday, 10 March, 2015 02:20
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh install on Debian

 

Hi,

Is there an fs_cli command that I can use that will get me round the current bug?

Its strange as it's only happened within the last month, as I built a system recently that worked fine out of the box.

 

Thanks

 

Richard

 

On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:

Hi Brian,

Removed the line, and rebooted, but still getting:

root at FreeSWITCH:~# fs_cli
[ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]

Richard

 

On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:

remove  

::1     localhost ip6-localhost ip6-loopback

 

from /etc/hosts

 

its a bug in debian.

 

On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com> wrote:

Hi,

 

Sorry, I should have clarified that this is running locally on the machine running FreeSWITCH.

 

Richard

 

On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:

On a fresh FS installation fs_cli only connects to 127.0.01 localhost. 

 

To connect from a remote machine put a valid routable interface address (although I have 0.0.0.0 in mine) in 

conf/autoload_configs/event_socket.conf.xml

 

and change the password and maybe even the port depending on the crackability of your network.

 

Then you?ll probably want to configure a profile configuration file with tight permissions to avoid having to type the parameters on the command line every time you start fs_cli.

 

Check the ?command-line Interface fs_cli? Confluence page for all the details.

 

Bote

 

 

From: Richard Mace
Sent: Friday, 06 March, 2015 14:34
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on Debian

 

Hi All,

I did a fresh install of both Debian and FreeSWITCH today, following the article here:

https://freeswitch.org/confluence/display/FREESWITCH/Debian

 

However, after installation, fs_cli will not connect. Any ideas?

 

Thanks

 

Richard

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/49b1cf46/attachment.html 

From richard.mace at gmail.com  Tue Mar 10 11:16:03 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 10 Mar 2015 08:16:03 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
Message-ID: <CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>

Hi Bote,
Tried this as well, on the local machine:

root at FreeSWITCH:~# fs_cli 127.0.0.1
[ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]

root at FreeSWITCH:~# /etc/init.d/freeswitch status
[ ok ] freeswitch is running.

Richard


On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:

> You could explicitly direct fs_cli to a particular i.p. address either on
> the command line or using a profile definition.
>
>
>
> Bote
>
>
>
>
>
> *From:* Richard Mace
> *Sent:* Tuesday, 10 March, 2015 02:20
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] fs_cli will not connect on Fresh
> install on Debian
>
>
>
> Hi,
>
> Is there an fs_cli command that I can use that will get me round the
> current bug?
>
> Its strange as it's only happened within the last month, as I built a
> system recently that worked fine out of the box.
>
>
>
> Thanks
>
>
>
> Richard
>
>
>
> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
>
> Hi Brian,
>
> Removed the line, and rebooted, but still getting:
>
> root at FreeSWITCH:~# fs_cli
> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>
> Richard
>
>
>
> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>
> remove
>
> ::1     localhost ip6-localhost ip6-loopback
>
>
>
> from /etc/hosts
>
>
>
> its a bug in debian.
>
>
>
> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
> wrote:
>
> Hi,
>
>
>
> Sorry, I should have clarified that this is running locally on the machine
> running FreeSWITCH.
>
>
>
> Richard
>
>
>
> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>
> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>
>
>
> To connect from a remote machine put a valid routable interface address
> (although I have 0.0.0.0 in mine) in
>
> conf/autoload_configs/event_socket.conf.xml
>
>
>
> and change the password and maybe even the port depending on the
> crackability of your network.
>
>
>
> Then you?ll probably want to configure a profile configuration file with
> tight permissions to avoid having to type the parameters on the command
> line every time you start fs_cli.
>
>
>
> Check the ?command-line Interface fs_cli? Confluence page for all the
> details.
>
>
>
> Bote
>
>
>
>
>
> *From:* Richard Mace
> *Sent:* Friday, 06 March, 2015 14:34
> *Subject:* [Freeswitch-users] fs_cli will not connect on Fresh install on
> Debian
>
>
>
> Hi All,
>
> I did a fresh install of both Debian and FreeSWITCH today, following the
> article here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>
>
>
> However, after installation, fs_cli will not connect. Any ideas?
>
>
>
> Thanks
>
>
>
> Richard
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/06ce7b93/attachment-0001.html 

From sukithaj at gmail.com  Tue Mar 10 11:39:39 2015
From: sukithaj at gmail.com (sukitha jayasinghe)
Date: Tue, 10 Mar 2015 14:09:39 +0530
Subject: [Freeswitch-users] Bridge event socket channels with early media
Message-ID: <CAAwzf1nXR-JR25Jo1mKvy9H+LMNb9OK9rc9SWribEaOj7VHNwg@mail.gmail.com>

I have developed a event socket application to handle my business logic
with greater channel control. When internal user make outbound call it
calls socket application and connect with the server, then server make
outbound call using xml_rpc api with socket application parameters like
["&socket({0}:{1} async full)"] and that channel also connect with the
server.I want to bridge channels before answer in-order to A-leg to listen
telco messages such as "user not in service area". What is the proper
mechanism for achive this.

Regards,
Sukitha
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/de4ad637/attachment.html 

From ssinyagin at gmail.com  Tue Mar 10 12:04:56 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 10:04:56 +0100
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CAPJLwtVkBwpuWX_CpKXUq9Ebbh8jY0qtbVhaHAufwibQeV_tLw@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
	<CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
	<CAPJLwtVkBwpuWX_CpKXUq9Ebbh8jY0qtbVhaHAufwibQeV_tLw@mail.gmail.com>
Message-ID: <CACOYK=gYH8EWBmv9HkYnfbQ1tj5OUjPvnALEMa_GsrghO7AT7w@mail.gmail.com>

there's also a nice tool called sngrep
https://github.com/irontec/sngrep
It needs a wide terminal with color support, like xterm.



On Tue, Mar 10, 2015 at 7:23 AM, Tito Cumpen <soulofmischief87 at gmail.com> wrote:
> If you have issues getting to the console. Use ngrep or sipgrep check this
> wiki out https://wiki.freeswitch.org/wiki/Packet_Capture.
>
> On Mar 10, 2015 2:21 AM, "Tito Cumpen" <soulofmischief87 at gmail.com> wrote:
>>
>> Richard,
>>
>> You may view registrations through the fs_cli console. You can get very
>> insightful debug ibformation through Sofia.check out Sofia debug
>> http://wiki.freeswitch.org/wiki/Sofia-SIP
>>
>> On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:
>>>
>>> Hi All,
>>> Is it possible to see when sip registrations happen in real time?
>>>
>>> Thanks
>>>
>>> Richard
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ali.jibran44 at gmail.com  Tue Mar 10 12:05:06 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Tue, 10 Mar 2015 14:05:06 +0500
Subject: [Freeswitch-users] Console Loglevel Query
Message-ID: <CAP9PqkT=cZqj1FSb8XNvcyBEeJAPDAqA_UQ4dcHtrDPspzWrHw@mail.gmail.com>

Newbie to freeswitch so I apologize if it sounds basic.

I was wondering if there was any way to access freeswitch console from
fs_cli?
I know fs_cli is a debug console for FS but is there any way I can access
the original freeswitch console?

I hope I make sense. Like I start freeswitch in background. Then I access
it through fs_cli. Can I access the background-ed FS?

Also if I can't, can I get the same loglevel in fs_cli as in FS?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/b91a002a/attachment.html 

From ssinyagin at gmail.com  Tue Mar 10 12:06:35 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 10:06:35 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
Message-ID: <CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>

this is exactly what I saw on a system which had no IPv4 address on
its ethernet ports. FreeSWITCH was just not listening to port 8021,
without any errors in the log.

On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com> wrote:
> Hi Bote,
> Tried this as well, on the local machine:
>
> root at FreeSWITCH:~# fs_cli 127.0.0.1
> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>
> root at FreeSWITCH:~# /etc/init.d/freeswitch status
> [ ok ] freeswitch is running.
>
> Richard
>
>
> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>>
>> You could explicitly direct fs_cli to a particular i.p. address either on
>> the command line or using a profile definition.
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>> From: Richard Mace
>> Sent: Tuesday, 10 March, 2015 02:20
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh install
>> on Debian
>>
>>
>>
>> Hi,
>>
>> Is there an fs_cli command that I can use that will get me round the
>> current bug?
>>
>> Its strange as it's only happened within the last month, as I built a
>> system recently that worked fine out of the box.
>>
>>
>>
>> Thanks
>>
>>
>>
>> Richard
>>
>>
>>
>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
>>
>> Hi Brian,
>>
>> Removed the line, and rebooted, but still getting:
>>
>> root at FreeSWITCH:~# fs_cli
>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>>
>> Richard
>>
>>
>>
>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>>
>> remove
>>
>> ::1     localhost ip6-localhost ip6-loopback
>>
>>
>>
>> from /etc/hosts
>>
>>
>>
>> its a bug in debian.
>>
>>
>>
>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>>
>> Hi,
>>
>>
>>
>> Sorry, I should have clarified that this is running locally on the machine
>> running FreeSWITCH.
>>
>>
>>
>> Richard
>>
>>
>>
>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>
>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>
>>
>>
>> To connect from a remote machine put a valid routable interface address
>> (although I have 0.0.0.0 in mine) in
>>
>> conf/autoload_configs/event_socket.conf.xml
>>
>>
>>
>> and change the password and maybe even the port depending on the
>> crackability of your network.
>>
>>
>>
>> Then you?ll probably want to configure a profile configuration file with
>> tight permissions to avoid having to type the parameters on the command line
>> every time you start fs_cli.
>>
>>
>>
>> Check the ?command-line Interface fs_cli? Confluence page for all the
>> details.
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>> From: Richard Mace
>> Sent: Friday, 06 March, 2015 14:34
>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
>> Debian
>>
>>
>>
>> Hi All,
>>
>> I did a fresh install of both Debian and FreeSWITCH today, following the
>> article here:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>
>>
>>
>> However, after installation, fs_cli will not connect. Any ideas?
>>
>>
>>
>> Thanks
>>
>>
>>
>> Richard
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Tue Mar 10 12:07:10 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 10:07:10 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
Message-ID: <CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>

Richard, what do you see in the output:

netstat -an | grep 8021



On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
<ssinyagin at gmail.com> wrote:
> this is exactly what I saw on a system which had no IPv4 address on
> its ethernet ports. FreeSWITCH was just not listening to port 8021,
> without any errors in the log.
>
> On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com> wrote:
>> Hi Bote,
>> Tried this as well, on the local machine:
>>
>> root at FreeSWITCH:~# fs_cli 127.0.0.1
>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>>
>> root at FreeSWITCH:~# /etc/init.d/freeswitch status
>> [ ok ] freeswitch is running.
>>
>> Richard
>>
>>
>> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>>>
>>> You could explicitly direct fs_cli to a particular i.p. address either on
>>> the command line or using a profile definition.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>> From: Richard Mace
>>> Sent: Tuesday, 10 March, 2015 02:20
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh install
>>> on Debian
>>>
>>>
>>>
>>> Hi,
>>>
>>> Is there an fs_cli command that I can use that will get me round the
>>> current bug?
>>>
>>> Its strange as it's only happened within the last month, as I built a
>>> system recently that worked fine out of the box.
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>> Richard
>>>
>>>
>>>
>>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
>>>
>>> Hi Brian,
>>>
>>> Removed the line, and rebooted, but still getting:
>>>
>>> root at FreeSWITCH:~# fs_cli
>>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>>>
>>> Richard
>>>
>>>
>>>
>>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>>>
>>> remove
>>>
>>> ::1     localhost ip6-localhost ip6-loopback
>>>
>>>
>>>
>>> from /etc/hosts
>>>
>>>
>>>
>>> its a bug in debian.
>>>
>>>
>>>
>>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>>> wrote:
>>>
>>> Hi,
>>>
>>>
>>>
>>> Sorry, I should have clarified that this is running locally on the machine
>>> running FreeSWITCH.
>>>
>>>
>>>
>>> Richard
>>>
>>>
>>>
>>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>>
>>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>>
>>>
>>>
>>> To connect from a remote machine put a valid routable interface address
>>> (although I have 0.0.0.0 in mine) in
>>>
>>> conf/autoload_configs/event_socket.conf.xml
>>>
>>>
>>>
>>> and change the password and maybe even the port depending on the
>>> crackability of your network.
>>>
>>>
>>>
>>> Then you?ll probably want to configure a profile configuration file with
>>> tight permissions to avoid having to type the parameters on the command line
>>> every time you start fs_cli.
>>>
>>>
>>>
>>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>> details.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>> From: Richard Mace
>>> Sent: Friday, 06 March, 2015 14:34
>>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
>>> Debian
>>>
>>>
>>>
>>> Hi All,
>>>
>>> I did a fresh install of both Debian and FreeSWITCH today, following the
>>> article here:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>>
>>>
>>>
>>> However, after installation, fs_cli will not connect. Any ideas?
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>> Richard
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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  Tue Mar 10 12:24:26 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 10 Mar 2015 09:24:26 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
Message-ID: <CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>

Nothing, by the looks of things

root at FreeSWITCH:~# netstat -an | grep 8021
root at FreeSWITCH:~#

Thanks


On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> Richard, what do you see in the output:
>
> netstat -an | grep 8021
>
>
>
> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
> <ssinyagin at gmail.com> wrote:
> > this is exactly what I saw on a system which had no IPv4 address on
> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
> > without any errors in the log.
> >
> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com>
> wrote:
> >> Hi Bote,
> >> Tried this as well, on the local machine:
> >>
> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
> >>
> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
> >> [ ok ] freeswitch is running.
> >>
> >> Richard
> >>
> >>
> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
> >>>
> >>> You could explicitly direct fs_cli to a particular i.p. address either
> on
> >>> the command line or using a profile definition.
> >>>
> >>>
> >>>
> >>> Bote
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> From: Richard Mace
> >>> Sent: Tuesday, 10 March, 2015 02:20
> >>> To: FreeSWITCH Users Help
> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
> install
> >>> on Debian
> >>>
> >>>
> >>>
> >>> Hi,
> >>>
> >>> Is there an fs_cli command that I can use that will get me round the
> >>> current bug?
> >>>
> >>> Its strange as it's only happened within the last month, as I built a
> >>> system recently that worked fine out of the box.
> >>>
> >>>
> >>>
> >>> Thanks
> >>>
> >>>
> >>>
> >>> Richard
> >>>
> >>>
> >>>
> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
> >>>
> >>> Hi Brian,
> >>>
> >>> Removed the line, and rebooted, but still getting:
> >>>
> >>> root at FreeSWITCH:~# fs_cli
> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
> >>>
> >>> Richard
> >>>
> >>>
> >>>
> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
> >>>
> >>> remove
> >>>
> >>> ::1     localhost ip6-localhost ip6-loopback
> >>>
> >>>
> >>>
> >>> from /etc/hosts
> >>>
> >>>
> >>>
> >>> its a bug in debian.
> >>>
> >>>
> >>>
> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
> >>> wrote:
> >>>
> >>> Hi,
> >>>
> >>>
> >>>
> >>> Sorry, I should have clarified that this is running locally on the
> machine
> >>> running FreeSWITCH.
> >>>
> >>>
> >>>
> >>> Richard
> >>>
> >>>
> >>>
> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
> >>>
> >>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
> >>>
> >>>
> >>>
> >>> To connect from a remote machine put a valid routable interface address
> >>> (although I have 0.0.0.0 in mine) in
> >>>
> >>> conf/autoload_configs/event_socket.conf.xml
> >>>
> >>>
> >>>
> >>> and change the password and maybe even the port depending on the
> >>> crackability of your network.
> >>>
> >>>
> >>>
> >>> Then you?ll probably want to configure a profile configuration file
> with
> >>> tight permissions to avoid having to type the parameters on the
> command line
> >>> every time you start fs_cli.
> >>>
> >>>
> >>>
> >>> Check the ?command-line Interface fs_cli? Confluence page for all the
> >>> details.
> >>>
> >>>
> >>>
> >>> Bote
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> From: Richard Mace
> >>> Sent: Friday, 06 March, 2015 14:34
> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
> >>> Debian
> >>>
> >>>
> >>>
> >>> Hi All,
> >>>
> >>> I did a fresh install of both Debian and FreeSWITCH today, following
> the
> >>> article here:
> >>>
> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
> >>>
> >>>
> >>>
> >>> However, after installation, fs_cli will not connect. Any ideas?
> >>>
> >>>
> >>>
> >>> Thanks
> >>>
> >>>
> >>>
> >>> Richard
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/6fe808dc/attachment-0001.html 

From ssinyagin at gmail.com  Tue Mar 10 12:29:25 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 10:29:25 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
	<CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
Message-ID: <CACOYK=jnNsNYCjSNgt=euzxpnGhJCUNZCWskXGBUzsyxZNChzw@mail.gmail.com>

alright, I'll file a Jira ticket

On Tue, Mar 10, 2015 at 10:24 AM, Richard Mace <richard.mace at gmail.com> wrote:
> Nothing, by the looks of things
>
> root at FreeSWITCH:~# netstat -an | grep 8021
> root at FreeSWITCH:~#
>
> Thanks
>
>
> On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>
>> Richard, what do you see in the output:
>>
>> netstat -an | grep 8021
>>
>>
>>
>> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
>> <ssinyagin at gmail.com> wrote:
>> > this is exactly what I saw on a system which had no IPv4 address on
>> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
>> > without any errors in the log.
>> >
>> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com>
>> > wrote:
>> >> Hi Bote,
>> >> Tried this as well, on the local machine:
>> >>
>> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
>> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>> >>
>> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
>> >> [ ok ] freeswitch is running.
>> >>
>> >> Richard
>> >>
>> >>
>> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>> >>>
>> >>> You could explicitly direct fs_cli to a particular i.p. address either
>> >>> on
>> >>> the command line or using a profile definition.
>> >>>
>> >>>
>> >>>
>> >>> Bote
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> From: Richard Mace
>> >>> Sent: Tuesday, 10 March, 2015 02:20
>> >>> To: FreeSWITCH Users Help
>> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
>> >>> install
>> >>> on Debian
>> >>>
>> >>>
>> >>>
>> >>> Hi,
>> >>>
>> >>> Is there an fs_cli command that I can use that will get me round the
>> >>> current bug?
>> >>>
>> >>> Its strange as it's only happened within the last month, as I built a
>> >>> system recently that worked fine out of the box.
>> >>>
>> >>>
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
>> >>>
>> >>> Hi Brian,
>> >>>
>> >>> Removed the line, and rebooted, but still getting:
>> >>>
>> >>> root at FreeSWITCH:~# fs_cli
>> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
>> >>> Error]
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>> >>>
>> >>> remove
>> >>>
>> >>> ::1     localhost ip6-localhost ip6-loopback
>> >>>
>> >>>
>> >>>
>> >>> from /etc/hosts
>> >>>
>> >>>
>> >>>
>> >>> its a bug in debian.
>> >>>
>> >>>
>> >>>
>> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>> >>> wrote:
>> >>>
>> >>> Hi,
>> >>>
>> >>>
>> >>>
>> >>> Sorry, I should have clarified that this is running locally on the
>> >>> machine
>> >>> running FreeSWITCH.
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>> >>>
>> >>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>> >>>
>> >>>
>> >>>
>> >>> To connect from a remote machine put a valid routable interface
>> >>> address
>> >>> (although I have 0.0.0.0 in mine) in
>> >>>
>> >>> conf/autoload_configs/event_socket.conf.xml
>> >>>
>> >>>
>> >>>
>> >>> and change the password and maybe even the port depending on the
>> >>> crackability of your network.
>> >>>
>> >>>
>> >>>
>> >>> Then you?ll probably want to configure a profile configuration file
>> >>> with
>> >>> tight permissions to avoid having to type the parameters on the
>> >>> command line
>> >>> every time you start fs_cli.
>> >>>
>> >>>
>> >>>
>> >>> Check the ?command-line Interface fs_cli? Confluence page for all the
>> >>> details.
>> >>>
>> >>>
>> >>>
>> >>> Bote
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> From: Richard Mace
>> >>> Sent: Friday, 06 March, 2015 14:34
>> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
>> >>> on
>> >>> Debian
>> >>>
>> >>>
>> >>>
>> >>> Hi All,
>> >>>
>> >>> I did a fresh install of both Debian and FreeSWITCH today, following
>> >>> the
>> >>> article here:
>> >>>
>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>> >>>
>> >>>
>> >>>
>> >>> However, after installation, fs_cli will not connect. Any ideas?
>> >>>
>> >>>
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://confluence.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Tue Mar 10 12:30:33 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 10:30:33 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CACOYK=jnNsNYCjSNgt=euzxpnGhJCUNZCWskXGBUzsyxZNChzw@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
	<CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
	<CACOYK=jnNsNYCjSNgt=euzxpnGhJCUNZCWskXGBUzsyxZNChzw@mail.gmail.com>
Message-ID: <CACOYK=haWkgJ4kAjoc03vcHhpytN9-YaAHeRDbRACo3MAA0sDQ@mail.gmail.com>

in the mean time, you can attach the socket initerface to the IPv6
loopback like I mentioned earlier. That works, although a bit
inconvenient

On Tue, Mar 10, 2015 at 10:29 AM, Stanislav Sinyagin
<ssinyagin at gmail.com> wrote:
> alright, I'll file a Jira ticket
>
> On Tue, Mar 10, 2015 at 10:24 AM, Richard Mace <richard.mace at gmail.com> wrote:
>> Nothing, by the looks of things
>>
>> root at FreeSWITCH:~# netstat -an | grep 8021
>> root at FreeSWITCH:~#
>>
>> Thanks
>>
>>
>> On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>>
>>> Richard, what do you see in the output:
>>>
>>> netstat -an | grep 8021
>>>
>>>
>>>
>>> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
>>> <ssinyagin at gmail.com> wrote:
>>> > this is exactly what I saw on a system which had no IPv4 address on
>>> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
>>> > without any errors in the log.
>>> >
>>> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com>
>>> > wrote:
>>> >> Hi Bote,
>>> >> Tried this as well, on the local machine:
>>> >>
>>> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
>>> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>>> >>
>>> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
>>> >> [ ok ] freeswitch is running.
>>> >>
>>> >> Richard
>>> >>
>>> >>
>>> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>>> >>>
>>> >>> You could explicitly direct fs_cli to a particular i.p. address either
>>> >>> on
>>> >>> the command line or using a profile definition.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Bote
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> From: Richard Mace
>>> >>> Sent: Tuesday, 10 March, 2015 02:20
>>> >>> To: FreeSWITCH Users Help
>>> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
>>> >>> install
>>> >>> on Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> Hi,
>>> >>>
>>> >>> Is there an fs_cli command that I can use that will get me round the
>>> >>> current bug?
>>> >>>
>>> >>> Its strange as it's only happened within the last month, as I built a
>>> >>> system recently that worked fine out of the box.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com> wrote:
>>> >>>
>>> >>> Hi Brian,
>>> >>>
>>> >>> Removed the line, and rebooted, but still getting:
>>> >>>
>>> >>> root at FreeSWITCH:~# fs_cli
>>> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
>>> >>> Error]
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>>> >>>
>>> >>> remove
>>> >>>
>>> >>> ::1     localhost ip6-localhost ip6-loopback
>>> >>>
>>> >>>
>>> >>>
>>> >>> from /etc/hosts
>>> >>>
>>> >>>
>>> >>>
>>> >>> its a bug in debian.
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>>> >>> wrote:
>>> >>>
>>> >>> Hi,
>>> >>>
>>> >>>
>>> >>>
>>> >>> Sorry, I should have clarified that this is running locally on the
>>> >>> machine
>>> >>> running FreeSWITCH.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>> >>>
>>> >>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>>> >>>
>>> >>>
>>> >>>
>>> >>> To connect from a remote machine put a valid routable interface
>>> >>> address
>>> >>> (although I have 0.0.0.0 in mine) in
>>> >>>
>>> >>> conf/autoload_configs/event_socket.conf.xml
>>> >>>
>>> >>>
>>> >>>
>>> >>> and change the password and maybe even the port depending on the
>>> >>> crackability of your network.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Then you?ll probably want to configure a profile configuration file
>>> >>> with
>>> >>> tight permissions to avoid having to type the parameters on the
>>> >>> command line
>>> >>> every time you start fs_cli.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>> >>> details.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Bote
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> From: Richard Mace
>>> >>> Sent: Friday, 06 March, 2015 14:34
>>> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
>>> >>> on
>>> >>> Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> Hi All,
>>> >>>
>>> >>> I did a fresh install of both Debian and FreeSWITCH today, following
>>> >>> the
>>> >>> article here:
>>> >>>
>>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> However, after installation, fs_cli will not connect. Any ideas?
>>> >>>
>>> >>>
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://confluence.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From zvi at lexifone.com  Tue Mar 10 13:02:33 2015
From: zvi at lexifone.com (Zvi Agmon)
Date: Tue, 10 Mar 2015 12:02:33 +0200
Subject: [Freeswitch-users] Call codec sample rate and call recording sample
	rate relation
Message-ID: <CAKYJ0qZA89PZyDdp_pX-K5C1Z2=U8jgi7FKt5F0p0QPfGdJbRg@mail.gmail.com>

Hello,

I'm trying to understand the relation between the the call codec and
recording format in terms of sample rate and audio quality.
The question is: when a call use a wide band codec (i.e. speex at 16kh) and
then saved with "record_sample_rate=16000" -  does the saved file keeps the
16kh audio quality?


Thanks a lot

Zvi Agmon
Lexifone <http://www.lexifone.com/>
email: zvi at lexifone.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/70e3b390/attachment.html 

From steveayre at gmail.com  Tue Mar 10 14:01:06 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 10 Mar 2015 11:01:06 +0000
Subject: [Freeswitch-users] Console Loglevel Query
In-Reply-To: <CAP9PqkT=cZqj1FSb8XNvcyBEeJAPDAqA_UQ4dcHtrDPspzWrHw@mail.gmail.com>
References: <CAP9PqkT=cZqj1FSb8XNvcyBEeJAPDAqA_UQ4dcHtrDPspzWrHw@mail.gmail.com>
Message-ID: <CAFiqYumvrhgcrk2c=r2B8aTt4C61=HzuTXdMF3QeRzZAfZaJ-A@mail.gmail.com>

Both consoles give you the same level as access.

Also if I can't, can I get the same loglevel in fs_cli as in FS?


Use the fs_cli /log command, eg "/log DEBUG"


On 10 March 2015 at 09:05, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> Newbie to freeswitch so I apologize if it sounds basic.
>
> I was wondering if there was any way to access freeswitch console from
> fs_cli?
> I know fs_cli is a debug console for FS but is there any way I can access
> the original freeswitch console?
>
> I hope I make sense. Like I start freeswitch in background. Then I access
> it through fs_cli. Can I access the background-ed FS?
>
> Also if I can't, can I get the same loglevel in fs_cli as in FS?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/eeb85cb0/attachment.html 

From steveayre at gmail.com  Tue Mar 10 14:04:54 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 10 Mar 2015 11:04:54 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
	<CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
Message-ID: <CAFiqYuk3O9PSj51kn8usZhXS3MyyJdi6pJ8+psbGx3Q94iVTwg@mail.gmail.com>

Are you sure it's trying to load mod_event_socket? Unless it loads that
module it won't listen on 8021.

If it is trying to, perhaps start freeswitch in the foreground (-c option)
and then (re)load the module once it's running. If there's an error that'll
hopefully show it up if it's not making it to the log file.

On 10 March 2015 at 09:24, Richard Mace <richard.mace at gmail.com> wrote:

> Nothing, by the looks of things
>
> root at FreeSWITCH:~# netstat -an | grep 8021
> root at FreeSWITCH:~#
>
> Thanks
>
>
> On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> Richard, what do you see in the output:
>>
>> netstat -an | grep 8021
>>
>>
>>
>> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
>> <ssinyagin at gmail.com> wrote:
>> > this is exactly what I saw on a system which had no IPv4 address on
>> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
>> > without any errors in the log.
>> >
>> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>> >> Hi Bote,
>> >> Tried this as well, on the local machine:
>> >>
>> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
>> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection Error]
>> >>
>> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
>> >> [ ok ] freeswitch is running.
>> >>
>> >> Richard
>> >>
>> >>
>> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>> >>>
>> >>> You could explicitly direct fs_cli to a particular i.p. address
>> either on
>> >>> the command line or using a profile definition.
>> >>>
>> >>>
>> >>>
>> >>> Bote
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> From: Richard Mace
>> >>> Sent: Tuesday, 10 March, 2015 02:20
>> >>> To: FreeSWITCH Users Help
>> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
>> install
>> >>> on Debian
>> >>>
>> >>>
>> >>>
>> >>> Hi,
>> >>>
>> >>> Is there an fs_cli command that I can use that will get me round the
>> >>> current bug?
>> >>>
>> >>> Its strange as it's only happened within the last month, as I built a
>> >>> system recently that worked fine out of the box.
>> >>>
>> >>>
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com>
>> wrote:
>> >>>
>> >>> Hi Brian,
>> >>>
>> >>> Removed the line, and rebooted, but still getting:
>> >>>
>> >>> root at FreeSWITCH:~# fs_cli
>> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
>> Error]
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>> >>>
>> >>> remove
>> >>>
>> >>> ::1     localhost ip6-localhost ip6-loopback
>> >>>
>> >>>
>> >>>
>> >>> from /etc/hosts
>> >>>
>> >>>
>> >>>
>> >>> its a bug in debian.
>> >>>
>> >>>
>> >>>
>> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>> >>> wrote:
>> >>>
>> >>> Hi,
>> >>>
>> >>>
>> >>>
>> >>> Sorry, I should have clarified that this is running locally on the
>> machine
>> >>> running FreeSWITCH.
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>> >>>
>> >>> On a fresh FS installation fs_cli only connects to 127.0.01 localhost.
>> >>>
>> >>>
>> >>>
>> >>> To connect from a remote machine put a valid routable interface
>> address
>> >>> (although I have 0.0.0.0 in mine) in
>> >>>
>> >>> conf/autoload_configs/event_socket.conf.xml
>> >>>
>> >>>
>> >>>
>> >>> and change the password and maybe even the port depending on the
>> >>> crackability of your network.
>> >>>
>> >>>
>> >>>
>> >>> Then you?ll probably want to configure a profile configuration file
>> with
>> >>> tight permissions to avoid having to type the parameters on the
>> command line
>> >>> every time you start fs_cli.
>> >>>
>> >>>
>> >>>
>> >>> Check the ?command-line Interface fs_cli? Confluence page for all the
>> >>> details.
>> >>>
>> >>>
>> >>>
>> >>> Bote
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> From: Richard Mace
>> >>> Sent: Friday, 06 March, 2015 14:34
>> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
>> on
>> >>> Debian
>> >>>
>> >>>
>> >>>
>> >>> Hi All,
>> >>>
>> >>> I did a fresh install of both Debian and FreeSWITCH today, following
>> the
>> >>> article here:
>> >>>
>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>> >>>
>> >>>
>> >>>
>> >>> However, after installation, fs_cli will not connect. Any ideas?
>> >>>
>> >>>
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>>
>> >>> Richard
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://confluence.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/039acb46/attachment-0001.html 

From steveayre at gmail.com  Tue Mar 10 14:06:01 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 10 Mar 2015 11:06:01 +0000
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
	<CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
Message-ID: <CAFiqYunv72XARWoUCPDR_DV7uRTLptuf71RxX5EoTO=NL-emew@mail.gmail.com>

There are also ESL events you can subscribe to
http://wiki.freeswitch.org/wiki/Sofia-SIP#Custom_Events

On 10 March 2015 at 06:21, Tito Cumpen <soulofmischief87 at gmail.com> wrote:

> Richard,
>
> You may view registrations through the fs_cli console. You can get very
> insightful debug ibformation through Sofia.check out Sofia debug
> http://wiki.freeswitch.org/wiki/Sofia-SIP
> On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:
>
>> Hi All,
>> Is it possible to see when sip registrations happen in real time?
>>
>> Thanks
>>
>> Richard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/9eeffd57/attachment.html 

From ssinyagin at gmail.com  Tue Mar 10 14:19:59 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 12:19:59 +0100
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CAFiqYuk3O9PSj51kn8usZhXS3MyyJdi6pJ8+psbGx3Q94iVTwg@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
	<CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
	<CAFiqYuk3O9PSj51kn8usZhXS3MyyJdi6pJ8+psbGx3Q94iVTwg@mail.gmail.com>
Message-ID: <CACOYK=hNYnYNj965sRvbf4e554bY-V-ZJfRvouGpSw1Y1AKcJQ@mail.gmail.com>

I submitted https://freeswitch.org/jira/browse/FS-7358

mod_event_socket is loaded, and then
2015-03-10 11:29:48.367067 [NOTICE] switch_loadable_module.c:102
Thread ended for mod_event_socket



On Tue, Mar 10, 2015 at 12:04 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Are you sure it's trying to load mod_event_socket? Unless it loads that
> module it won't listen on 8021.
>
> If it is trying to, perhaps start freeswitch in the foreground (-c option)
> and then (re)load the module once it's running. If there's an error that'll
> hopefully show it up if it's not making it to the log file.
>
> On 10 March 2015 at 09:24, Richard Mace <richard.mace at gmail.com> wrote:
>>
>> Nothing, by the looks of things
>>
>> root at FreeSWITCH:~# netstat -an | grep 8021
>> root at FreeSWITCH:~#
>>
>> Thanks
>>
>>
>> On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>>
>>> Richard, what do you see in the output:
>>>
>>> netstat -an | grep 8021
>>>
>>>
>>>
>>> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
>>> <ssinyagin at gmail.com> wrote:
>>> > this is exactly what I saw on a system which had no IPv4 address on
>>> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
>>> > without any errors in the log.
>>> >
>>> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <richard.mace at gmail.com>
>>> > wrote:
>>> >> Hi Bote,
>>> >> Tried this as well, on the local machine:
>>> >>
>>> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
>>> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
>>> >> Error]
>>> >>
>>> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
>>> >> [ ok ] freeswitch is running.
>>> >>
>>> >> Richard
>>> >>
>>> >>
>>> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com> wrote:
>>> >>>
>>> >>> You could explicitly direct fs_cli to a particular i.p. address
>>> >>> either on
>>> >>> the command line or using a profile definition.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Bote
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> From: Richard Mace
>>> >>> Sent: Tuesday, 10 March, 2015 02:20
>>> >>> To: FreeSWITCH Users Help
>>> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
>>> >>> install
>>> >>> on Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> Hi,
>>> >>>
>>> >>> Is there an fs_cli command that I can use that will get me round the
>>> >>> current bug?
>>> >>>
>>> >>> Its strange as it's only happened within the last month, as I built a
>>> >>> system recently that worked fine out of the box.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com>
>>> >>> wrote:
>>> >>>
>>> >>> Hi Brian,
>>> >>>
>>> >>> Removed the line, and rebooted, but still getting:
>>> >>>
>>> >>> root at FreeSWITCH:~# fs_cli
>>> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
>>> >>> Error]
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
>>> >>>
>>> >>> remove
>>> >>>
>>> >>> ::1     localhost ip6-localhost ip6-loopback
>>> >>>
>>> >>>
>>> >>>
>>> >>> from /etc/hosts
>>> >>>
>>> >>>
>>> >>>
>>> >>> its a bug in debian.
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <richard.mace at gmail.com>
>>> >>> wrote:
>>> >>>
>>> >>> Hi,
>>> >>>
>>> >>>
>>> >>>
>>> >>> Sorry, I should have clarified that this is running locally on the
>>> >>> machine
>>> >>> running FreeSWITCH.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com> wrote:
>>> >>>
>>> >>> On a fresh FS installation fs_cli only connects to 127.0.01
>>> >>> localhost.
>>> >>>
>>> >>>
>>> >>>
>>> >>> To connect from a remote machine put a valid routable interface
>>> >>> address
>>> >>> (although I have 0.0.0.0 in mine) in
>>> >>>
>>> >>> conf/autoload_configs/event_socket.conf.xml
>>> >>>
>>> >>>
>>> >>>
>>> >>> and change the password and maybe even the port depending on the
>>> >>> crackability of your network.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Then you?ll probably want to configure a profile configuration file
>>> >>> with
>>> >>> tight permissions to avoid having to type the parameters on the
>>> >>> command line
>>> >>> every time you start fs_cli.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Check the ?command-line Interface fs_cli? Confluence page for all the
>>> >>> details.
>>> >>>
>>> >>>
>>> >>>
>>> >>> Bote
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> From: Richard Mace
>>> >>> Sent: Friday, 06 March, 2015 14:34
>>> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh install
>>> >>> on
>>> >>> Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> Hi All,
>>> >>>
>>> >>> I did a fresh install of both Debian and FreeSWITCH today, following
>>> >>> the
>>> >>> article here:
>>> >>>
>>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
>>> >>>
>>> >>>
>>> >>>
>>> >>> However, after installation, fs_cli will not connect. Any ideas?
>>> >>>
>>> >>>
>>> >>>
>>> >>> Thanks
>>> >>>
>>> >>>
>>> >>>
>>> >>> Richard
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://confluence.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ben at langfeld.co.uk  Tue Mar 10 14:33:01 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Tue, 10 Mar 2015 08:33:01 -0300
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <D123B137.9810D%krice@freeswitch.org>
References: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>
	<D123B137.9810D%krice@freeswitch.org>
Message-ID: <CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>

git fsck complains about it.

? git fsck
Checking object directories: 100% (256/256), done.
error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
expected 'committer' line
error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
expected 'committer' line
Checking objects: 100% (294760/294760), done.
dangling commit f62e8f0e250b35bbadb7367082127304f3e72e74
dangling commit dc844f983605a015bf600232864122b94a278e76

On 9 March 2015 at 23:21, Ken Rice <krice at freeswitch.org> wrote:

>  Git itself doesn?t say this is an invalid commit (we use stash/git and
> its perfectly fine with this)... There is a git-lint process that github
> uses and it rejects this commit
>
>
> On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>
> I understand very well why rewriting history is undesirable and how git
> works. What I wonder is what process was used to convince git to create a
> commit which it would later say is invalid. As I said, I'm curious.
>
> On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:
>
> Plus it's rather annoying to do so (rewrite history). The identifier of
> each commit is a hash computed from the content of the commit plus the
> metadata which includes the authors. Changing the author would change the
> identifier of the commit. That then changes the identifier of every commit
> afterwards. That then breaks every checkout / fork based off the tree as
> they no longer know where they are forked from. And since identifiers have
> all been rewritten we would no longer know what version you were running,
> or what version bug reports were reported against.
>
> (that's why git makes it easy to amend your latest uncommitted commit
> message but rather difficult to edit any others)
>
>
>
>
> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>
> We are not rewriting history to fix this so it doesn't really matter who
> is right or wrong.
>
> Mike
>
> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
> I'm curious about this one... If git fsck complains about the issue, what
> is the justification for saying that Github is broken? How were these
> commits created with a format that git itself complains about?
>
> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>
> I switched to bitbucket.org <http://bitbucket.org/>  just for the
> FreeSWITCH repo to work around this.
>
>
> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>
> This is a known issue with github and will not be fixed
>
>
>
> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com <
> http://aronp at guaranteedplus.com/> > wrote:
>
> Hi,
>
> I'm trying to push the freeswitch git repo to my github, but I get the
> following error
>
>
> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
> format - expected 'committer' line
> remote: fatal: Error in object
> channel_by_id: 0: bad id: channel free
> Received window adjust for non-open channel 0.
> error: pack-objects died of signal 13
>
> This is caused by having multiple authors on a commit (which in general is
> not allowed by git) and github verifies the commits and rejects it.
>
> here is the output of git fsck
>
> Checking object directories: 100% (256/256), done.
> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
> expected 'committer' line
> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
> expected 'committer' line
> Checking objects: 100% (254227/254227), done.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/bd2b587c/attachment-0001.html 

From dp.siddharth at eng.knowlarity.com  Tue Mar 10 16:20:10 2015
From: dp.siddharth at eng.knowlarity.com (DP Siddharth)
Date: Tue, 10 Mar 2015 18:50:10 +0530
Subject: [Freeswitch-users] ESL object taking large memory
Message-ID: <CAFDf6-9Rn1ro4yYhhEMxGH5jeYv4zsyRS2Yw8KhaOTiYKSa4tg@mail.gmail.com>

Hi All,

I am working on python/esl based server. We are seeing memory getting
increase by ~6.5MB when con = ESLconnection() get called.

We tried to cleanup this object as soon calls gets complete, but somehow we
are not successful.

Further looking into esl/src/include/esl.h we found

#define BUF_CHUNK 65536 * 50
#define BUF_START 65536 * 100

modifying these values help in controlling python server memory growth.

Can someone help me in understanding why we have this buffer size of 6.5MB?


-- 
Thanks & Regards,
D P Siddharth
Director (Platform)
Knowlarity Communications
Ph: +919999115231
dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>

*"Come together to build a lasting world-class cloud telephony company that
helps businesses grow"*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/35392689/attachment.html 

From steveayre at gmail.com  Tue Mar 10 16:56:41 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 10 Mar 2015 13:56:41 +0000
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>
References: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>
	<D123B137.9810D%krice@freeswitch.org>
	<CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>
Message-ID: <CAFiqYumb2aFP27eHPhkNFqKCdg=mSbNzxxNYu=hW0cSrSjgiuA@mail.gmail.com>

I suspect it's performing some sanity checks that git commit does not.
Which you could then argue is either a bug or feature in git.



On 10 March 2015 at 11:33, Ben Langfeld <ben at langfeld.co.uk> wrote:

> git fsck complains about it.
>
> ? git fsck
> Checking object directories: 100% (256/256), done.
> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format -
> expected 'committer' line
> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format -
> expected 'committer' line
> Checking objects: 100% (294760/294760), done.
> dangling commit f62e8f0e250b35bbadb7367082127304f3e72e74
> dangling commit dc844f983605a015bf600232864122b94a278e76
>
> On 9 March 2015 at 23:21, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Git itself doesn?t say this is an invalid commit (we use stash/git and
>> its perfectly fine with this)... There is a git-lint process that github
>> uses and it rejects this commit
>>
>>
>> On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>
>> I understand very well why rewriting history is undesirable and how git
>> works. What I wonder is what process was used to convince git to create a
>> commit which it would later say is invalid. As I said, I'm curious.
>>
>> On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Plus it's rather annoying to do so (rewrite history). The identifier of
>> each commit is a hash computed from the content of the commit plus the
>> metadata which includes the authors. Changing the author would change the
>> identifier of the commit. That then changes the identifier of every commit
>> afterwards. That then breaks every checkout / fork based off the tree as
>> they no longer know where they are forked from. And since identifiers have
>> all been rewritten we would no longer know what version you were running,
>> or what version bug reports were reported against.
>>
>> (that's why git makes it easy to amend your latest uncommitted commit
>> message but rather difficult to edit any others)
>>
>>
>>
>>
>> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>>
>> We are not rewriting history to fix this so it doesn't really matter who
>> is right or wrong.
>>
>> Mike
>>
>> On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>> I'm curious about this one... If git fsck complains about the issue, what
>> is the justification for saying that Github is broken? How were these
>> commits created with a format that git itself complains about?
>>
>> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>>
>> I switched to bitbucket.org <http://bitbucket.org/>  just for the
>> FreeSWITCH repo to work around this.
>>
>>
>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>
>> This is a known issue with github and will not be fixed
>>
>>
>>
>> On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com <
>> http://aronp at guaranteedplus.com/> > wrote:
>>
>> Hi,
>>
>> I'm trying to push the freeswitch git repo to my github, but I get the
>> following error
>>
>>
>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>> format - expected 'committer' line
>> remote: fatal: Error in object
>> channel_by_id: 0: bad id: channel free
>> Received window adjust for non-open channel 0.
>> error: pack-objects died of signal 13
>>
>> This is caused by having multiple authors on a commit (which in general
>> is not allowed by git) and github verifies the commits and rejects it.
>>
>> here is the output of git fsck
>>
>> Checking object directories: 100% (256/256), done.
>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>> - expected 'committer' line
>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>> - expected 'committer' line
>> Checking objects: 100% (254227/254227), done.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>> Twitter: @FreeSWITCH
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/362f8bbd/attachment.html 

From victor.chukalovskiy at gmail.com  Tue Mar 10 17:06:39 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Tue, 10 Mar 2015 10:06:39 -0400
Subject: [Freeswitch-users] What exactly enable_soa = false changes?
In-Reply-To: <CAFiqYumhpce6YVE7s4OU67BUtbq-pKrp=cw46FNRAK4raOdTfQ@mail.gmail.com>
References: <54FDCA85.3060508@gmail.com>
	<CAFiqYumhpce6YVE7s4OU67BUtbq-pKrp=cw46FNRAK4raOdTfQ@mail.gmail.com>
Message-ID: <54FEFA6F.7010506@gmail.com>

Good day Steven, thanks for this high-level debrief.

Do I understand right that soa portion of the stack was in use by FS for 
a long time now?
And default is enable_soa= true and has been like that for 2-3 years at 
least?

Thank you!

On 15-03-09 07:15 PM, Steven Ayre wrote:
> The high level answer is it controls whether the soa portion of the 
> sofia stack is used (http://sofia-sip.sourceforge.net/refdocs/soa/), 
> disabling it would use an alternative implementation. Obviously there 
> are differences between the implementations from what you've observed. 
> I can't tell you exactly what though.
>
> On 9 March 2015 at 16:29, Victor Chukalovskiy 
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>> 
> wrote:
>
>     Good day,
>
>     I find documentation regarding enable_soa param very scarce, so
>     looking
>     for some clarifications here.
>
>     What exactly changes in FS behavior with enable_soa = false?
>
>     For example, I observe that with soa disabled, FS receiving "183" with
>     SDP on one leg substitutes it with plain "180 Ringing" on another leg.
>     This is not correct, but I'm not sure if it's a bug or an expected
>     outcome of disabling soa.
>
>     Thanks!!
>     -Victor
>
>     _________________________________________________________________________
>     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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/610359dd/attachment-0001.html 

From anuraagmishra92 at gmail.com  Tue Mar 10 15:38:52 2015
From: anuraagmishra92 at gmail.com (Anurag Mishra)
Date: Tue, 10 Mar 2015 18:08:52 +0530
Subject: [Freeswitch-users] G729 Codec is not loading
Message-ID: <CACGrPnhCjZ35mqONcAGYo8WwntLUTTWbEREh=BufE0UinLMEjw@mail.gmail.com>

Hi All,

Few days back, I have installed G729 Codec that time everything was fine.
But suddenly its stopped working, I was tried to load it again from CLI but
nothing happened, its shows just a prompt without any error.

Please advise.

Thanks,
Anurag Mishra
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/c6584917/attachment.html 

From brian at freeswitch.org  Tue Mar 10 17:57:19 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 09:57:19 -0500
Subject: [Freeswitch-users] Call codec sample rate and call recording
 sample rate relation
In-Reply-To: <CAKYJ0qZA89PZyDdp_pX-K5C1Z2=U8jgi7FKt5F0p0QPfGdJbRg@mail.gmail.com>
References: <CAKYJ0qZA89PZyDdp_pX-K5C1Z2=U8jgi7FKt5F0p0QPfGdJbRg@mail.gmail.com>
Message-ID: <CAEJMVkA+0R1Y5TVrQQAqMYt7pB1sLy2Yqohs8ryA_TZDL61jFg@mail.gmail.com>

By default the channels sample rate is at is what we will record at, there
is no reason to set record_sample_rate in those cases.

On Tue, Mar 10, 2015 at 5:02 AM, Zvi Agmon <zvi at lexifone.com> wrote:

> Hello,
>
> I'm trying to understand the relation between the the call codec and
> recording format in terms of sample rate and audio quality.
> The question is: when a call use a wide band codec (i.e. speex at 16kh) and
> then saved with "record_sample_rate=16000" -  does the saved file keeps
> the 16kh audio quality?
>
>
> Thanks a lot
>
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/abfbaa4c/attachment.html 

From brian at freeswitch.org  Tue Mar 10 18:09:55 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 10:09:55 -0500
Subject: [Freeswitch-users] G729 Codec is not loading
In-Reply-To: <CACGrPnhCjZ35mqONcAGYo8WwntLUTTWbEREh=BufE0UinLMEjw@mail.gmail.com>
References: <CACGrPnhCjZ35mqONcAGYo8WwntLUTTWbEREh=BufE0UinLMEjw@mail.gmail.com>
Message-ID: <CAEJMVkCLPzkX1YCy1fEV_L-opJ=eBkzuJwqJD_Zoza9dKrbS4w@mail.gmail.com>

Logs, what commands are you running or some more details about what you see
on your screen would be helpful.


On Tue, Mar 10, 2015 at 7:38 AM, Anurag Mishra <anuraagmishra92 at gmail.com>
wrote:

> Hi All,
>
> Few days back, I have installed G729 Codec that time everything was fine.
> But suddenly its stopped working, I was tried to load it again from CLI but
> nothing happened, its shows just a prompt without any error.
>
> Please advise.
>
> Thanks,
> Anurag Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/cf9a87a3/attachment.html 

From brian at freeswitch.org  Tue Mar 10 18:11:16 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 10:11:16 -0500
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CAFiqYunv72XARWoUCPDR_DV7uRTLptuf71RxX5EoTO=NL-emew@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>
	<CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>
	<CAFiqYunv72XARWoUCPDR_DV7uRTLptuf71RxX5EoTO=NL-emew@mail.gmail.com>
Message-ID: <CAEJMVkDA=phYV6MOiz5Jrj+g4q8-6G6XgvmVF2nXbh4vR7Vjzw@mail.gmail.com>

https://github.com/weave-lab/flanders

On Tue, Mar 10, 2015 at 6:06 AM, Steven Ayre <steveayre at gmail.com> wrote:

> There are also ESL events you can subscribe to
> http://wiki.freeswitch.org/wiki/Sofia-SIP#Custom_Events
>
> On 10 March 2015 at 06:21, Tito Cumpen <soulofmischief87 at gmail.com> wrote:
>
>> Richard,
>>
>> You may view registrations through the fs_cli console. You can get very
>> insightful debug ibformation through Sofia.check out Sofia debug
>> http://wiki.freeswitch.org/wiki/Sofia-SIP
>> On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:
>>
>>> Hi All,
>>> Is it possible to see when sip registrations happen in real time?
>>>
>>> Thanks
>>>
>>> Richard
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/2f38ccce/attachment-0001.html 

From anuraagmishra92 at gmail.com  Tue Mar 10 18:13:31 2015
From: anuraagmishra92 at gmail.com (Anurag Mishra)
Date: Tue, 10 Mar 2015 20:43:31 +0530
Subject: [Freeswitch-users] G729 Codec is not loading
In-Reply-To: <CAEJMVkCLPzkX1YCy1fEV_L-opJ=eBkzuJwqJD_Zoza9dKrbS4w@mail.gmail.com>
References: <CACGrPnhCjZ35mqONcAGYo8WwntLUTTWbEREh=BufE0UinLMEjw@mail.gmail.com>
	<CAEJMVkCLPzkX1YCy1fEV_L-opJ=eBkzuJwqJD_Zoza9dKrbS4w@mail.gmail.com>
Message-ID: <CACGrPnhdJWNsgGcx5iSgWtzHCW+Bivk8Pb6d8z2=oAj_SB6kSQ@mail.gmail.com>

Hi Brain,

Thanks for your reply, I am using "load mod_com_g729" and when I have run
this command nothing happen but just the prompt.

Thanks,
Anurag Mishra

On Tue, Mar 10, 2015 at 8:39 PM, Brian West <brian at freeswitch.org> wrote:

> Logs, what commands are you running or some more details about what you
> see on your screen would be helpful.
>
>
> On Tue, Mar 10, 2015 at 7:38 AM, Anurag Mishra <anuraagmishra92 at gmail.com>
> wrote:
>
>> Hi All,
>>
>> Few days back, I have installed G729 Codec that time everything was fine.
>> But suddenly its stopped working, I was tried to load it again from CLI but
>> nothing happened, its shows just a prompt without any error.
>>
>> Please advise.
>>
>> Thanks,
>> Anurag Mishra
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Anurag Mishra
*Viithiisys Technologies*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/3c0fa277/attachment.html 

From zvi at lexifone.com  Tue Mar 10 18:17:51 2015
From: zvi at lexifone.com (Zvi Agmon)
Date: Tue, 10 Mar 2015 17:17:51 +0200
Subject: [Freeswitch-users] Call codec sample rate and call recording
 sample rate relation
In-Reply-To: <CAEJMVkA+0R1Y5TVrQQAqMYt7pB1sLy2Yqohs8ryA_TZDL61jFg@mail.gmail.com>
References: <CAKYJ0qZA89PZyDdp_pX-K5C1Z2=U8jgi7FKt5F0p0QPfGdJbRg@mail.gmail.com>
	<CAEJMVkA+0R1Y5TVrQQAqMYt7pB1sLy2Yqohs8ryA_TZDL61jFg@mail.gmail.com>
Message-ID: <CAKYJ0qas+x5sFELcPPb6p83ZcAG1evX_Og2cTw36ECznGC3NAw@mail.gmail.com>

Thanks Brian for the quick reply.

So, I understand that the recording audio quality (in terms of sample rate)
is the same as the call audio quality - is that correct?

Thanks
Zvi Agmon

Best regards

Zvi Agmon
Lexifone <http://www.lexifone.com/>
email: zvi at lexifone.com
Office: +972-4-6817711
Cell: +972-54-4505109

On Tue, Mar 10, 2015 at 4:57 PM, Brian West <brian at freeswitch.org> wrote:

> By default the channels sample rate is at is what we will record at, there
> is no reason to set record_sample_rate in those cases.
>
> On Tue, Mar 10, 2015 at 5:02 AM, Zvi Agmon <zvi at lexifone.com> wrote:
>
>> Hello,
>>
>> I'm trying to understand the relation between the the call codec and
>> recording format in terms of sample rate and audio quality.
>> The question is: when a call use a wide band codec (i.e. speex at 16kh) and
>> then saved with "record_sample_rate=16000" -  does the saved file keeps
>> the 16kh audio quality?
>>
>>
>> Thanks a lot
>>
>> Zvi Agmon
>> Lexifone <http://www.lexifone.com/>
>> email: zvi at lexifone.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/6e923a08/attachment.html 

From olegstolyar at gmail.com  Tue Mar 10 18:32:59 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 08:32:59 -0700
Subject: [Freeswitch-users] WebRTC, CPU Load and Connection quality
Message-ID: <CAC8JMO+MTH_s6rj_yShVa3MuTy64G1qs8qLp==F+ZkgSSMpW=g@mail.gmail.com>

Hi guys,

Not really a question - just an observation.

I have a system where my internal users use WebRTC to connect to
FreeSWITCH.

My FS servers are in the US.

I noticed that CPU utilization per call is much higher for callers from
Brazil than from the US or Europe.  The difference is 2-3 times.  I am
guessing that it's because WebRTC has to do more retries on a bad
connection causing this difference.

Any comments?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/5361b1c8/attachment.html 

From paul.atreides83 at googlemail.com  Tue Mar 10 18:57:32 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Tue, 10 Mar 2015 16:57:32 +0100
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAKzWOxXAnfh47aLdL0OWik2jYeRH6c7tO5zCSOwq3YZCrcQBDA@mail.gmail.com>
References: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
	<CAKzWOxXAnfh47aLdL0OWik2jYeRH6c7tO5zCSOwq3YZCrcQBDA@mail.gmail.com>
Message-ID: <CAP6oyxGU7gRKG6H7UhxHiq=h6rmBd_RzEyYpYdMHRxm8FnrBYA@mail.gmail.com>

>In other words, if user A did a blind x-fer of caller C to user B and user
B doesn't answer (for whatever failure reason) then >caller C would start
ringing back to user A? Just making sure we understand the scope of the
feature you're implementing.

Exactly, at the moment the GXP disconnects A and makes a new call to B by
using the default dial plan. So freeswitch thinks its a normal call and
precedes as usual. I need a condition where I can check if its a transfer /
attended call or not

In the log below I found the channel variable
variable_pre_transfer_caller_id_number and variable_sip_refer_to but these
are are only set when it is a blind transfer. But for some reason there is
no match for the condition.

Here is my default dial plan
http://pastebin.freeswitch.org/23993

Here is the debug from the console when I start the transfer.
http://pastebin.freeswitch.org/23994



On Mon, Mar 9, 2015 at 7:18 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sat, Mar 7, 2015 at 9:03 AM, Paul Atreides <
> paul.atreides83 at googlemail.com> wrote:
>
>> When I do a blind transfer then I want freeswitch to call back the origin
>> who initiated the call.
>> But I am not able the capture the transfer event?
>>
>
> In other words, if user A did a blind x-fer of caller C to user B and user
> B doesn't answer (for whatever failure reason) then caller C would start
> ringing back to user A? Just making sure we understand the scope of the
> feature you're implementing.
>
> How does the GXP do the x-fer? Some kind of hook-flash and DTMF code? Can
> you pastebin the dialplan that the transferor uses when sending the call?
>
> -MC
>
>
>> <condition field="destination_number" expression="^att_xfer$">
>> <condition field="destination_number" expression="^is_transfer$">
>>
>> They seem do be ignored by the dialplan. Is there a list what kind of
>> values destionation_number can have besides the called numbers?
>>
>>
>> I am doing the transfer with a grandstream gxp2140
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/d364a181/attachment-0001.html 

From mishehu at freeswitch.org  Tue Mar 10 19:14:24 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Tue, 10 Mar 2015 11:14:24 -0500
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAFiqYumb2aFP27eHPhkNFqKCdg=mSbNzxxNYu=hW0cSrSjgiuA@mail.gmail.com>
References: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>	<D123B137.9810D%krice@freeswitch.org>	<CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>
	<CAFiqYumb2aFP27eHPhkNFqKCdg=mSbNzxxNYu=hW0cSrSjgiuA@mail.gmail.com>
Message-ID: <54FF1860.4030504@freeswitch.org>

Not only that but it seems that it works differently on different builds 
and/or versions of git.  I only receive the first two messages about 
expected 'committer' line, but not the others, on my system.  I guess 
it's sometimes better to just sweep some very minor things under the rug 
than trying to gut out everything to fix it - especially when what we're 
talking about is simply a metadata "problem".

-- 
Yossi Neiman


On 03/10/2015 08:56 AM, Steven Ayre wrote:
> I suspect it's performing some sanity checks that git commit does not. 
> Which you could then argue is either a bug or feature in git.
>
>
>
> On 10 March 2015 at 11:33, Ben Langfeld <ben at langfeld.co.uk 
> <mailto:ben at langfeld.co.uk>> wrote:
>
>     git fsck complains about it.
>
>     ? git fsck
>     Checking object directories: 100% (256/256), done.
>     error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid
>     format - expected 'committer' line
>     error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid
>     format - expected 'committer' line
>     Checking objects: 100% (294760/294760), done.
>     dangling commit f62e8f0e250b35bbadb7367082127304f3e72e74
>     dangling commit dc844f983605a015bf600232864122b94a278e76
>
>     On 9 March 2015 at 23:21, Ken Rice <krice at freeswitch.org
>     <mailto:krice at freeswitch.org>> wrote:
>
>         Git itself doesn?t say this is an invalid commit (we use
>         stash/git and its perfectly fine with this)... There is a
>         git-lint process that github uses and it rejects this commit
>
>
>         On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk
>         <http://ben at langfeld.co.uk>> wrote:
>
>             I understand very well why rewriting history is
>             undesirable and how git works. What I wonder is what
>             process was used to convince git to create a commit which
>             it would later say is invalid. As I said, I'm curious.
>
>             On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com
>             <http://steveayre at gmail.com>> wrote:
>
>                 Plus it's rather annoying to do so (rewrite history).
>                 The identifier of each commit is a hash computed from
>                 the content of the commit plus the metadata which
>                 includes the authors. Changing the author would change
>                 the identifier of the commit. That then changes the
>                 identifier of every commit afterwards. That then
>                 breaks every checkout / fork based off the tree as
>                 they no longer know where they are forked from. And
>                 since identifiers have all been rewritten we would no
>                 longer know what version you were running, or what
>                 version bug reports were reported against.
>
>                 (that's why git makes it easy to amend your latest
>                 uncommitted commit message but rather difficult to
>                 edit any others)
>
>
>
>
>                 On 9 March 2015 at 13:41, Michael Jerris
>                 <mike at jerris.com <http://mike at jerris.com>> wrote:
>
>                     We are not rewriting history to fix this so it
>                     doesn't really matter who is right or wrong.
>
>                     Mike
>
>                         On Mar 9, 2015, at 9:13 AM, Ben Langfeld
>                         <ben at langfeld.co.uk
>                         <http://ben at langfeld.co.uk>> wrote:
>
>                         I'm curious about this one... If git fsck
>                         complains about the issue, what is the
>                         justification for saying that Github is
>                         broken? How were these commits created with a
>                         format that git itself complains about?
>
>                         On 9 March 2015 at 09:01,  <cmrienzo at gmail.com
>                         <http://cmrienzo at gmail.com>> wrote:
>
>                             I switched to bitbucket.org
>                             <http://bitbucket.org>
>                             <http://bitbucket.org/>  just for the
>                             FreeSWITCH repo to work around this.
>
>
>                             On Mar 8, 2015, at 20:34, Ken Rice
>                             <krice at freeswitch.org
>                             <http://krice at freeswitch.org>> wrote:
>
>                                 This is a known issue with github and
>                                 will not be fixed
>
>
>
>                                 On 3/8/15, 3:39 PM, "Podrigal, Aron"
>                                 <aronp at guaranteedplus.com
>                                 <http://aronp at guaranteedplus.com>
>                                 <http://aronp at guaranteedplus.com/> >
>                                 wrote:
>
>                                     Hi,
>
>                                     I'm trying to push the freeswitch
>                                     git repo to my github, but I get
>                                     the following error
>
>
>                                     remote: error: object
>                                     487128950df6ee433c131b5feaafe81ee86629f4:invalid
>                                     format - expected 'committer' line
>                                     remote: fatal: Error in object
>                                     channel_by_id: 0: bad id: channel free
>                                     Received window adjust for
>                                     non-open channel 0.
>                                     error: pack-objects died of signal 13
>
>                                     This is caused by having multiple
>                                     authors on a commit (which in
>                                     general is not allowed by git) and
>                                     github verifies the commits and
>                                     rejects it.
>
>                                     here is the output of git fsck
>
>                                     Checking object directories: 100%
>                                     (256/256), done.
>                                     error in commit
>                                     487128950df6ee433c131b5feaafe81ee86629f4:
>                                     invalid format - expected
>                                     'committer' line
>                                     error in commit
>                                     8574988c3a378b4d5861ecaeb0e958657635703b:
>                                     invalid format - expected
>                                     'committer' line
>                                     Checking objects: 100%
>                                     (254227/254227), done.
>
>
>
>                     _________________________________________________________________________
>                     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://confluence.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://confluence.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://confluence.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
>         Twitter: @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://confluence.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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/a23a60ba/attachment-0001.html 

From ben at langfeld.co.uk  Tue Mar 10 19:27:56 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Tue, 10 Mar 2015 13:27:56 -0300
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <54FF1860.4030504@freeswitch.org>
References: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>
	<D123B137.9810D%krice@freeswitch.org>
	<CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>
	<CAFiqYumb2aFP27eHPhkNFqKCdg=mSbNzxxNYu=hW0cSrSjgiuA@mail.gmail.com>
	<54FF1860.4030504@freeswitch.org>
Message-ID: <CAAyX+KHaDutirGmQ-=Z97+bpiBwiHvPp6hdBgnVD=uEJy8cMUg@mail.gmail.com>

Sure, that's fine. As I've said, I'm not campaigning for any kind of fix,
I'm simply trying to understand how it happened. Might be useless
information, but if anyone knows, I'd be interested to find out.

It does seem unfair, however, to blame it on Github ;)

On 10 March 2015 at 13:14, I put the Who? in Mishehu <mishehu at freeswitch.org
> wrote:

>  Not only that but it seems that it works differently on different builds
> and/or versions of git.  I only receive the first two messages about
> expected 'committer' line, but not the others, on my system.  I guess it's
> sometimes better to just sweep some very minor things under the rug than
> trying to gut out everything to fix it - especially when what we're talking
> about is simply a metadata "problem".
>
> --
> Yossi Neiman
>
>
>
> On 03/10/2015 08:56 AM, Steven Ayre wrote:
>
> I suspect it's performing some sanity checks that git commit does not.
> Which you could then argue is either a bug or feature in git.
>
>
>
> On 10 March 2015 at 11:33, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> git fsck complains about it.
>>
>>  ? git fsck
>> Checking object directories: 100% (256/256), done.
>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>> - expected 'committer' line
>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>> - expected 'committer' line
>>  Checking objects: 100% (294760/294760), done.
>> dangling commit f62e8f0e250b35bbadb7367082127304f3e72e74
>> dangling commit dc844f983605a015bf600232864122b94a278e76
>>
>> On 9 March 2015 at 23:21, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Git itself doesn?t say this is an invalid commit (we use stash/git and
>>> its perfectly fine with this)... There is a git-lint process that github
>>> uses and it rejects this commit
>>>
>>>
>>> On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>
>>>  I understand very well why rewriting history is undesirable and how
>>> git works. What I wonder is what process was used to convince git to create
>>> a commit which it would later say is invalid. As I said, I'm curious.
>>>
>>> On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Plus it's rather annoying to do so (rewrite history). The identifier of
>>> each commit is a hash computed from the content of the commit plus the
>>> metadata which includes the authors. Changing the author would change the
>>> identifier of the commit. That then changes the identifier of every commit
>>> afterwards. That then breaks every checkout / fork based off the tree as
>>> they no longer know where they are forked from. And since identifiers have
>>> all been rewritten we would no longer know what version you were running,
>>> or what version bug reports were reported against.
>>>
>>> (that's why git makes it easy to amend your latest uncommitted commit
>>> message but rather difficult to edit any others)
>>>
>>>
>>>
>>>
>>> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> We are not rewriting history to fix this so it doesn't really matter who
>>> is right or wrong.
>>>
>>> Mike
>>>
>>>  On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>>
>>> I'm curious about this one... If git fsck complains about the issue,
>>> what is the justification for saying that Github is broken? How were these
>>> commits created with a format that git itself complains about?
>>>
>>> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>>>
>>> I switched to bitbucket.org <http://bitbucket.org/>  just for the
>>> FreeSWITCH repo to work around this.
>>>
>>>
>>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>  This is a known issue with github and will not be fixed
>>>
>>>
>>>
>>>   On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com <
>>> http://aronp at guaranteedplus.com/> > wrote:
>>>
>>>    Hi,
>>>
>>> I'm trying to push the freeswitch git repo to my github, but I get the
>>> following error
>>>
>>>
>>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>>> format - expected 'committer' line
>>> remote: fatal: Error in object
>>> channel_by_id: 0: bad id: channel free
>>> Received window adjust for non-open channel 0.
>>> error: pack-objects died of signal 13
>>>
>>> This is caused by having multiple authors on a commit (which in general
>>> is not allowed by git) and github verifies the commits and rejects it.
>>>
>>> here is the output of git fsck
>>>
>>> Checking object directories: 100% (256/256), done.
>>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>>> - expected 'committer' line
>>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>>> - expected 'committer' line
>>> Checking objects: 100% (254227/254227), done.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>> Twitter: @FreeSWITCH
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/a16e130d/attachment-0001.html 

From ing.antonyam at gmail.com  Tue Mar 10 21:00:40 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Tue, 10 Mar 2015 12:00:40 -0600
Subject: [Freeswitch-users] Synchronized shared database with XML_CURL
Message-ID: <CABFbAiW_GuXiVA5vgGPrbU2GrZqy74QYJ-KCAb3Ljmm5gRXitQ@mail.gmail.com>

Hi,

I want to configure 2 freeswitch synchronized with shared database by
xml_curl module [directory].

Currently I have problem are 2 different domains [1.1.1.1] [1.1.1.2] and
xml_curl must configure a domain directory:

<section name = "directory" description = "FreeSWITCH Directory">
<domain name = "$$ {domain}">
<params>

testing directly with the domain if it works but occupy a variable as $$
{domain} or local_ip_v4 and does not work.

Any idea to work?

Regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/561fe855/attachment.html 

From zoell at zoell.us  Tue Mar 10 21:22:03 2015
From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=)
Date: Tue, 10 Mar 2015 18:22:03 +0000
Subject: [Freeswitch-users] Set channel variables before bridge leg B
	hangup
In-Reply-To: <CAJrD+vcvPGNepvDSNCffE5Mxw4Lx4+ZxfxJEPksUdMktrpW9AA@mail.gmail.com>
References: <CAJrD+vfD1++606CPE9+zWf7gUCTRD5K-Q-rPfLr-5GcdmAp=NQ@mail.gmail.com>
	<CAC-LwPNmM+h5UgmSWMpJfOjseAq5+2zu0QEF3HYRPkKKyQO9aw@mail.gmail.com>
	<CAJrD+vcvPGNepvDSNCffE5Mxw4Lx4+ZxfxJEPksUdMktrpW9AA@mail.gmail.com>
Message-ID: <CAJrD+vfZ8eOU_=Bsi2yNaiQet+erV4pdCecCmidZxv6oxoKQDA@mail.gmail.com>

So this is not possible, right?

2015-03-06 8:53 GMT+00:00 Zolt?n Szab? <zoell at zoell.us>:

> How can I reference the session variable in the hook lua script?
>
> Thank you
>
> 2015-03-05 17:10 GMT+00:00 Vik Killa <vipkilla at gmail.com>:
>
>> You could try using the api_on_hangup to set a variable.
>> or there maybe an execute_on_hangup too.
>>
>> On Thu, Mar 5, 2015 at 12:06 PM, Zolt?n Szab? <zoell at zoell.us> wrote:
>>
>>> Hi,
>>>
>>> In lua I bridge two sessions. When leg B hangup the call I need to set
>>> up some custom channel variables for odbc_cdr reporting.
>>>
>>> freeswitch.bridge(session1, session2);
>>> session2:execute("set", "custom_var1=asdf");
>>>
>>> But when the set command tries to run, the log says "channel is hangup
>>> already".
>>>
>>> Is there any way to do this properly?
>>>
>>> Many thanks,
>>> Zoltan
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/62d46f11/attachment.html 

From s.safarov at gmail.com  Tue Mar 10 21:25:41 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 10 Mar 2015 18:25:41 +0000
Subject: [Freeswitch-users] Synchronized shared database with XML_CURL
References: <CABFbAiW_GuXiVA5vgGPrbU2GrZqy74QYJ-KCAb3Ljmm5gRXitQ@mail.gmail.com>
Message-ID: <CAHtxdDcL785pROcs0gxK1ECHWtPcStjWtKRZmAEbeX8mJzeSKg@mail.gmail.com>

Required configure domain variable in vars.xml

??, 10 ????? 2015, 21:02, Antony Aguirre Morales <ing.antonyam at gmail.com>:

> Hi,
>
> I want to configure 2 freeswitch synchronized with shared database by
> xml_curl module [directory].
>
> Currently I have problem are 2 different domains [1.1.1.1] [1.1.1.2] and
> xml_curl must configure a domain directory:
>
> <section name = "directory" description = "FreeSWITCH Directory">
> <domain name = "$$ {domain}">
> <params>
>
> testing directly with the domain if it works but occupy a variable as $$
> {domain} or local_ip_v4 and does not work.
>
> Any idea to work?
>
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/c8e54f96/attachment.html 

From dp.siddharth at eng.knowlarity.com  Tue Mar 10 22:00:02 2015
From: dp.siddharth at eng.knowlarity.com (DP Siddharth)
Date: Wed, 11 Mar 2015 00:30:02 +0530
Subject: [Freeswitch-users] ESL object taking large memory
In-Reply-To: <CAFDf6-9Rn1ro4yYhhEMxGH5jeYv4zsyRS2Yw8KhaOTiYKSa4tg@mail.gmail.com>
References: <CAFDf6-9Rn1ro4yYhhEMxGH5jeYv4zsyRS2Yw8KhaOTiYKSa4tg@mail.gmail.com>
Message-ID: <CAFDf6-9wp5+V9v36ewbv=Y0WgpYmkt8SQuG8c5=FTg9OA=qw_A@mail.gmail.com>

Going through logs I found following commit id, where these changes were
done. Any suggetion about having

#define BUF_CHUNK 65536 * 50
#define BUF_START 65536 * 100

to some smaller values?

commit 2081bf97b9836f5299c22edbb1ead077842ea2bc
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Thu Dec 16 11:33:38 2010 -0600

    use a packet buffer for ESL

On Tue, Mar 10, 2015 at 6:50 PM, DP Siddharth <
dp.siddharth at eng.knowlarity.com> wrote:

> Hi All,
>
> I am working on python/esl based server. We are seeing memory getting
> increase by ~6.5MB when con = ESLconnection() get called.
>
> We tried to cleanup this object as soon calls gets complete, but somehow
> we are not successful.
>
> Further looking into esl/src/include/esl.h we found
>
> #define BUF_CHUNK 65536 * 50
> #define BUF_START 65536 * 100
>
> modifying these values help in controlling python server memory growth.
>
> Can someone help me in understanding why we have this buffer size of
> 6.5MB?
>
>
> --
> Thanks & Regards,
> D P Siddharth
> Director (Platform)
> Knowlarity Communications
> Ph: +919999115231
> dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>
>
> *"Come together to build a lasting world-class cloud telephony company
> that helps businesses grow"*
>



-- 
Thanks & Regards,
D P Siddharth
Director (Platform)
Knowlarity Communications
Ph: +919999115231
dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>

*"Come together to build a lasting world-class cloud telephony company that
helps businesses grow"*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/68913c6e/attachment.html 

From brian at freeswitch.org  Tue Mar 10 22:05:16 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 14:05:16 -0500
Subject: [Freeswitch-users] git push invalid format
In-Reply-To: <CAAyX+KHaDutirGmQ-=Z97+bpiBwiHvPp6hdBgnVD=uEJy8cMUg@mail.gmail.com>
References: <CAAyX+KF0fRr7CiPP11hT_106DNtY3QB2kBi6Ymh9o8Z-nE7aAg@mail.gmail.com>
	<D123B137.9810D%krice@freeswitch.org>
	<CAAyX+KFBNKoAY+yB3SV4H1-BLmJM4MyYFKk4t1t+Vc=qhAhnyA@mail.gmail.com>
	<CAFiqYumb2aFP27eHPhkNFqKCdg=mSbNzxxNYu=hW0cSrSjgiuA@mail.gmail.com>
	<54FF1860.4030504@freeswitch.org>
	<CAAyX+KHaDutirGmQ-=Z97+bpiBwiHvPp6hdBgnVD=uEJy8cMUg@mail.gmail.com>
Message-ID: <CAEJMVkCSwVHtuXONay0_=ZuA-ejutGyQq_G1-vAit0sZBPbOwQ@mail.gmail.com>

I think in the end it falls on git itself, why does it allow you to do this
in the first place if its not valid.

On Tue, Mar 10, 2015 at 11:27 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Sure, that's fine. As I've said, I'm not campaigning for any kind of fix,
> I'm simply trying to understand how it happened. Might be useless
> information, but if anyone knows, I'd be interested to find out.
>
> It does seem unfair, however, to blame it on Github ;)
>
> On 10 March 2015 at 13:14, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>>  Not only that but it seems that it works differently on different
>> builds and/or versions of git.  I only receive the first two messages about
>> expected 'committer' line, but not the others, on my system.  I guess it's
>> sometimes better to just sweep some very minor things under the rug than
>> trying to gut out everything to fix it - especially when what we're talking
>> about is simply a metadata "problem".
>>
>> --
>> Yossi Neiman
>>
>>
>>
>> On 03/10/2015 08:56 AM, Steven Ayre wrote:
>>
>> I suspect it's performing some sanity checks that git commit does not.
>> Which you could then argue is either a bug or feature in git.
>>
>>
>>
>> On 10 March 2015 at 11:33, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> git fsck complains about it.
>>>
>>>  ? git fsck
>>> Checking object directories: 100% (256/256), done.
>>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid format
>>> - expected 'committer' line
>>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid format
>>> - expected 'committer' line
>>>  Checking objects: 100% (294760/294760), done.
>>> dangling commit f62e8f0e250b35bbadb7367082127304f3e72e74
>>> dangling commit dc844f983605a015bf600232864122b94a278e76
>>>
>>> On 9 March 2015 at 23:21, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>>  Git itself doesn?t say this is an invalid commit (we use stash/git
>>>> and its perfectly fine with this)... There is a git-lint process that
>>>> github uses and it rejects this commit
>>>>
>>>>
>>>> On 3/9/15, 6:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>>
>>>>  I understand very well why rewriting history is undesirable and how
>>>> git works. What I wonder is what process was used to convince git to create
>>>> a commit which it would later say is invalid. As I said, I'm curious.
>>>>
>>>> On 9 March 2015 at 20:12, Steven Ayre <steveayre at gmail.com> wrote:
>>>>
>>>> Plus it's rather annoying to do so (rewrite history). The identifier of
>>>> each commit is a hash computed from the content of the commit plus the
>>>> metadata which includes the authors. Changing the author would change the
>>>> identifier of the commit. That then changes the identifier of every commit
>>>> afterwards. That then breaks every checkout / fork based off the tree as
>>>> they no longer know where they are forked from. And since identifiers have
>>>> all been rewritten we would no longer know what version you were running,
>>>> or what version bug reports were reported against.
>>>>
>>>> (that's why git makes it easy to amend your latest uncommitted commit
>>>> message but rather difficult to edit any others)
>>>>
>>>>
>>>>
>>>>
>>>> On 9 March 2015 at 13:41, Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>> We are not rewriting history to fix this so it doesn't really matter
>>>> who is right or wrong.
>>>>
>>>> Mike
>>>>
>>>>  On Mar 9, 2015, at 9:13 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>>>
>>>> I'm curious about this one... If git fsck complains about the issue,
>>>> what is the justification for saying that Github is broken? How were these
>>>> commits created with a format that git itself complains about?
>>>>
>>>> On 9 March 2015 at 09:01,  <cmrienzo at gmail.com> wrote:
>>>>
>>>> I switched to bitbucket.org <http://bitbucket.org/>  just for the
>>>> FreeSWITCH repo to work around this.
>>>>
>>>>
>>>> On Mar 8, 2015, at 20:34, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>  This is a known issue with github and will not be fixed
>>>>
>>>>
>>>>
>>>>   On 3/8/15, 3:39 PM, "Podrigal, Aron" <aronp at guaranteedplus.com <
>>>> http://aronp at guaranteedplus.com/> > wrote:
>>>>
>>>>    Hi,
>>>>
>>>> I'm trying to push the freeswitch git repo to my github, but I get the
>>>> following error
>>>>
>>>>
>>>> remote: error: object 487128950df6ee433c131b5feaafe81ee86629f4:invalid
>>>> format - expected 'committer' line
>>>> remote: fatal: Error in object
>>>> channel_by_id: 0: bad id: channel free
>>>> Received window adjust for non-open channel 0.
>>>> error: pack-objects died of signal 13
>>>>
>>>> This is caused by having multiple authors on a commit (which in general
>>>> is not allowed by git) and github verifies the commits and rejects it.
>>>>
>>>> here is the output of git fsck
>>>>
>>>> Checking object directories: 100% (256/256), done.
>>>> error in commit 487128950df6ee433c131b5feaafe81ee86629f4: invalid
>>>> format - expected 'committer' line
>>>> error in commit 8574988c3a378b4d5861ecaeb0e958657635703b: invalid
>>>> format - expected 'committer' line
>>>> Checking objects: 100% (254227/254227), done.
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/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://confluence.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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/a4b872a5/attachment-0001.html 

From olegstolyar at gmail.com  Tue Mar 10 22:08:00 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 12:08:00 -0700
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
Message-ID: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>

Hi guys,

I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
getting this error on FS:

2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet not
written

I tested it with the FS Verto test site and it also does not work.
https://webrtc.freeswitch.org/verto/index.html#page-main

Is this a known issue?

Of course 42 Beta also does not recognize Java for me, so hopefully it's
just a crappy build and Google will fix it but my concern is that other
WebRTC test sites still work, so should we look into this in case it's not
fixed by the time Chrome 42 becomes stable?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/3b7b550a/attachment.html 

From ssinyagin at gmail.com  Tue Mar 10 22:22:03 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 10 Mar 2015 20:22:03 +0100
Subject: [Freeswitch-users] Synchronized shared database with XML_CURL
In-Reply-To: <CABFbAiW_GuXiVA5vgGPrbU2GrZqy74QYJ-KCAb3Ljmm5gRXitQ@mail.gmail.com>
References: <CABFbAiW_GuXiVA5vgGPrbU2GrZqy74QYJ-KCAb3Ljmm5gRXitQ@mail.gmail.com>
Message-ID: <CACOYK=iJRGt2F8HyY_QBqN2pfn2isgYigGmS-=eK6mWDqRKUkA@mail.gmail.com>

I suggest you to stop using IP addresses as domain names, and go for
multi-tenant configuration.

Here I started to outline some details, but the document is by far not
ready yet:
https://github.com/voxserv/freeswitch_conf_minimal/blob/tutorials/docs/tutorial_01_simple_pbx.md



On Tue, Mar 10, 2015 at 7:00 PM, Antony Aguirre Morales
<ing.antonyam at gmail.com> wrote:
> Hi,
>
> I want to configure 2 freeswitch synchronized with shared database by
> xml_curl module [directory].
>
> Currently I have problem are 2 different domains [1.1.1.1] [1.1.1.2] and
> xml_curl must configure a domain directory:
>
> <section name = "directory" description = "FreeSWITCH Directory">
> <domain name = "$$ {domain}">
> <params>
>
> testing directly with the domain if it works but occupy a variable as $$
> {domain} or local_ip_v4 and does not work.
>
> Any idea to work?
>
> Regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From auge at virtues.net  Tue Mar 10 22:28:37 2015
From: auge at virtues.net (Thomas Auge)
Date: Tue, 10 Mar 2015 16:28:37 -0300
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
 FreeSWITCH
In-Reply-To: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
Message-ID: <54FF45E5.6020203@virtues.net>

M43 (canary) works fine (with opus).

On 10.03.2015 16:08, Oleg Stolyar wrote:
> Hi guys,
>
> I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am getting this error on FS:
>
> 2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet not written
>
> I tested it with the FS Verto test site and it also does not work.
> https://webrtc.freeswitch.org/verto/index.html#page-main
>
> Is this a known issue?
>
> Of course 42 Beta also does not recognize Java for me, so hopefully it's just a crappy build and Google will fix it 
> but my concern is that other WebRTC test sites still work, so should we look into this in case it's not fixed by the 
> time Chrome 42 becomes stable?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/dac2a26a/attachment.html 

From mike at jerris.com  Tue Mar 10 22:32:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 10 Mar 2015 15:32:24 -0400
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <54FF45E5.6020203@virtues.net>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
Message-ID: <014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>

They did all kinds of messed up stuff between 41 and 43.  43 is still somewhat broken as to video and using different resolutions.  I would not be surprised if 42 is complete garbage.

> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
> 
> M43 (canary) works fine (with opus).
> 
> On 10.03.2015 16:08, Oleg Stolyar wrote:
>> Hi guys,
>> 
>> I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am getting this error on FS:
>> 
>> 2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet not written
>> 
>> I tested it with the FS Verto test site and it also does not work.
>> https://webrtc.freeswitch.org/verto/index.html#page-main <https://webrtc.freeswitch.org/verto/index.html#page-main>
>> 
>> Is this a known issue?
>> 
>> Of course 42 Beta also does not recognize Java for me, so hopefully it's just a crappy build and Google will fix it but my concern is that other WebRTC test sites still work, so should we look into this in case it's not fixed by the time Chrome 42 becomes stable?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/b1260c80/attachment.html 

From Sharath.Kumar at meZocliq.com  Tue Mar 10 22:44:57 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Tue, 10 Mar 2015 19:44:57 +0000
Subject: [Freeswitch-users] Outbound call status and conference_auto_outcall
Message-ID: <DM2PR0501MB1101F67817A5AFFDE95106B984180@DM2PR0501MB1101.namprd05.prod.outlook.com>

Hello all,


I am using "conference_auto_outcall" to dial a bunch of extensions. It works great. However, I need to know the status of whether the calls were answered or rejected ? Is there any way I can know this from the dialplan ? or is there any api that I can call from the dialplan ?
Also, I am aware there is some sort of presence module but these phones are webrtc based and don't yet really support any presence standards.

I looked around for a solution to this and couldn't find a solution. I appreciate if anyone can point me in the right direction.

Thanks,
Sharath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/bd0122b8/attachment.html 

From olegstolyar at gmail.com  Tue Mar 10 23:06:25 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 13:06:25 -0700
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
Message-ID: <CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>

Canary 43 does not work for me either.  Same symptoms both on my own FS and
on the Verto Demo site (with Opus).

On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com> wrote:

> They did all kinds of messed up stuff between 41 and 43.  43 is still
> somewhat broken as to video and using different resolutions.  I would not
> be surprised if 42 is complete garbage.
>
> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>
>  M43 (canary) works fine (with opus).
>
> On 10.03.2015 16:08, Oleg Stolyar wrote:
>
> Hi guys,
>
>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
> getting this error on FS:
>
>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet not
> written
>
>  I tested it with the FS Verto test site and it also does not work.
> https://webrtc.freeswitch.org/verto/index.html#page-main
>
>  Is this a known issue?
>
>  Of course 42 Beta also does not recognize Java for me, so hopefully it's
> just a crappy build and Google will fix it but my concern is that other
> WebRTC test sites still work, so should we look into this in case it's not
> fixed by the time Chrome 42 becomes stable?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/4aeb5cbe/attachment-0001.html 

From richard.mace at gmail.com  Tue Mar 10 23:22:02 2015
From: richard.mace at gmail.com (Richard Mace)
Date: Tue, 10 Mar 2015 20:22:02 +0000
Subject: [Freeswitch-users] fs_cli will not connect on Fresh install on
	Debian
In-Reply-To: <CACOYK=hNYnYNj965sRvbf4e554bY-V-ZJfRvouGpSw1Y1AKcJQ@mail.gmail.com>
References: <CAK+KT3_jMQ+j5X-roq=UHDWu-i-BmwrwVfu+9BzfWkGVsPz4+w@mail.gmail.com>
	<00c301d05848$833f55c0$89be0140$@botecomm.com>
	<CAK+KT3_dvWvtc8sENFP02cA2NfhA8=xf-sOhtYMGetHkZ8w7aQ@mail.gmail.com>
	<CAEJMVkA1mrgBLW8CaOJ3SUQE3mSDRLLmXAh4cCFm1ujJ8UO7aw@mail.gmail.com>
	<CAK+KT38qwy63i5HKkp71OGPb8ctVAfYZbhbHPZr_fWYmy3v-KA@mail.gmail.com>
	<CAK+KT38qBuf7xWW9zP26KkttMycugw+ySZhErJYGufaeArZrWQ@mail.gmail.com>
	<04bb01d05b01$6996cd90$3cc468b0$@botecomm.com>
	<CAK+KT384JsNWPtSFEUw3J=HPQ_GOOvVbeBfwt1ONAiMutrdPzA@mail.gmail.com>
	<CACOYK=hC815p6reUmkG4jVyqcxeGEToVgZ+RTXGvkqjzA9a0dw@mail.gmail.com>
	<CACOYK=gbX4YhBYPVHkdbzWeLRRSRdNOKFopkOLkHLXCW63O2MQ@mail.gmail.com>
	<CAK+KT389JzEZ2QO2HJqtwM2P7dW9Hx=Ej4A5kGMPOo6fjkRJEA@mail.gmail.com>
	<CAFiqYuk3O9PSj51kn8usZhXS3MyyJdi6pJ8+psbGx3Q94iVTwg@mail.gmail.com>
	<CACOYK=hNYnYNj965sRvbf4e554bY-V-ZJfRvouGpSw1Y1AKcJQ@mail.gmail.com>
Message-ID: <CAK+KT3_jeWE7GAq+a6FVcE_+UUp27VfQDPQC=21kD-DJHDPPSg@mail.gmail.com>

Thanks Stanislav,
Much appreciated.

Richard

On 10 March 2015 at 11:19, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> I submitted https://freeswitch.org/jira/browse/FS-7358
>
> mod_event_socket is loaded, and then
> 2015-03-10 11:29:48.367067 [NOTICE] switch_loadable_module.c:102
> Thread ended for mod_event_socket
>
>
>
> On Tue, Mar 10, 2015 at 12:04 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > Are you sure it's trying to load mod_event_socket? Unless it loads that
> > module it won't listen on 8021.
> >
> > If it is trying to, perhaps start freeswitch in the foreground (-c
> option)
> > and then (re)load the module once it's running. If there's an error
> that'll
> > hopefully show it up if it's not making it to the log file.
> >
> > On 10 March 2015 at 09:24, Richard Mace <richard.mace at gmail.com> wrote:
> >>
> >> Nothing, by the looks of things
> >>
> >> root at FreeSWITCH:~# netstat -an | grep 8021
> >> root at FreeSWITCH:~#
> >>
> >> Thanks
> >>
> >>
> >> On 10 March 2015 at 09:07, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> >>>
> >>> Richard, what do you see in the output:
> >>>
> >>> netstat -an | grep 8021
> >>>
> >>>
> >>>
> >>> On Tue, Mar 10, 2015 at 10:06 AM, Stanislav Sinyagin
> >>> <ssinyagin at gmail.com> wrote:
> >>> > this is exactly what I saw on a system which had no IPv4 address on
> >>> > its ethernet ports. FreeSWITCH was just not listening to port 8021,
> >>> > without any errors in the log.
> >>> >
> >>> > On Tue, Mar 10, 2015 at 9:16 AM, Richard Mace <
> richard.mace at gmail.com>
> >>> > wrote:
> >>> >> Hi Bote,
> >>> >> Tried this as well, on the local machine:
> >>> >>
> >>> >> root at FreeSWITCH:~# fs_cli 127.0.0.1
> >>> >> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
> >>> >> Error]
> >>> >>
> >>> >> root at FreeSWITCH:~# /etc/init.d/freeswitch status
> >>> >> [ ok ] freeswitch is running.
> >>> >>
> >>> >> Richard
> >>> >>
> >>> >>
> >>> >> On 10 March 2015 at 07:11, Bote Man <bote_radio at botecomm.com>
> wrote:
> >>> >>>
> >>> >>> You could explicitly direct fs_cli to a particular i.p. address
> >>> >>> either on
> >>> >>> the command line or using a profile definition.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Bote
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> From: Richard Mace
> >>> >>> Sent: Tuesday, 10 March, 2015 02:20
> >>> >>> To: FreeSWITCH Users Help
> >>> >>> Subject: Re: [Freeswitch-users] fs_cli will not connect on Fresh
> >>> >>> install
> >>> >>> on Debian
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Hi,
> >>> >>>
> >>> >>> Is there an fs_cli command that I can use that will get me round
> the
> >>> >>> current bug?
> >>> >>>
> >>> >>> Its strange as it's only happened within the last month, as I
> built a
> >>> >>> system recently that worked fine out of the box.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Thanks
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Richard
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> On 9 March 2015 at 10:37, Richard Mace <richard.mace at gmail.com>
> >>> >>> wrote:
> >>> >>>
> >>> >>> Hi Brian,
> >>> >>>
> >>> >>> Removed the line, and rebooted, but still getting:
> >>> >>>
> >>> >>> root at FreeSWITCH:~# fs_cli
> >>> >>> [ERROR] fs_cli.c:1565 main() Error Connecting [Socket Connection
> >>> >>> Error]
> >>> >>>
> >>> >>> Richard
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> On 6 March 2015 at 20:42, Brian West <brian at freeswitch.org> wrote:
> >>> >>>
> >>> >>> remove
> >>> >>>
> >>> >>> ::1     localhost ip6-localhost ip6-loopback
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> from /etc/hosts
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> its a bug in debian.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> On Fri, Mar 6, 2015 at 3:33 PM, Richard Mace <
> richard.mace at gmail.com>
> >>> >>> wrote:
> >>> >>>
> >>> >>> Hi,
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Sorry, I should have clarified that this is running locally on the
> >>> >>> machine
> >>> >>> running FreeSWITCH.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Richard
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> On 6 March 2015 at 20:02, Bote Man <bote_radio at botecomm.com>
> wrote:
> >>> >>>
> >>> >>> On a fresh FS installation fs_cli only connects to 127.0.01
> >>> >>> localhost.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> To connect from a remote machine put a valid routable interface
> >>> >>> address
> >>> >>> (although I have 0.0.0.0 in mine) in
> >>> >>>
> >>> >>> conf/autoload_configs/event_socket.conf.xml
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> and change the password and maybe even the port depending on the
> >>> >>> crackability of your network.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Then you?ll probably want to configure a profile configuration file
> >>> >>> with
> >>> >>> tight permissions to avoid having to type the parameters on the
> >>> >>> command line
> >>> >>> every time you start fs_cli.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Check the ?command-line Interface fs_cli? Confluence page for all
> the
> >>> >>> details.
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Bote
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> From: Richard Mace
> >>> >>> Sent: Friday, 06 March, 2015 14:34
> >>> >>> Subject: [Freeswitch-users] fs_cli will not connect on Fresh
> install
> >>> >>> on
> >>> >>> Debian
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Hi All,
> >>> >>>
> >>> >>> I did a fresh install of both Debian and FreeSWITCH today,
> following
> >>> >>> the
> >>> >>> article here:
> >>> >>>
> >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Debian
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> However, after installation, fs_cli will not connect. Any ideas?
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Thanks
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>> Richard
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> >>> >>>
> _________________________________________________________________________
> >>> >>> Professional FreeSWITCH Consulting Services:
> >>> >>> consulting at freeswitch.org
> >>> >>> http://www.freeswitchsolutions.com
> >>> >>>
> >>> >>> Official FreeSWITCH Sites
> >>> >>> http://www.freeswitch.org
> >>> >>> http://confluence.freeswitch.org
> >>> >>> http://www.cluecon.com
> >>> >>>
> >>> >>> FreeSWITCH-users mailing list
> >>> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>>
> >>> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> http://www.freeswitch.org
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> >>> >>
> _________________________________________________________________________
> >>> >> Professional FreeSWITCH Consulting Services:
> >>> >> consulting at freeswitch.org
> >>> >> http://www.freeswitchsolutions.com
> >>> >>
> >>> >> Official FreeSWITCH Sites
> >>> >> http://www.freeswitch.org
> >>> >> http://confluence.freeswitch.org
> >>> >> http://www.cluecon.com
> >>> >>
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/1acc4d0b/attachment-0001.html 

From brian at freeswitch.org  Tue Mar 10 23:53:18 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 15:53:18 -0500
Subject: [Freeswitch-users] Outbound call status and
	conference_auto_outcall
In-Reply-To: <DM2PR0501MB1101F67817A5AFFDE95106B984180@DM2PR0501MB1101.namprd05.prod.outlook.com>
References: <DM2PR0501MB1101F67817A5AFFDE95106B984180@DM2PR0501MB1101.namprd05.prod.outlook.com>
Message-ID: <CAEJMVkC+12mWx5jSZQnvcjTqNTnTA48AFjOhcfFpT89s51L9YQ@mail.gmail.com>

set the execute_on_hangup to a lua script to give you this possibly, or the
api_hangup_hook ... just off the top of my head that is.. would need
testing.

On Tue, Mar 10, 2015 at 2:44 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Hello all,
>
>
>
>
>
> I am using ?conference_auto_outcall? to dial a bunch of extensions. It
> works great. However, I need to know the status of whether the calls were
> answered or rejected ? Is there any way I can know this from the dialplan ?
> or is there any api that I can call from the dialplan ?
>
> Also, I am aware there is some sort of presence module but these phones
> are webrtc based and don?t yet really support any presence standards.
>
>
>
> I looked around for a solution to this and couldn?t find a solution. I
> appreciate if anyone can point me in the right direction.
>
>
>
> Thanks,
>
> Sharath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/3e7c467c/attachment.html 

From brian at freeswitch.org  Tue Mar 10 23:54:50 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 15:54:50 -0500
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
Message-ID: <CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>

Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org, Just
updated and tested.

On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Canary 43 does not work for me either.  Same symptoms both on my own FS
> and on the Verto Demo site (with Opus).
>
> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>> somewhat broken as to video and using different resolutions.  I would not
>> be surprised if 42 is complete garbage.
>>
>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>
>>  M43 (canary) works fine (with opus).
>>
>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>
>> Hi guys,
>>
>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
>> getting this error on FS:
>>
>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>> not written
>>
>>  I tested it with the FS Verto test site and it also does not work.
>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>
>>  Is this a known issue?
>>
>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>> it's just a crappy build and Google will fix it but my concern is that
>> other WebRTC test sites still work, so should we look into this in case
>> it's not fixed by the time Chrome 42 becomes stable?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/81546d8b/attachment.html 

From brian at freeswitch.org  Tue Mar 10 23:57:00 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 15:57:00 -0500
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
Message-ID: <CAEJMVkCbLU2DJocKKYDMyQ6Qn1kGVgUA9JD9aGDeN__M0z=OAA@mail.gmail.com>

Version 43.0.2328.0 canary (64-bit) seems broken.  41.0.2272.89 is what I
get when I updated my Chrome on my mac, which seems to work fine.

On Tue, Mar 10, 2015 at 3:54 PM, Brian West <brian at freeswitch.org> wrote:

> Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org, Just
> updated and tested.
>
> On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Canary 43 does not work for me either.  Same symptoms both on my own FS
>> and on the Verto Demo site (with Opus).
>>
>> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>>> somewhat broken as to video and using different resolutions.  I would not
>>> be surprised if 42 is complete garbage.
>>>
>>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>>
>>>  M43 (canary) works fine (with opus).
>>>
>>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>>
>>> Hi guys,
>>>
>>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
>>> getting this error on FS:
>>>
>>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>>> not written
>>>
>>>  I tested it with the FS Verto test site and it also does not work.
>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>
>>>  Is this a known issue?
>>>
>>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>>> it's just a crappy build and Google will fix it but my concern is that
>>> other WebRTC test sites still work, so should we look into this in case
>>> it's not fixed by the time Chrome 42 becomes stable?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/9c830ed2/attachment-0001.html 

From brian at freeswitch.org  Tue Mar 10 23:58:46 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Mar 2015 15:58:46 -0500
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAEJMVkCbLU2DJocKKYDMyQ6Qn1kGVgUA9JD9aGDeN__M0z=OAA@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
	<CAEJMVkCbLU2DJocKKYDMyQ6Qn1kGVgUA9JD9aGDeN__M0z=OAA@mail.gmail.com>
Message-ID: <CAEJMVkDfVAcbQwQf=O1E2Axsgfkox5P2HPmaLQuEe6Tto0SFhA@mail.gmail.com>

Chromium Version 42.0.2304.0 (64-bit) I built this morning is working.

On Tue, Mar 10, 2015 at 3:57 PM, Brian West <brian at freeswitch.org> wrote:

> Version 43.0.2328.0 canary (64-bit) seems broken.  41.0.2272.89 is what I
> get when I updated my Chrome on my mac, which seems to work fine.
>
> On Tue, Mar 10, 2015 at 3:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org,
>> Just updated and tested.
>>
>> On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>>
>>> Canary 43 does not work for me either.  Same symptoms both on my own FS
>>> and on the Verto Demo site (with Opus).
>>>
>>> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>>>> somewhat broken as to video and using different resolutions.  I would not
>>>> be surprised if 42 is complete garbage.
>>>>
>>>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>>>
>>>>  M43 (canary) works fine (with opus).
>>>>
>>>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>>>
>>>> Hi guys,
>>>>
>>>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
>>>> getting this error on FS:
>>>>
>>>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>>>> not written
>>>>
>>>>  I tested it with the FS Verto test site and it also does not work.
>>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>>
>>>>  Is this a known issue?
>>>>
>>>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>>>> it's just a crappy build and Google will fix it but my concern is that
>>>> other WebRTC test sites still work, so should we look into this in case
>>>> it's not fixed by the time Chrome 42 becomes stable?
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/752063cc/attachment.html 

From Sharath.Kumar at meZocliq.com  Wed Mar 11 00:07:43 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Tue, 10 Mar 2015 21:07:43 +0000
Subject: [Freeswitch-users] Outbound call status and
 conference_auto_outcall
Message-ID: <DM2PR0501MB11013DD0DDAB1704D06FBAC384180@DM2PR0501MB1101.namprd05.prod.outlook.com>

Thank you Brian!. I will test it and let you know.
-Sharath

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, March 10, 2015 4:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Outbound call status and conference_auto_outcall

set the execute_on_hangup to a lua script to give you this possibly, or the api_hangup_hook ... just off the top of my head that is.. would need testing.

On Tue, Mar 10, 2015 at 2:44 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
Hello all,


I am using ?conference_auto_outcall? to dial a bunch of extensions. It works great. However, I need to know the status of whether the calls were answered or rejected ? Is there any way I can know this from the dialplan ? or is there any api that I can call from the dialplan ?
Also, I am aware there is some sort of presence module but these phones are webrtc based and don?t yet really support any presence standards.

I looked around for a solution to this and couldn?t find a solution. I appreciate if anyone can point me in the right direction.

Thanks,
Sharath

_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[Image removed by sender.]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/8a0f8e97/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ~WRD000.jpg
Type: image/jpeg
Size: 823 bytes
Desc: ~WRD000.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/8a0f8e97/attachment-0001.jpg 

From olegstolyar at gmail.com  Wed Mar 11 00:07:58 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 14:07:58 -0700
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
Message-ID: <CAC8JMOLf9AdK5LPiHa7kO2y-ucC3m04BLArorc8nQdRnaViGYw@mail.gmail.com>

Hi Brian,

41 always worked fine for me and still does.  It's 42 beta and 43 canary
that are not working.

On Tue, Mar 10, 2015 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:

> Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org, Just
> updated and tested.
>
> On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Canary 43 does not work for me either.  Same symptoms both on my own FS
>> and on the Verto Demo site (with Opus).
>>
>> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>>> somewhat broken as to video and using different resolutions.  I would not
>>> be surprised if 42 is complete garbage.
>>>
>>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>>
>>>  M43 (canary) works fine (with opus).
>>>
>>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>>
>>> Hi guys,
>>>
>>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
>>> getting this error on FS:
>>>
>>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>>> not written
>>>
>>>  I tested it with the FS Verto test site and it also does not work.
>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>
>>>  Is this a known issue?
>>>
>>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>>> it's just a crappy build and Google will fix it but my concern is that
>>> other WebRTC test sites still work, so should we look into this in case
>>> it's not fixed by the time Chrome 42 becomes stable?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/98bd714e/attachment.html 

From olegstolyar at gmail.com  Wed Mar 11 01:39:34 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 15:39:34 -0700
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAC8JMOLf9AdK5LPiHa7kO2y-ucC3m04BLArorc8nQdRnaViGYw@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
	<CAC8JMOLf9AdK5LPiHa7kO2y-ucC3m04BLArorc8nQdRnaViGYw@mail.gmail.com>
Message-ID: <CAC8JMO+sDeoR50EkNRNmD2gm0-s-Q51xGDse5RqVn+MduSRk8A@mail.gmail.com>

Update on this:

My laptop on which I got these errors is running Windows 8 (don't ask why
:-)).

After getting these errors I tested this on 42 and 43 on a Windows 7
machine and on a Chromebook.  Both worked fine!

I will try to find another windows 8 machine to check if this is a problem
with just my laptop or with Windows 8 in general.

If anyone is wondering - yes, I rebooted the machine :-).  Also Opera and
Firefox on the same machine are working just fine as did Chrome 41.

On Tue, Mar 10, 2015 at 2:07 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Hi Brian,
>
> 41 always worked fine for me and still does.  It's 42 beta and 43 canary
> that are not working.
>
> On Tue, Mar 10, 2015 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org,
>> Just updated and tested.
>>
>> On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>>
>>> Canary 43 does not work for me either.  Same symptoms both on my own FS
>>> and on the Verto Demo site (with Opus).
>>>
>>> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>>>> somewhat broken as to video and using different resolutions.  I would not
>>>> be surprised if 42 is complete garbage.
>>>>
>>>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>>>
>>>>  M43 (canary) works fine (with opus).
>>>>
>>>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>>>
>>>> Hi guys,
>>>>
>>>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am
>>>> getting this error on FS:
>>>>
>>>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>>>> not written
>>>>
>>>>  I tested it with the FS Verto test site and it also does not work.
>>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>>
>>>>  Is this a known issue?
>>>>
>>>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>>>> it's just a crappy build and Google will fix it but my concern is that
>>>> other WebRTC test sites still work, so should we look into this in case
>>>> it's not fixed by the time Chrome 42 becomes stable?
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/f8883cd8/attachment-0001.html 

From olegstolyar at gmail.com  Wed Mar 11 02:09:06 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 10 Mar 2015 16:09:06 -0700
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with
	FreeSWITCH
In-Reply-To: <CAC8JMO+sDeoR50EkNRNmD2gm0-s-Q51xGDse5RqVn+MduSRk8A@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>
	<54FF45E5.6020203@virtues.net>
	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>
	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>
	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>
	<CAC8JMOLf9AdK5LPiHa7kO2y-ucC3m04BLArorc8nQdRnaViGYw@mail.gmail.com>
	<CAC8JMO+sDeoR50EkNRNmD2gm0-s-Q51xGDse5RqVn+MduSRk8A@mail.gmail.com>
Message-ID: <CAC8JMOLKDfPn4sVJbBA=XCbZMQ2WZdYZNJt=4mrw3J5BAFq1UQ@mail.gmail.com>

Last update:  Believe it or not this was solved by getting the latest
Windows 8 updates.

Sorry about the false alarm!

On Tue, Mar 10, 2015 at 3:39 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Update on this:
>
> My laptop on which I got these errors is running Windows 8 (don't ask why
> :-)).
>
> After getting these errors I tested this on 42 and 43 on a Windows 7
> machine and on a Chromebook.  Both worked fine!
>
> I will try to find another windows 8 machine to check if this is a problem
> with just my laptop or with Windows 8 in general.
>
> If anyone is wondering - yes, I rebooted the machine :-).  Also Opera and
> Firefox on the same machine are working just fine as did Chrome 41.
>
> On Tue, Mar 10, 2015 at 2:07 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Hi Brian,
>>
>> 41 always worked fine for me and still does.  It's 42 beta and 43 canary
>> that are not working.
>>
>> On Tue, Mar 10, 2015 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org,
>>> Just updated and tested.
>>>
>>> On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>> wrote:
>>>
>>>> Canary 43 does not work for me either.  Same symptoms both on my own FS
>>>> and on the Verto Demo site (with Opus).
>>>>
>>>> On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> They did all kinds of messed up stuff between 41 and 43.  43 is still
>>>>> somewhat broken as to video and using different resolutions.  I would not
>>>>> be surprised if 42 is complete garbage.
>>>>>
>>>>> On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:
>>>>>
>>>>>  M43 (canary) works fine (with opus).
>>>>>
>>>>> On 10.03.2015 16:08, Oleg Stolyar wrote:
>>>>>
>>>>> Hi guys,
>>>>>
>>>>>  I just got Chrome 42 beta and it does not work with FreeSWITCH.  I
>>>>> am getting this error on FS:
>>>>>
>>>>>  2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet
>>>>> not written
>>>>>
>>>>>  I tested it with the FS Verto test site and it also does not work.
>>>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>>>
>>>>>  Is this a known issue?
>>>>>
>>>>>  Of course 42 Beta also does not recognize Java for me, so hopefully
>>>>> it's just a crappy build and Google will fix it but my concern is that
>>>>> other WebRTC test sites still work, so should we look into this in case
>>>>> it's not fixed by the time Chrome 42 becomes stable?
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150310/fd77e52d/attachment.html 

From terry at digital-outpost.com  Wed Mar 11 02:30:51 2015
From: terry at digital-outpost.com (Terry Barnum)
Date: Tue, 10 Mar 2015 16:30:51 -0700
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
Message-ID: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>

When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor.

Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.) 

I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share?

Thanks,
-Terry

From ssinyagin at gmail.com  Wed Mar 11 02:41:11 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 11 Mar 2015 00:41:11 +0100
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
References: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
Message-ID: <CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>

your best choice would be an SMS gateway provider which allows you to
set arbitrary sender ID. Then you would buy a subscription there and
send your SMS'es via their API. You could then receive SMS
notifications with the sender ID equal to the original caller ID, and
your phone will automatically look up the caller in the phonebook.

Here in Switzerland I'm using http://www.inetworx.ch/  for this
purpose. I believe threre should also be providers in your area.

Then, making a hook in the mailer program is an easy task.
mod_voicemail in Confluence gives an example of such a script in
Python.



On Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
<terry at digital-outpost.com> wrote:
> When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor.
>
> Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.)
>
> I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share?
>
> Thanks,
> -Terry
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From sukithaj at gmail.com  Wed Mar 11 04:22:54 2015
From: sukithaj at gmail.com (sukitha jayasinghe)
Date: Wed, 11 Mar 2015 06:52:54 +0530
Subject: [Freeswitch-users] bridge calls in event socket app
Message-ID: <CAAwzf1=5dVjMKouOoL_ijh3Gk7L0EEL=m1trnWnbje-W0rwoCA@mail.gmail.com>

I want bridge two unanswered calls (in eventsocke) to listen to early
media. What is the mechanism for that?

Best Regards,
Sukitha
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/b73dbeca/attachment.html 

From jason at dickson.st  Wed Mar 11 07:38:01 2015
From: jason at dickson.st (Jason Lewis)
Date: Wed, 11 Mar 2015 15:38:01 +1100
Subject: [Freeswitch-users] How to configure G722 for internal and G729
 for	external calls
In-Reply-To: <049801d05af5$96f684d0$c4e38e70$@botecomm.com>
References: <54FE62BA.4070601@dickson.st>
	<049801d05af5$96f684d0$c4e38e70$@botecomm.com>
Message-ID: <54FFC6A9.2020502@dickson.st>

Hi Bote,

Appologies for the long log output. I'll use the pastebin in future.

I have purchased and installed 2 G729 licenses. I have tested and shown
that I can use them in my configuration.

Hi Bote,

Thanks for your guidance, and sorry for the long log post.

I eventually got it working by setting the absolute_code_string in the
dialplan.

    <action application="export" data="absolute_codec_string=G729"/>
 
Thanks to BKW and Drestreyf for help on that.

Jason



Bote Man wrote on 10/03/2015 4:46 PM:
> 1) Please don't post extensive log output to the mailing list. The developers much prefer that you use the FreeSWITCH pastebin and choose the FreeSWITCH log syntax highlighting:
> https://pastebin.freeswitch.org/
> and pay close attention to the instructions in the prompt for credentials that pops up.
>
> 2) I see:
> 2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
>
> Did you license G.729 codec? Because FS will need to transcode from the G.722 used internally to G.729 towards your carrier. My guess is that this is where your problem lies.
>
> With the vanilla configuration I believe that FS can agree on G.729 with the far end as long as it is merely passing the RTP stream through the switch untouched. Once FS needs to transcode it between G.729 and G.722 you need to fork over money for the number of simultaneous G.729 calls that you expect since it is a commercially restricted codec.
>
> Details on the Confluence wiki at:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>
> Of course, I could be totally wrong about this, but if you are down under then you'll be asleep when the everybody else wakes up so I figure I'd give it a stab to give you a head-start.
>
> Bote
>
>  
>
>
> -----Original Message-----
> From: Jason Lewis
> Sent: Monday, 09 March, 2015 23:19
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] How to configure G722 for internal and G729 for external calls
>
> hi,
>
> I'm trying to get freeswitch to use G722 for internal calls and G729 for
> external calls.
>
> I'm using vanilla.
>
> Am I missing something obvious here?
> .
> .
> .
> 2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4697 [zrtp_passthru] Setting a-leg inherit_codec=true
> 2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4700 [zrtp_passthru] Setting b-leg absolute_codec_string='G722 at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
> ...
> 2015-03-10 12:37:37.775786 [DEBUG] sofia_glue.c:1232 sofia/external/775 sending invite version: 1.4.15 -1 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1425927473 1425927474 IN IP4 aa.bbb.ccc.ddd
> s=FreeSWITCH
> c=IN IP4 aa.bbb.ccc.ddd
> t=0 0
> m=audio 23984 RTP/AVP 18 101 13
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6624 Remote SDP:
> v=0
> o=Sippy 2433467651418324771 2 IN IP4 202.85.243.105
> s=session
> t=0 0
> m=audio 10948 RTP/AVP 18 101
> c=IN IP4 202.85.243.53
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:2473 Set Codec sofia/external/775 G729/8000 20 ms 160 samples 8000 bits 1 channels
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_codec.c:111 sofia/external/775 Original read codec set to G729:18
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/external/775] 10.0.2.145 port 23984 -> 202.85.243.53 port 10948 codec: 18 ms: 20
> 2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 160 bytes per 20ms
> ...
> 2015-03-10 12:37:38.735794 [NOTICE] sofia.c:7475 Channel [sofia/external/775] has been answered
> 2015-03-10 12:37:38.735794 [DEBUG] switch_channel.c:3689 (sofia/external/775) Callstate Change DOWN -> ACTIVE
> 2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1001 at freeswitch.xyz.com.au G722/8000 20 ms 160 samples 64000 bits 1 channels
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at freeswitch.xyz.com.au Original read codec set to G722:9
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1001 at freeswitch.xyz.com.au] 10.0.2.145 port 20762 -> 10.0.2.129 port 11794 codec: 9 ms: 20
> 2015-03-10 12:37:38.755785 [DEBUG] switch_channel.c:3399 (sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change RINGING -> EARLY
> 2015-03-10 12:37:38.755785 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1001 at freeswitch.xyz.com.au:
> v=0
> o=FreeSWITCH 1425930696 1425930697 IN IP4 10.0.2.145
> s=FreeSWITCH
> c=IN IP4 10.0.2.145
> t=0 0
> m=audio 20762 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
>
>

-- 
Jason Lewis
http://emacstragic.net


-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 834 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/119b5bc4/attachment-0001.bin 

From covici at ccs.covici.com  Wed Mar 11 10:52:39 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 11 Mar 2015 03:52:39 -0400
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>
References: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
	<CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>
Message-ID: <8938.1426060359@ccs.covici.com>

There is provision in mod_voicemail to send a paging Email and so I just
send it as a text message to the phone's email address.

Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> your best choice would be an SMS gateway provider which allows you to
> set arbitrary sender ID. Then you would buy a subscription there and
> send your SMS'es via their API. You could then receive SMS
> notifications with the sender ID equal to the original caller ID, and
> your phone will automatically look up the caller in the phonebook.
> 
> Here in Switzerland I'm using http://www.inetworx.ch/  for this
> purpose. I believe threre should also be providers in your area.
> 
> Then, making a hook in the mailer program is an easy task.
> mod_voicemail in Confluence gives an example of such a script in
> Python.
> 
> 
> 
> On Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
> <terry at digital-outpost.com> wrote:
> > When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor.
> >
> > Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.)
> >
> > I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share?
> >
> > Thanks,
> > -Terry
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 adahary at gmail.com  Wed Mar 11 11:53:18 2015
From: adahary at gmail.com (Assaf Dahary)
Date: Wed, 11 Mar 2015 10:53:18 +0200
Subject: [Freeswitch-users] xml_cdr encoding issues
Message-ID: <0a7601d05bd8$d373f580$7a5be080$@gmail.com>

Hi,

I have installed a fresh FS on a DigitalOcean VPS (CentOS 6.5 64bit).
FreeSWITCH Version 1.5.15b+git~20150303T230402Z~86c68d906a~64bit (git
86c68d9 2015-03-03 23:04:02Z 64bit)

I'm using mod_cml_cdr to process call logs into DB using a PHP script ($xml
= simplexml_load_string($xml_string);) 

The problem is that the xml parsing is failing because the generated POST
XML CDR string received on my web is NOT url-encoded as should be.
I verified that encoding is enabled: <param name="encode" value="true"/>
(see full conf file below).

This un-encoded line from the posted xml CDR is one example for what is
causing the xml parsing error:
<sip_full_from>"99252627"
<sip:1234 at 11.11.11.11>;tag=y3Xr3g6HZZtBQ</sip_full_from>.

On other FS I get it encoded with no xml errors:
<sip_full_from>%3Csip%3A1234%40abc.com%3E%3Btag%3DnQGk6NEXS0yYrSeyIYCbrXmge7
80R7J4</sip_full_from>

I've reinstalled all PHP/XML/CURL related modules with no help.
I also tried to post to my other FS web server thinking that the local FS
web is decoding and then passing to the php xml script - still with no help.

Any tip/help/advise will be appreciated.

Regards

Assaf 


This is my xml_cdr.conf.xml file:

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
  <settings>
    <!-- the url to post to if blank web posting is disabled  -->
    <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->

    <!-- optional: credentials to send to web server -->
    <!--    <param name="cred" value="user:pass"/> -->

    <!-- the total number of retries (not counting the first 'try') to post
to webserver incase of failure -->
    <!-- <param name="retries" value="2"/> -->

    <!-- delay between retries in seconds, default is 5 seconds -->
    <!-- <param name="delay" value="1"/> -->

    <!-- optional: if not present we do not log every record to disk -->
    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank value will default to ${prefix}/logs/xml_cdr -->
    <param name="log-dir" value=""/>

    <!-- optional: if not present we do log the b leg -->
    <!-- true or false if we should create a cdr for the b leg of a call-->
    <param name="log-b-leg" value="false"/>
    
    <!-- optional: if not present, all filenames are the uuid of the call
-->
    <!-- true or false if a leg files are prefixed "a_" -->
    <param name="prefix-a-leg" value="true"/>

    <!-- encode the post data may be 'true' for url encoding, 'false' for no
encoding or 'base64' for base64 encoding -->
    <param name="encode" value="true"/>

    <!-- optional: set to true to disable Expect: 100-continue lighttpd
requires this setting --> 
    <!--<param name="disable-100-continue" value="true"/>--> 
    
    <!-- optional: full path to the error log dir for failed web posts if
not specified its the same as log-dir -->
    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
    <!-- <param name="err-log-dir" value="/tmp"/> -->

    <!-- optional: if enabled this will disable CA root certificate checks
by libcurl -->
    <!-- note: default value is disabled. only enable if you want this! -->
    <!-- <param name="ignore-cacert-check" value="true" /> -->

  </settings>
</configuration> 





From alexb at engagespark.com  Wed Mar 11 07:47:58 2015
From: alexb at engagespark.com (Alexander Baldeck)
Date: Wed, 11 Mar 2015 12:47:58 +0800
Subject: [Freeswitch-users] Caller ID not logged in CDR properly
Message-ID: <54FFC8FE.8030801@engagespark.com>

Hello,

we are using Freeswitch 1.5.12b+git~20140324T234505Z~4dd0a5848f~64bit
interfacing it with Plivo and experience an issue where our call detail
records list the destination_number twice when someone picks up a call
like so:

"Outbound Call","99901637777777","99901637777777","default","2015-02-28
13:15:00","2015-02-28 13:15:08","2015-02-28
13:16:32","92","84","NORMAL_CLEARING","24fdbb20-32a3-11e4-b883-e58f73531fe4","","","PCMA","PCMA"


If the call isn't picked up upon though, everything is alright and
caller_id_number is logged alongside destination_number like so:

"","0123456789","99901637777777","default","2015-02-28
13:11:00","","2015-02-28
13:12:00","60","0","NO_ANSWER","95b050f4-32a2-11e4-b87f-e58f73531fe4","","","",""

The used template in cdr_csv.xml looks like this and I made sure it is
being used:

<template
name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>

I'm at a loss what's going on here, has anyone got insight on where to look?


Best,
Alex
This email and its contents are confidential and may not be shared. If you are not the intended recipient of this email, please discard it. Please do not copy, forward, disclose, or use any part of it. Please delete it and all copies from your system and please notify the sender immediately by return e-mail. Thank you very much for your cooperation.


From mike at jerris.com  Wed Mar 11 15:14:03 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 11 Mar 2015 08:14:03 -0400
Subject: [Freeswitch-users] xml_cdr encoding issues
In-Reply-To: <0a7601d05bd8$d373f580$7a5be080$@gmail.com>
References: <0a7601d05bd8$d373f580$7a5be080$@gmail.com>
Message-ID: <4C779E2D-814D-45A7-A034-9BD5BEB36591@jerris.com>

https://freeswitch.org/jira/browse/FS-7258 <https://freeswitch.org/jira/browse/FS-7258>

> On Mar 11, 2015, at 4:53 AM, Assaf Dahary <adahary at gmail.com> wrote:
> 
> Hi,
> 
> I have installed a fresh FS on a DigitalOcean VPS (CentOS 6.5 64bit).
> FreeSWITCH Version 1.5.15b+git~20150303T230402Z~86c68d906a~64bit (git
> 86c68d9 2015-03-03 23:04:02Z 64bit)
> 
> I'm using mod_cml_cdr to process call logs into DB using a PHP script ($xml
> = simplexml_load_string($xml_string);) 
> 
> The problem is that the xml parsing is failing because the generated POST
> XML CDR string received on my web is NOT url-encoded as should be.
> I verified that encoding is enabled: <param name="encode" value="true"/>
> (see full conf file below).
> 
> This un-encoded line from the posted xml CDR is one example for what is
> causing the xml parsing error:
> <sip_full_from>"99252627"
> <sip:1234 at 11.11.11.11>;tag=y3Xr3g6HZZtBQ</sip_full_from>.
> 
> On other FS I get it encoded with no xml errors:
> <sip_full_from>%3Csip%3A1234%40abc.com%3E%3Btag%3DnQGk6NEXS0yYrSeyIYCbrXmge7
> 80R7J4</sip_full_from>
> 
> I've reinstalled all PHP/XML/CURL related modules with no help.
> I also tried to post to my other FS web server thinking that the local FS
> web is decoding and then passing to the php xml script - still with no help.
> 
> Any tip/help/advise will be appreciated.
> 
> Regards
> 
> Assaf 
> 
> 
> This is my xml_cdr.conf.xml file:
> 
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>  <settings>
>    <!-- the url to post to if blank web posting is disabled  -->
>    <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
> 
>    <!-- optional: credentials to send to web server -->
>    <!--    <param name="cred" value="user:pass"/> -->
> 
>    <!-- the total number of retries (not counting the first 'try') to post
> to webserver incase of failure -->
>    <!-- <param name="retries" value="2"/> -->
> 
>    <!-- delay between retries in seconds, default is 5 seconds -->
>    <!-- <param name="delay" value="1"/> -->
> 
>    <!-- optional: if not present we do not log every record to disk -->
>    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
> a blank value will default to ${prefix}/logs/xml_cdr -->
>    <param name="log-dir" value=""/>
> 
>    <!-- optional: if not present we do log the b leg -->
>    <!-- true or false if we should create a cdr for the b leg of a call-->
>    <param name="log-b-leg" value="false"/>
> 
>    <!-- optional: if not present, all filenames are the uuid of the call
> -->
>    <!-- true or false if a leg files are prefixed "a_" -->
>    <param name="prefix-a-leg" value="true"/>
> 
>    <!-- encode the post data may be 'true' for url encoding, 'false' for no
> encoding or 'base64' for base64 encoding -->
>    <param name="encode" value="true"/>
> 
>    <!-- optional: set to true to disable Expect: 100-continue lighttpd
> requires this setting --> 
>    <!--<param name="disable-100-continue" value="true"/>--> 
> 
>    <!-- optional: full path to the error log dir for failed web posts if
> not specified its the same as log-dir -->
>    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
> a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
>    <!-- <param name="err-log-dir" value="/tmp"/> -->
> 
>    <!-- optional: if enabled this will disable CA root certificate checks
> by libcurl -->
>    <!-- note: default value is disabled. only enable if you want this! -->
>    <!-- <param name="ignore-cacert-check" value="true" /> -->
> 
>  </settings>
> </configuration> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/7fb5dd70/attachment.html 

From mike at jerris.com  Wed Mar 11 15:19:46 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 11 Mar 2015 08:19:46 -0400
Subject: [Freeswitch-users] Caller ID not logged in CDR properly
In-Reply-To: <54FFC8FE.8030801@engagespark.com>
References: <54FFC8FE.8030801@engagespark.com>
Message-ID: <6DABC8CD-C715-454B-9019-323D5D682563@jerris.com>

Unfortunately we are unable to remove your email from the mailing list archives.  Please make sure in the future not to send private and confidential emails to the mailing list.

> On Mar 11, 2015, at 12:47 AM, Alexander Baldeck <alexb at engagespark.com> wrote:
> 
> 
> Best,
> Alex
> This email and its contents are confidential and may not be shared. If you are not the intended recipient of this email, please discard it. Please do not copy, forward, disclose, or use any part of it. Please delete it and all copies from your system and please notify the sender immediately by return e-mail. Thank you very much for your cooperation.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/2f6db203/attachment-0001.html 

From adahary at gmail.com  Wed Mar 11 16:06:53 2015
From: adahary at gmail.com (Assaf Dahary)
Date: Wed, 11 Mar 2015 15:06:53 +0200
Subject: [Freeswitch-users] xml_cdr encoding issues
In-Reply-To: <4C779E2D-814D-45A7-A034-9BD5BEB36591@jerris.com>
References: <0a7601d05bd8$d373f580$7a5be080$@gmail.com>
	<4C779E2D-814D-45A7-A034-9BD5BEB36591@jerris.com>
Message-ID: <0aa701d05bfc$40ddac60$c2990520$@gmail.com>

Michael,

 

I saw that there is a fix/patch for this issue.

But how I add/compile it (sorry - not much know HOWTO) ?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Wednesday, March 11, 2015 2:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] xml_cdr encoding issues

 

https://freeswitch.org/jira/browse/FS-7258

 

On Mar 11, 2015, at 4:53 AM, Assaf Dahary <adahary at gmail.com> wrote:

 

Hi,

I have installed a fresh FS on a DigitalOcean VPS (CentOS 6.5 64bit).
FreeSWITCH Version 1.5.15b+git~20150303T230402Z~86c68d906a~64bit (git
86c68d9 2015-03-03 23:04:02Z 64bit)

I'm using mod_cml_cdr to process call logs into DB using a PHP script ($xml
= simplexml_load_string($xml_string);) 

The problem is that the xml parsing is failing because the generated POST
XML CDR string received on my web is NOT url-encoded as should be.
I verified that encoding is enabled: <param name="encode" value="true"/>
(see full conf file below).

This un-encoded line from the posted xml CDR is one example for what is
causing the xml parsing error:
<sip_full_from>"99252627"
<sip:1234 at 11.11.11.11>;tag=y3Xr3g6HZZtBQ</sip_full_from>.

On other FS I get it encoded with no xml errors:
<sip_full_from>%3Csip%3A1234%40abc.com%3E%3Btag%3DnQGk6NEXS0yYrSeyIYCbrXmge7
80R7J4</sip_full_from>

I've reinstalled all PHP/XML/CURL related modules with no help.
I also tried to post to my other FS web server thinking that the local FS
web is decoding and then passing to the php xml script - still with no help.

Any tip/help/advise will be appreciated.

Regards

Assaf 


This is my xml_cdr.conf.xml file:

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
 <settings>
   <!-- the url to post to if blank web posting is disabled  -->
   <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->

   <!-- optional: credentials to send to web server -->
   <!--    <param name="cred" value="user:pass"/> -->

   <!-- the total number of retries (not counting the first 'try') to post
to webserver incase of failure -->
   <!-- <param name="retries" value="2"/> -->

   <!-- delay between retries in seconds, default is 5 seconds -->
   <!-- <param name="delay" value="1"/> -->

   <!-- optional: if not present we do not log every record to disk -->
   <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank value will default to ${prefix}/logs/xml_cdr -->
   <param name="log-dir" value=""/>

   <!-- optional: if not present we do log the b leg -->
   <!-- true or false if we should create a cdr for the b leg of a call-->
   <param name="log-b-leg" value="false"/>

   <!-- optional: if not present, all filenames are the uuid of the call
-->
   <!-- true or false if a leg files are prefixed "a_" -->
   <param name="prefix-a-leg" value="true"/>

   <!-- encode the post data may be 'true' for url encoding, 'false' for no
encoding or 'base64' for base64 encoding -->
   <param name="encode" value="true"/>

   <!-- optional: set to true to disable Expect: 100-continue lighttpd
requires this setting --> 
   <!--<param name="disable-100-continue" value="true"/>--> 

   <!-- optional: full path to the error log dir for failed web posts if
not specified its the same as log-dir -->
   <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
   <!-- <param name="err-log-dir" value="/tmp"/> -->

   <!-- optional: if enabled this will disable CA root certificate checks
by libcurl -->
   <!-- note: default value is disabled. only enable if you want this! -->
   <!-- <param name="ignore-cacert-check" value="true" /> -->

 </settings>
</configuration> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/3a9b0e16/attachment.html 

From raphael.lechner at gmail.com  Wed Mar 11 16:25:09 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Wed, 11 Mar 2015 14:25:09 +0100
Subject: [Freeswitch-users] blf status stress testing
Message-ID: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>

Hi,

We have the problem that on different installations the blf keys in combination with yealink phones don?t always show the correct status, sometimes ringing or busy is not shown.
The installations are relative small from 3 to 80 Phones.
We still have the V72 Version of firmware on the phones and not all installations have the current 1.4.15 version of FreeSWITCH.
I?m trying to build some kind of stress test environment to try to reproduce the issue do identify if the problem is resolved with the new phone firmware and FreeSWITCH version or to identify what is going wrong.

Can someone recommend tools for stress testing the blf keys?

Thank you,
Raphael

From ssinyagin at gmail.com  Wed Mar 11 16:33:37 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 11 Mar 2015 14:33:37 +0100
Subject: [Freeswitch-users] blf status stress testing
In-Reply-To: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
References: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
Message-ID: <CACOYK=hDB7g443WzMKebHnmLrqvXAtx1GXA-jQU3nxWv9PVb2Q@mail.gmail.com>

sipsak, probably?
http://thr3ads.net/asterisk-users/2008/09/1417638-Polycom-BLF-multiple-buddies

you can also use a second FreeSWITCH machine which would register at
your server with multiple accounts and generate calls from it.
Here's my call generator, just in case:
https://github.com/voxserv/freeswitch-perf-dialer



On Wed, Mar 11, 2015 at 2:25 PM, Raphael Lechner
<raphael.lechner at gmail.com> wrote:
> Hi,
>
> We have the problem that on different installations the blf keys in combination with yealink phones don?t always show the correct status, sometimes ringing or busy is not shown.
> The installations are relative small from 3 to 80 Phones.
> We still have the V72 Version of firmware on the phones and not all installations have the current 1.4.15 version of FreeSWITCH.
> I?m trying to build some kind of stress test environment to try to reproduce the issue do identify if the problem is resolved with the new phone firmware and FreeSWITCH version or to identify what is going wrong.
>
> Can someone recommend tools for stress testing the blf keys?
>
> Thank you,
> Raphael
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From terry at digital-outpost.com  Wed Mar 11 20:46:55 2015
From: terry at digital-outpost.com (Terry Barnum)
Date: Wed, 11 Mar 2015 10:46:55 -0700
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <8938.1426060359@ccs.covici.com>
References: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
	<CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>
	<8938.1426060359@ccs.covici.com>
Message-ID: <6A0C6E41-561F-4C12-859D-12C6B0FCD75E@digital-outpost.com>

Thank you John and Stanislav. I'll have a look at mod_voicemail.

-Terry

> On Mar 11, 2015, at 12:52 AM, covici at ccs.covici.com wrote:
> 
> There is provision in mod_voicemail to send a paging Email and so I just
> send it as a text message to the phone's email address.
> 
> Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
>> your best choice would be an SMS gateway provider which allows you to
>> set arbitrary sender ID. Then you would buy a subscription there and
>> send your SMS'es via their API. You could then receive SMS
>> notifications with the sender ID equal to the original caller ID, and
>> your phone will automatically look up the caller in the phonebook.
>> 
>> Here in Switzerland I'm using http://www.inetworx.ch/  for this
>> purpose. I believe threre should also be providers in your area.
>> 
>> Then, making a hook in the mailer program is an easy task.
>> mod_voicemail in Confluence gives an example of such a script in
>> Python.
>> 
>> 
>> 
>> On Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
>> <terry at digital-outpost.com> wrote:
>>> When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor.
>>> 
>>> Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.)
>>> 
>>> I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share?
>>> 
>>> Thanks,
>>> -Terry
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From shabbirabbasi92 at gmail.com  Wed Mar 11 21:28:39 2015
From: shabbirabbasi92 at gmail.com (Shabbir abbasi)
Date: Wed, 11 Mar 2015 23:28:39 +0500
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <6A0C6E41-561F-4C12-859D-12C6B0FCD75E@digital-outpost.com>
References: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
	<CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>
	<8938.1426060359@ccs.covici.com>
	<6A0C6E41-561F-4C12-859D-12C6B0FCD75E@digital-outpost.com>
Message-ID: <CAGpk5BQP+tbnc66_u6s7-4bREbvc1XA-2LUvkNnaD=2B1KYjKg@mail.gmail.com>

i think your solution is in  mod_gsmopen  kindly read in wiki

On Wed, Mar 11, 2015 at 10:46 PM, Terry Barnum <terry at digital-outpost.com>
wrote:

> Thank you John and Stanislav. I'll have a look at mod_voicemail.
>
> -Terry
>
> > On Mar 11, 2015, at 12:52 AM, covici at ccs.covici.com wrote:
> >
> > There is provision in mod_voicemail to send a paging Email and so I just
> > send it as a text message to the phone's email address.
> >
> > Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> >
> >> your best choice would be an SMS gateway provider which allows you to
> >> set arbitrary sender ID. Then you would buy a subscription there and
> >> send your SMS'es via their API. You could then receive SMS
> >> notifications with the sender ID equal to the original caller ID, and
> >> your phone will automatically look up the caller in the phonebook.
> >>
> >> Here in Switzerland I'm using http://www.inetworx.ch/  for this
> >> purpose. I believe threre should also be providers in your area.
> >>
> >> Then, making a hook in the mailer program is an easy task.
> >> mod_voicemail in Confluence gives an example of such a script in
> >> Python.
> >>
> >>
> >>
> >> On Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
> >> <terry at digital-outpost.com> wrote:
> >>> When someone leaves a voicemail at the house, FS + FusionPBX
> successfully emails the Google transcription along with the audio
> attachment of the voicemail. It's working well. High Spousal Approval
> Factor.
> >>>
> >>> Now I would like to send a SMS text message notification via cellular
> to our mobile phones when a voicemail is left. Possible? I don't
> necessarily need the audio of the voicemail just a notification with the
> caller id in the SMS. (Though if it's easy I'll try it.)
> >>>
> >>> I've looked at the mod_sms wiki docs but all the examples appear to be
> between FS clients. How to send out to a cellular mobile number? Anyone
> have working examples they can share?
> >>>
> >>> Thanks,
> >>> -Terry
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/80f4a6da/attachment-0001.html 

From vladget at gmail.com  Wed Mar 11 23:14:19 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Wed, 11 Mar 2015 22:14:19 +0200
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAEJMVkD=oy1FRLB5mKYBGe4S3L5eAFSFEP-9_4eZmPj80q3wBQ@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
	<CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
	<CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>
	<CAMxcU5Jk2=ROwCUNPzjmGHRRikCWdiX-=zQ--NrRhpTDj11erg@mail.gmail.com>
	<CAEJMVkD=oy1FRLB5mKYBGe4S3L5eAFSFEP-9_4eZmPj80q3wBQ@mail.gmail.com>
Message-ID: <CAMxcU5Ls8eBW0b2J7QKFcTe43uNf-QJk6SPyPhQbE+tDb5pWUw@mail.gmail.com>

ok, just debug log or mod_cdr_csv debug?

On Wed, Mar 4, 2015 at 12:00 AM, Brian West <brian at freeswitch.org> wrote:

> Without looking a the logs I can only guess that something is triggering
> it...happen to have a debug log of this?
>
> On Tue, Mar 3, 2015 at 4:30 PM, Vladimir Getmanshchuk <vladget at gmail.com>
> wrote:
>
>> As I have said before configs on both FS boxes are same. So Proxy-Media
>> enabled on both FS boxes.
>>
>> On Tue, Mar 3, 2015 at 7:54 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> If you enable proxy media it will say proxy too....
>>>
>>>
>>> On Tuesday, March 3, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
>>> wrote:
>>>
>>>> Same traffic balanced between these two FS boxes and all CDRs from
>>>> 1.4.5 came with PROXY, but all CDRs from 1.4.5 came with real codec were in
>>>> streams.
>>>>
>>>>
>>>>
>>>> On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>>>> ZRTP hash in the sdp will cause it to toggle on too!
>>>>>
>>>>>
>>>>> On Saturday, February 28, 2015, Vladimir Getmanshchuk <
>>>>> vladget at gmail.com> wrote:
>>>>>
>>>>>> Bote,
>>>>>> When I said identical configuration I mean files at FS configuration
>>>>>> directory.
>>>>>> G.729 license? No, I use proxy-media mode with no transcoding.
>>>>>>
>>>>>> Brian,
>>>>>> Both FS boxes configured for proxing media:
>>>>>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>>>>>> /sip_profiles/internal.xml
>>>>>> <param name="inbound-proxy-media" value="true"/>
>>>>>>
>>>>>> I do not understand why FS version 1.4.15 trying to hide actual
>>>>>> read/write codecs and change it by "PROXY"?
>>>>>>
>>>>>> Thank you.
>>>>>>
>>>>>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org>
>>>>>> wrote:
>>>>>>
>>>>>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>>>>>
>>>>>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>>
>>>>>>>> I would say that these are not "absolutely" identical. As the
>>>>>>>> FreeSWITCH
>>>>>>>> development team never sleeps it is likely that there are
>>>>>>>> differences in the
>>>>>>>> code that you now see. The first thing is to bring both machines up
>>>>>>>> to the
>>>>>>>> same release before comparing behaviors.
>>>>>>>>
>>>>>>>> Another suggestion is to confirm your G.729 license and
>>>>>>>> configuration, if
>>>>>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>>>>>> file(s)
>>>>>>>> in the correct locations and the other machine does not?
>>>>>>>>
>>>>>>>> Hope this helps.
>>>>>>>>
>>>>>>>> Bote
>>>>>>>>
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: Vladimir Getmanshchuk
>>>>>>>> Sent: Friday, 27 February, 2015 07:37
>>>>>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>>>>>> different FS
>>>>>>>> versions...
>>>>>>>>
>>>>>>>> Hello Everyone!
>>>>>>>>
>>>>>>>> I have two installations of FS with absolutely identical
>>>>>>>> configurations.
>>>>>>>> Both has SIP profiles with proxy-media enabled.
>>>>>>>>
>>>>>>>> But on
>>>>>>>> freeswitch at internal> version
>>>>>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>>>
>>>>>>>> I have values in read_codec/write_codec variables at CDRs:
>>>>>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>>>>>
>>>>>>>> but on another one
>>>>>>>> freeswitch at internal> version
>>>>>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>>
>>>>>>>> I have:
>>>>>>>> "read_codec":"G729","write_codec":"G729",
>>>>>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>>>>>
>>>>>>>> So Why?
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Yours sincerely,
>>>>>>>> Vladimir Getmanshchuk
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> *Brian West*
>>>>>>> brian at freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>> http://www.freeswitchbook.com
>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>
>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Yours sincerely,
>>>>>> Vladimir Getmanshchuk
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Yours sincerely,
>>>> Vladimir Getmanshchuk
>>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Yours sincerely,
>> Vladimir Getmanshchuk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/b3217f9d/attachment.html 

From bote_radio at botecomm.com  Wed Mar 11 23:30:32 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 11 Mar 2015 16:30:32 -0400
Subject: [Freeswitch-users] Chrome 42 Beta (on Windows) not working
	with	FreeSWITCH
In-Reply-To: <CAC8JMOLKDfPn4sVJbBA=XCbZMQ2WZdYZNJt=4mrw3J5BAFq1UQ@mail.gmail.com>
References: <CAC8JMOKWqZs5AzkjYayvqE2V+ZmZem0694zG6RK0C-z2mb-3cQ@mail.gmail.com>	<54FF45E5.6020203@virtues.net>	<014F5CCE-B94D-4EB0-8CEB-83E8DA8F8BF7@jerris.com>	<CAC8JMOJD=EnxvXrDHtOG-OXf1gU=1kLrub5U7-ykedyRiwkAkg@mail.gmail.com>	<CAEJMVkB7YxkUSEcRLTTLiNkz7D0OQCHRhRm-2kE=u_VU5TNNQg@mail.gmail.com>	<CAC8JMOLf9AdK5LPiHa7kO2y-ucC3m04BLArorc8nQdRnaViGYw@mail.gmail.com>	<CAC8JMO+sDeoR50EkNRNmD2gm0-s-Q51xGDse5RqVn+MduSRk8A@mail.gmail.com>
	<CAC8JMOLKDfPn4sVJbBA=XCbZMQ2WZdYZNJt=4mrw3J5BAFq1UQ@mail.gmail.com>
Message-ID: <06e201d05c3a$390a9dc0$ab1fd940$@botecomm.com>

On the weekly ClueCon conference call today neither Firefox 36.0.1 nor Chrome 41 would provide me receive conference audio. I called in on a physical phone but also had the Verto caller display up in Chrome and it repeatedly disconnected after about 8 minutes with the cause ?DESTINATION_OUT_OF_ORDER? which it very will might be J

 

This happened both with and without ?Use STUN? checked.

 

Ken offered that it might be that the FS server running on Big Blue Button needs to be refreshed as the browser code might have creeped away from spec.

 

Bote

 

 

From: Oleg Stolyar
Sent: Tuesday, 10 March, 2015 19:09
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Chrome 42 Beta (on Windows) not working with FreeSWITCH

 

Last update:  Believe it or not this was solved by getting the latest Windows 8 updates.  

 

Sorry about the false alarm!

 

On Tue, Mar 10, 2015 at 3:39 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

Update on this:

 

My laptop on which I got these errors is running Windows 8 (don't ask why :-)).  

 

After getting these errors I tested this on 42 and 43 on a Windows 7 machine and on a Chromebook.  Both worked fine!

 

I will try to find another windows 8 machine to check if this is a problem with just my laptop or with Windows 8 in general.  

 

If anyone is wondering - yes, I rebooted the machine :-).  Also Opera and Firefox on the same machine are working just fine as did Chrome 41.

 

On Tue, Mar 10, 2015 at 2:07 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

Hi Brian,

 

41 always worked fine for me and still does.  It's 42 beta and 43 canary that are not working.

 

On Tue, Mar 10, 2015 at 1:54 PM, Brian West <brian at freeswitch.org> wrote:

Version 41.0.2272.89 (64-bit) works fine with webrtc.freeswitch.org, Just updated and tested.

 

On Tue, Mar 10, 2015 at 3:06 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

Canary 43 does not work for me either.  Same symptoms both on my own FS and on the Verto Demo site (with Opus).

 

On Tue, Mar 10, 2015 at 12:32 PM, Michael Jerris <mike at jerris.com> wrote:

They did all kinds of messed up stuff between 41 and 43.  43 is still somewhat broken as to video and using different resolutions.  I would not be surprised if 42 is complete garbage.

 

On Mar 10, 2015, at 3:28 PM, Thomas Auge <auge at virtues.net> wrote:

 

M43 (canary) works fine (with opus).

On 10.03.2015 16:08, Oleg Stolyar wrote:

Hi guys, 

 

I just got Chrome 42 beta and it does not work with FreeSWITCH.  I am getting this error on FS:

 

2015-03-10 19:04:12.272683 [ERR] switch_rtp.c:2800 audio DTLS packet not written

 

I tested it with the FS Verto test site and it also does not work.

https://webrtc.freeswitch.org/verto/index.html#page-main

 

Is this a known issue?

 

Of course 42 Beta also does not recognize Java for me, so hopefully it's just a crappy build and Google will fix it but my concern is that other WebRTC test sites still work, so should we look into this in case it's not fixed by the time Chrome 42 becomes stable?

 

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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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

 


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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





 

-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 <tel:%2B19184209001>  | F:+19184209002 <tel:%2B19184209002>  | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/4a69dc7b/attachment-0001.html 

From bote_radio at botecomm.com  Thu Mar 12 00:22:20 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 11 Mar 2015 17:22:20 -0400
Subject: [Freeswitch-users] blf status stress testing
In-Reply-To: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
References: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
Message-ID: <070901d05c41$75bc03b0$61340b10$@botecomm.com>

The core developers caution us that BLF subscriptions put a good amount of stress on the FreeSWITCH server, so 80 phones might be too many for your installation, I don't know. 

There is also the consideration that different telephone firmware versions behave differently, so it should help to try other phones as a test to see how they perform with BLF displays.

Bote


-----Original Message-----
From: Raphael Lechner
Sent: Wednesday, 11 March, 2015 09:25
Subject: [Freeswitch-users] blf status stress testing

Hi,

We have the problem that on different installations the blf keys in combination with yealink phones don?t always show the correct status, sometimes ringing or busy is not shown.
The installations are relative small from 3 to 80 Phones.
We still have the V72 Version of firmware on the phones and not all installations have the current 1.4.15 version of FreeSWITCH.
I?m trying to build some kind of stress test environment to try to reproduce the issue do identify if the problem is resolved with the new phone firmware and FreeSWITCH version or to identify what is going wrong.

Can someone recommend tools for stress testing the blf keys?

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




From ssinyagin at gmail.com  Thu Mar 12 01:38:42 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 11 Mar 2015 23:38:42 +0100
Subject: [Freeswitch-users] blf status stress testing
In-Reply-To: <070901d05c41$75bc03b0$61340b10$@botecomm.com>
References: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
	<070901d05c41$75bc03b0$61340b10$@botecomm.com>
Message-ID: <CACOYK=hStetGB5AyJFWFjAjnuimmkDaMExmQNDN+8a2OnDRCbg@mail.gmail.com>

back to the question of stress testing,

FreeSWITCH can act as a subscription client, so you can have a second
FreeSWITCH machine to register on your tested PBX and subscribe to BLF
events. Then you can easily see if 80 clients is a lot or not really.
https://wiki.freeswitch.org/wiki/Presence



On Wed, Mar 11, 2015 at 10:22 PM, Bote Man <bote_radio at botecomm.com> wrote:
> The core developers caution us that BLF subscriptions put a good amount of stress on the FreeSWITCH server, so 80 phones might be too many for your installation, I don't know.
>
> There is also the consideration that different telephone firmware versions behave differently, so it should help to try other phones as a test to see how they perform with BLF displays.
>
> Bote
>
>
> -----Original Message-----
> From: Raphael Lechner
> Sent: Wednesday, 11 March, 2015 09:25
> Subject: [Freeswitch-users] blf status stress testing
>
> Hi,
>
> We have the problem that on different installations the blf keys in combination with yealink phones don?t always show the correct status, sometimes ringing or busy is not shown.
> The installations are relative small from 3 to 80 Phones.
> We still have the V72 Version of firmware on the phones and not all installations have the current 1.4.15 version of FreeSWITCH.
> I?m trying to build some kind of stress test environment to try to reproduce the issue do identify if the problem is resolved with the new phone firmware and FreeSWITCH version or to identify what is going wrong.
>
> Can someone recommend tools for stress testing the blf keys?
>
> Thank you,
> Raphael
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From raphael.lechner at gmail.com  Thu Mar 12 13:17:58 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Thu, 12 Mar 2015 11:17:58 +0100
Subject: [Freeswitch-users] blf status stress testing
In-Reply-To: <070901d05c41$75bc03b0$61340b10$@botecomm.com>
References: <092BA580-6667-41AD-BB7D-1D3DEF700EE9@gmail.com>
	<070901d05c41$75bc03b0$61340b10$@botecomm.com>
Message-ID: <02EE3EC6-947E-4426-B8A9-584371E9508F@gmail.com>

ok with the 80 phones is for sure a lot of stress for the FreeSWITCH server, but we had the same issue with only 5 phone, so probably I think it?s more
a phone software/hardware problem.We will upgrade the 80 Phones installation next week to the 1.4.15 and in an next step the yealink firmware to the latest 73 version.

Raphael
  

> On 11 Mar 2015, at 22:22, Bote Man <bote_radio at botecomm.com> wrote:
> 
> The core developers caution us that BLF subscriptions put a good amount of stress on the FreeSWITCH server, so 80 phones might be too many for your installation, I don't know. 
> 
> There is also the consideration that different telephone firmware versions behave differently, so it should help to try other phones as a test to see how they perform with BLF displays.
> 
> Bote
> 
> 
> -----Original Message-----
> From: Raphael Lechner
> Sent: Wednesday, 11 March, 2015 09:25
> Subject: [Freeswitch-users] blf status stress testing
> 
> Hi,
> 
> We have the problem that on different installations the blf keys in combination with yealink phones don?t always show the correct status, sometimes ringing or busy is not shown.
> The installations are relative small from 3 to 80 Phones.
> We still have the V72 Version of firmware on the phones and not all installations have the current 1.4.15 version of FreeSWITCH.
> I?m trying to build some kind of stress test environment to try to reproduce the issue do identify if the problem is resolved with the new phone firmware and FreeSWITCH version or to identify what is going wrong.
> 
> Can someone recommend tools for stress testing the blf keys?
> 
> Thank you,
> Raphael
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From osblinnikov at gmail.com  Thu Mar 12 15:07:45 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Thu, 12 Mar 2015 13:07:45 +0100
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
	<CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
	<29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>
Message-ID: <CA+M05dCo87sc3kybYtz8u=PMdBAYZ2KNtMCWEECp6aZ0_E7bsw@mail.gmail.com>

Hi,

Unfortunately not everything runs smoothly. I run FreeSWITCH Version
1.5.15b+git~20150203T210457Z~4174fb9cbe~64bit (git 4174fb9 2015-02-03
21:04:57Z 64bit) with the default configuration + webrtc module + tweaked
bridge application for 1010 - 1019 extensions where I added
ignore_early_media=true because of chrome troubles with pranswer and
media_webrtc=true because one of my clients is actually JAIN SIP  without
WebSockets.

Now I call from Chrome to my Android app. In Android app I receive modified
SDP from FreeSwitch and all the media traffic goes though FreeSwitch. I
have the audio and video in both directions.

But when I createOffer in the Android application and send it to Chrome the
media is not flowing in any directions. In case I set "<action
application="set" data="bypass_media=true"/>" media starts flowing.

PS. May be it's irrelevant  but the only strange thing I noticed is that
Android App produces fingerprint in sha-1 then FreeSwitch changes it to
sha-256 and sends to Chrome. Chrome responds with sha-256 then FreeSwitch
modifies it back to sha-1.

I don't know, may be I forget about some other magic options?

PS: SDPs are in the attachment

On Fri, Mar 6, 2015 at 4:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Always nice to hear that we are magic!
>
> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
>
> thank you very much Michael, it magically works.
>
> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> you need to tell freeswitch to send a webrtc compatible SDP.
>>
>> https://wiki.freeswitch.org/wiki/Variable_media_webrtc
>>
>>
>> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
>>
>> Hi,
>>
>> I've made a simple Android Java application utilizing JAIN SIP,
>> webrtc.org android library and connected to FreeSwitch via UDP.
>>
>> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client
>> this SDP is not managed well by FreeSwitch for establishment WebRTC
>> PeerConnection.
>>
>> When I call `peerConnection.setRemoteDescription(new SDPObserver(),
>> sdp);` in my Android Application with the SDP from FreeSwitch I get:
>>
>> "onSetFailure Failed to set remote offer sdp: Called with SDP without
>> DTLS fingerprint."
>>
>> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/)
>> and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure
>> audio flow.
>>
>> Here is initial SDP from Chrome (http://tryit.jssip.net/):
>>
>> v=0
>> o=- 6887715720880489867 2 IN IP4 127.0.0.1
>> s=-
>> t=0 0
>> a=group:BUNDLE audio video
>> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
>> c=IN IP4 192.168.122.1
>> a=rtcp:38359 IN IP4 192.168.122.1
>> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
>> generation 0
>> .......
>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
>> active generation 0
>> a=ice-ufrag:bwrCv9yS8rCY12Az
>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>> a=ice-options:google-ice
>> a=*fingerprint*:sha-256
>> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>> a=setup:actpass
>> a=mid:audio
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=sendrecv
>> a=rtcp-mux
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=maxptime:60
>> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
>> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>> 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>> m=video 38359 RTP/SAVPF 100 116 117 96
>> c=IN IP4 192.168.122.1
>> a=rtcp:38359 IN IP4 192.168.122.1
>> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
>> generation 0
>> ............
>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
>> active generation 0
>> a=ice-ufrag:bwrCv9yS8rCY12Az
>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>> a=ice-options:google-ice
>> a=*fingerprint*:sha-256
>> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>> a=setup:actpass
>> a=mid:video
>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=recvonly
>> a=rtcp-mux
>> a=rtpmap:100 VP8/90000
>> a=rtcp-fb:100 ccm fir
>> a=rtcp-fb:100 nack
>> a=rtcp-fb:100 nack pli
>> a=rtcp-fb:100 goog-remb
>> a=rtpmap:116 red/90000
>> a=rtpmap:117 ulpfec/90000
>> a=rtpmap:96 rtx/90000
>> a=fmtp:96 apt=100
>>
>>
>> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>>
>> v=0
>> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
>> s=FreeSWITCH
>> c=IN IP4 192.168.131.253
>> t=0 0
>> m=audio 16390 RTP/AVP 111 0 8 101 13
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> m=video 16388 RTP/AVP 100
>> a=rtpmap:100 VP8/90000
>>
>>
>> I suppose that FreeSwitch wants to see WebRTC connection only on the
>> WebSocket ports and it doesn't know that my UDP client is actually WebRTC
>> client.
>>
>> So I'm wondering if it possible to connect SIP client to the WebSocket
>> port via TCP using standard SIP client and never upgrade connection to
>> WebSocket?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Oleg Blinnikov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/a2e2562d/attachment-0001.html 
-------------- next part --------------
#Offer from Android (by webrtc.org library):
v=0
o=- 5728876668167601575 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS ARDAMS
m=audio 9 RTP/SAVPF 111 103 9 102 0 8 106 105 13 127 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:nEhT7P/F24BeBDoB
a=ice-pwd:jnVtYxo22sUqQ5QnSyyfZdsY
a=fingerprint:sha-1 0C:49:3B:CA:9E:CD:43:B7:40:20:AC:0A:4D:E1:E7:44:87:92:E5:E1
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:127 red/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3730323322 cname:I5IDYTbgReozOmpK
a=ssrc:3730323322 msid:ARDAMS ARDAMSa0
a=ssrc:3730323322 mslabel:ARDAMS
a=ssrc:3730323322 label:ARDAMSa0
a=candidate:1871046784 1 udp 2122129151 192.168.128.249 51069 typ host generation 0
a=candidate:1871046784 2 udp 2122129151 192.168.128.249 51069 typ host generation 0
a=candidate:554046576 1 tcp 1518149375 192.168.128.249 53912 typ host tcptype passive generation 0
a=candidate:554046576 2 tcp 1518149375 192.168.128.249 53912 typ host tcptype passive generation 0
m=video 9 RTP/SAVPF 100 116 117 96
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:nEhT7P/F24BeBDoB
a=ice-pwd:jnVtYxo22sUqQ5QnSyyfZdsY
a=fingerprint:sha-1 0C:49:3B:CA:9E:CD:43:B7:40:20:AC:0A:4D:E1:E7:44:87:92:E5:E1
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 4043084481 952591881
a=ssrc:4043084481 cname:I5IDYTbgReozOmpK
a=ssrc:4043084481 msid:ARDAMS ARDAMSv0
a=ssrc:4043084481 mslabel:ARDAMS
a=ssrc:4043084481 label:ARDAMSv0
a=ssrc:952591881 cname:I5IDYTbgReozOmpK
a=ssrc:952591881 msid:ARDAMS ARDAMSv0
a=ssrc:952591881 mslabel:ARDAMS
a=ssrc:952591881 label:ARDAMSv0
a=candidate:1871046784 1 udp 2122129151 192.168.128.249 51069 typ host generation 0
a=candidate:1871046784 2 udp 2122129151 192.168.128.249 51069 typ host generation 0
a=candidate:554046576 1 tcp 1518149375 192.168.128.249 53912 typ host tcptype passive generation 0
a=candidate:554046576 2 tcp 1518149375 192.168.128.249 53912 typ host tcptype passive generation 0




#Offer received in Chrome from FreeSwitch:
v=0
o=FreeSWITCH 1426145176 1426145177 IN IP4 192.168.131.253
s=FreeSWITCH
c=IN IP4 192.168.131.253
t=0 0
a=msid-semantic: WMS cmSQsnGZmAjyQE9LSdNN71eThgHwKa1N
m=audio 16386 RTP/SAVPF 111 9 0 8 101 13
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256 34:46:35:C3:98:97:79:67:E8:B9:5F:FF:82:C4:EE:4E:2D:CA:DE:46:BD:B2:6D:D5:A6:07:C9:26:5B:3F:AE:10
a=rtcp-mux
a=rtcp:16386 IN IP4 192.168.131.253
a=ssrc:1024766842 cname:mKuFX0u9qEy78yIQ
a=ssrc:1024766842 msid:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1N a0
a=ssrc:1024766842 mslabel:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1N
a=ssrc:1024766842 label:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1Na0
a=ice-ufrag:CluAdzjkFQHfRzT4
a=ice-pwd:9dzX3UXkolinKQ4c2pC6OMhk
a=candidate:8906128254 1 udp 659136 192.168.131.253 16386 typ host generation 0
a=candidate:8906128254 2 udp 659136 192.168.131.253 16386 typ host generation 0
a=ptime:20
m=video 16388 RTP/SAVPF 100
b=AS:256
a=rtpmap:100 VP8/90000
a=fingerprint:sha-256 34:46:35:C3:98:97:79:67:E8:B9:5F:FF:82:C4:EE:4E:2D:CA:DE:46:BD:B2:6D:D5:A6:07:C9:26:5B:3F:AE:10
a=rtcp-mux
a=rtcp:16388 IN IP4 192.168.131.253
a=rtcp-fb:100 ccm fir
a=ssrc:311706813 cname:mKuFX0u9qEy78yIQ
a=ssrc:311706813 msid:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1N v0
a=ssrc:311706813 mslabel:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1N
a=ssrc:311706813 label:cmSQsnGZmAjyQE9LSdNN71eThgHwKa1Nv0
a=ice-ufrag:r8kiofwWRbcruXG5
a=ice-pwd:cZIuIMxYd8z7nQ7EPhM3m9P4
a=candidate:8142942344 1 udp 659136 192.168.131.253 16388 typ host generation 0
a=candidate:8142942344 2 udp 659134 192.168.131.253 16388 typ host generation 0




#Answer generated in Chrome:
v=0
o=- 1379360262027728742 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS imsh1etBhqjVisP1YbY3IW5ML1FrCLyPVD5S
m=audio 41747 RTP/SAVPF 111 9 0 8 101 13
c=IN IP4 192.168.131.253
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:211962667 1 udp 2122260223 10.0.3.1 41747 typ host generation 0
a=candidate:896282172 1 udp 2122194687 192.168.131.253 58504 typ host generation 0
a=candidate:2441410931 1 udp 2122129151 172.17.42.1 45609 typ host generation 0
a=candidate:868527874 1 udp 1686052607 192.168.131.253 41747 typ srflx raddr 10.0.3.1 rport 41747 generation 0
a=candidate:953906854 1 udp 1685921535 192.168.131.253 45609 typ srflx raddr 172.17.42.1 rport 45609 generation 0
a=candidate:1109506011 1 tcp 1518280447 10.0.3.1 0 typ host tcptype active generation 0
a=candidate:2079314636 1 tcp 1518214911 192.168.131.253 0 typ host tcptype active generation 0
a=candidate:3741779331 1 tcp 1518149375 172.17.42.1 0 typ host tcptype active generation 0
a=ice-ufrag:4IDqem5vOr15nQSg
a=ice-pwd:5+LtaRvHaFdEv7+vUDnP8Ofl
a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
a=setup:active
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:13 CN/8000
a=maxptime:60
a=ssrc:982941744 cname:MMvCy0RouNFfAfdZ
a=ssrc:982941744 msid:imsh1etBhqjVisP1YbY3IW5ML1FrCLyPVD5S e4bf2510-f73a-4df4-8cfe-5308a89ca8f7
a=ssrc:982941744 mslabel:imsh1etBhqjVisP1YbY3IW5ML1FrCLyPVD5S
a=ssrc:982941744 label:e4bf2510-f73a-4df4-8cfe-5308a89ca8f7
m=video 36031 RTP/SAVPF 100
c=IN IP4 192.168.131.253
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:211962667 1 udp 2122260223 10.0.3.1 36031 typ host generation 0
a=candidate:896282172 1 udp 2122194687 192.168.131.253 35064 typ host generation 0
a=candidate:2441410931 1 udp 2122129151 172.17.42.1 37784 typ host generation 0
a=candidate:868527874 1 udp 1686052607 192.168.131.253 36031 typ srflx raddr 10.0.3.1 rport 36031 generation 0
a=candidate:953906854 1 udp 1685921535 192.168.131.253 37784 typ srflx raddr 172.17.42.1 rport 37784 generation 0
a=candidate:1109506011 1 tcp 1518280447 10.0.3.1 0 typ host tcptype active generation 0
a=candidate:2079314636 1 tcp 1518214911 192.168.131.253 0 typ host tcptype active generation 0
a=candidate:3741779331 1 tcp 1518149375 172.17.42.1 0 typ host tcptype active generation 0
a=ice-ufrag:JopYjvWKl1fi3YnD
a=ice-pwd:FWkLjxPvcbF7QRDLW9ML5tc8
a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
a=setup:active
a=mid:video
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=ssrc:1880102335 cname:MMvCy0RouNFfAfdZ
a=ssrc:1880102335 msid:imsh1etBhqjVisP1YbY3IW5ML1FrCLyPVD5S 05869856-a385-424a-8817-d3d748823465
a=ssrc:1880102335 mslabel:imsh1etBhqjVisP1YbY3IW5ML1FrCLyPVD5S
a=ssrc:1880102335 label:05869856-a385-424a-8817-d3d748823465


#Answer received in Android from FreeSwitch:
v=0
o=FreeSWITCH 1426145178 1426145179 IN IP4 192.168.131.253
s=FreeSWITCH
c=IN IP4 192.168.131.253
t=0 0
a=msid-semantic: WMS BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5
m=audio 16384 RTP/SAVPF 111 126 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:126 telephone-event/8000
a=rtpmap:106 CN/8000
a=ptime:20
a=fingerprint:sha-1 73:08:7C:79:1F:8F:5A:E7:4B:C1:D7:05:EA:4D:98:F3:36:46:39:72
a=rtcp-mux
a=rtcp:16384 IN IP4 192.168.131.253
a=ssrc:1024631706 cname:g0X2GoBYCY2J5usA
a=ssrc:1024631706 msid:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5 a0
a=ssrc:1024631706 mslabel:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5
a=ssrc:1024631706 label:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5a0
a=ice-ufrag:lMaDMkwg5zIz3kv9
a=ice-pwd:aj2HxtvZiozdJSEUvEn7PKdL
a=candidate:8379233368 1 udp 659136 192.168.131.253 16384 typ host generation 0
m=video 16390 RTP/SAVPF 100
b=AS:256
a=rtpmap:100 VP8/90000
a=fingerprint:sha-1 73:08:7C:79:1F:8F:5A:E7:4B:C1:D7:05:EA:4D:98:F3:36:46:39:72
a=rtcp-mux
a=rtcp:16390 IN IP4 192.168.131.253
a=rtcp-fb:* fir
a=rtcp-fb:100 ccm fir
a=ssrc:311571677 cname:g0X2GoBYCY2J5usA
a=ssrc:311571677 msid:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5 v0
a=ssrc:311571677 mslabel:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5
a=ssrc:311571677 label:BNo1VQAJroxehgCTF0uKs9k3EUdBEDL5v0
a=ice-ufrag:afvyUrPqxfjH9tPf
a=ice-pwd:snn0NUc6MAcrFkqGqM5a3Krr
a=candidate:6545229027 1 udp 659136 192.168.131.253 16390 typ host generation 0

From tfred31 at yahoo.com  Thu Mar 12 17:15:15 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Thu, 12 Mar 2015 07:15:15 -0700
Subject: [Freeswitch-users] Aastra 9133i phones Intercom/AutoAnswer
	w/FreeSWITCH
Message-ID: <1426169715.90923.YahooMailBasic@web160201.mail.bf1.yahoo.com>

Hello,

My current challenge is getting the  AutoAnswer to work on the 9133i phones to support Intercom functionality.

Within the   conf/dialplan/default.xml file  I am basically using the FreeSWITCH default settings
        <extension name = "extension-intercom">
            <condition field = "destination_number" expression = "^8(1[012][0-9])|8(10[012][0-9])$">   <!-- 3 or 4 digit extensions -->
                <action application = "set" data = "dialed_extension=$1"/>
                <action application = "export" data = "sip_auto_answer=true"/>
                <action application = "bridge" data = "user/${dialed_extension}@${domain_name}"/>
            </condition>
        </extension>

The AutoAnswer is working on the single Aastra 6731i phone that I have in the system, but not on any of the 9133i phones.

This is in spite of both phone models have the  Auto-Answer parameter set in their individual internal configurations.
   9133i:    Incoming Intercom Settings --  Auto-Answer  -- Enabled
   6731i:    Incoming Intercom Settings --  Auto-Answer  -- Enabled

NOTE - the 9133i extension's will ring, but not go into the AutoAnswer/Speaker/Intercom mode

While searching the web for an answer I came across an old posting (Feb 11, 2011) where  Tim St. Pierre  indicates that he has a relatively large FreeSWITCH phone system incorporating a number of Aastra models.  He goes on to say that he has   Intercom calls (auto-answer to speaker phone)   working on the 9133i phones (among others)

I'd appreciate any advice anyone might have so that I can get the FreeSWITCH Intercom auto-answer working on the 9133i  phones as well.

Thanks,
TF




From mike at jerris.com  Thu Mar 12 17:57:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 Mar 2015 10:57:27 -0400
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <CA+M05dCo87sc3kybYtz8u=PMdBAYZ2KNtMCWEECp6aZ0_E7bsw@mail.gmail.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
	<CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
	<29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>
	<CA+M05dCo87sc3kybYtz8u=PMdBAYZ2KNtMCWEECp6aZ0_E7bsw@mail.gmail.com>
Message-ID: <D766B545-6A6C-4847-A741-3777BAD3FEF7@jerris.com>

its not changing it back and forth to sha-256/sha-1 those are 2 different channels... the leg to android is always sha-1.  That being said, Nothing we have tested against uses sha-1.  That could be an issue.  You should look at the full debug log of the call between fs and android and see if there is anything useful there.

> On Mar 12, 2015, at 8:07 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> Hi,
> 
> Unfortunately not everything runs smoothly. I run FreeSWITCH Version 1.5.15b+git~20150203T210457Z~4174fb9cbe~64bit (git 4174fb9 2015-02-03 21:04:57Z 64bit) with the default configuration + webrtc module + tweaked bridge application for 1010 - 1019 extensions where I added ignore_early_media=true because of chrome troubles with pranswer and media_webrtc=true because one of my clients is actually JAIN SIP  without WebSockets.
> 
> Now I call from Chrome to my Android app. In Android app I receive modified SDP from FreeSwitch and all the media traffic goes though FreeSwitch. I have the audio and video in both directions. 
> 
> But when I createOffer in the Android application and send it to Chrome the media is not flowing in any directions. In case I set "<action application="set" data="bypass_media=true"/>" media starts flowing.
> 
> PS. May be it's irrelevant  but the only strange thing I noticed is that Android App produces fingerprint in sha-1 then FreeSwitch changes it to sha-256 and sends to Chrome. Chrome responds with sha-256 then FreeSwitch modifies it back to sha-1.
> 
> I don't know, may be I forget about some other magic options? 
> 
> PS: SDPs are in the attachment
> 
> On Fri, Mar 6, 2015 at 4:43 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Always nice to hear that we are magic!  
> 
>> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>> 
>> thank you very much Michael, it magically works.
>> 
>> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> you need to tell freeswitch to send a webrtc compatible SDP.
>> 
>> https://wiki.freeswitch.org/wiki/Variable_media_webrtc <https://wiki.freeswitch.org/wiki/Variable_media_webrtc>
>> 
>> 
>>> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>>> 
>>> Hi,
>>> 
>>> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org <http://webrtc.org/> android library and connected to FreeSwitch via UDP.
>>> 
>>> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection.
>>> 
>>> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:
>>> 
>>> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
>>> 
>>> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/ <http://tryit.jssip.net/>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.
>>> 
>>> Here is initial SDP from Chrome (http://tryit.jssip.net/ <http://tryit.jssip.net/>):
>>> 
>>> v=0
>>> o=- 6887715720880489867 2 IN IP4 127.0.0.1
>>> s=-
>>> t=0 0
>>> a=group:BUNDLE audio video
>>> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
>>> c=IN IP4 192.168.122.1
>>> a=rtcp:38359 IN IP4 192.168.122.1
>>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>>> .......
>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>> a=ice-options:google-ice
>>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>> a=setup:actpass
>>> a=mid:audio
>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>> a=sendrecv
>>> a=rtcp-mux
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10
>>> a=rtpmap:103 ISAC/16000
>>> a=rtpmap:104 ISAC/32000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:106 CN/32000
>>> a=rtpmap:105 CN/16000
>>> a=rtpmap:13 CN/8000
>>> a=rtpmap:126 telephone-event/8000
>>> a=maxptime:60
>>> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
>>> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>> m=video 38359 RTP/SAVPF 100 116 117 96
>>> c=IN IP4 192.168.122.1
>>> a=rtcp:38359 IN IP4 192.168.122.1
>>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>>> ............
>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>> a=ice-options:google-ice
>>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>> a=setup:actpass
>>> a=mid:video
>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>> a=recvonly
>>> a=rtcp-mux
>>> a=rtpmap:100 VP8/90000
>>> a=rtcp-fb:100 ccm fir
>>> a=rtcp-fb:100 nack
>>> a=rtcp-fb:100 nack pli
>>> a=rtcp-fb:100 goog-remb
>>> a=rtpmap:116 red/90000
>>> a=rtpmap:117 ulpfec/90000
>>> a=rtpmap:96 rtx/90000
>>> a=fmtp:96 apt=100
>>> 
>>> 
>>> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>>> 
>>> v=0
>>> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.131.253
>>> t=0 0
>>> m=audio 16390 RTP/AVP 111 0 8 101 13
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> m=video 16388 RTP/AVP 100
>>> a=rtpmap:100 VP8/90000
>>> 
>>> 
>>> I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. 
>>> 
>>> So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/675244cc/attachment.html 

From jdickson at evolvetsi.com  Thu Mar 12 04:05:02 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Wed, 11 Mar 2015 21:05:02 -0400
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
Message-ID: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>

Greetings!

I'm working on putting together an HA FreeSwitch setup, and I'm having
trouble getting sofia recovery to work.  I've got to the point where call
state is shared successfully (master-master MySQL at the moment), and where
the new node attempts to recover the calls.

The problem seems to be in the A leg re-invite.  I've tried a few different
A-leg user agents, and get the same issue with all... FS hangs up the leg
with INCOMPATIBLE_DESTINATION after sending an ACK to the A leg's OK.  The
SDP looks fine to me, and I don't see any error messages about codec
selection etc..

I turned up logging as far as I knew how (I'm still pretty new to FS, so if
there's a better log method, let me know!)..  The output is on pastebin
here:

http://pastebin.com/WK8A7Kx7

This log snippet is of the log messages generated after a sofia recover on
my secondary node (being issued by pacemaker).  There was a single call
connected at the time of the failover..

Any ideas?  I've seen a couple JIRA entries and mailing list entries for
things that are similar, but usually the resolution was to update to the
newest FS..  I'm running the latest 1.4 from the freeswitch debian repo..

The A leg client from this log trace is Zoiper Free.

version info:
# freeswitch -version
FreeSWITCH version: 1.4.15-1~64bit (-1 64bit)

Any pointers would be appreciated!

Joe
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150311/d3aec96f/attachment-0001.html 

From lists at telefaks.de  Thu Mar 12 19:28:16 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 12 Mar 2015 17:28:16 +0100
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with no
	challenge
Message-ID: <5501BEA0.4090708@telefaks.de>

Hello,

we receive a number of Invites from certain IPs, who want to break into
our system and call external premium rate numbers
Unwanted registers we can block already, but we still have the issue to
block specific invites from fraudulent IPs inside the iptables firewall.

In the Freeswitch log we see:
2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
signal sofia/internal/149 at 10.11.12.13 [BREAK]
2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
signal sofia/internal/149 at 10.11.12.13 [BREAK]
2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
Rejected by acl "domains". Falling back to Digest auth.
2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/149 at 10.11.12.13) State NEW
2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
signal sofia/internal/149 at 10.11.12.13 [BREAK]
2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
167bb9ee-c8d0-11e4-9f31-b39e581405c5
2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
Abandoned   

The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
10.11.12.13 is the (anonymized) IP of our server.

The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
sends "authentication required". Freeswitch then logs this entry with
"Abandoned" (see last line above) and that's it.

So Is there any way to make Freeswitch show up a log line with the
fraudulent IP 15.194.164.26 and some text like "abandonned"?
Example for extending a current log line
    2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
Abandoned for IP 15.194.164.26
This would enable us to process this entry with fail2ban and block this
IP in the Firewall.

Any other hint is welcome.

-- 
With kind regards
Marvin Keil 

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/db1fb7c9/attachment.html 

From anthony.minessale at gmail.com  Thu Mar 12 19:42:06 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Mar 2015 11:42:06 -0500
Subject: [Freeswitch-users] ESL object taking large memory
In-Reply-To: <CAFDf6-9wp5+V9v36ewbv=Y0WgpYmkt8SQuG8c5=FTg9OA=qw_A@mail.gmail.com>
References: <CAFDf6-9Rn1ro4yYhhEMxGH5jeYv4zsyRS2Yw8KhaOTiYKSa4tg@mail.gmail.com>
	<CAFDf6-9wp5+V9v36ewbv=Y0WgpYmkt8SQuG8c5=FTg9OA=qw_A@mail.gmail.com>
Message-ID: <CAKbxfG8yZ+44HDNhZisEc6=xgw289LGDMp2NDFmgCoMOSXawUA@mail.gmail.com>

File a jira and confirm you can make it work with just  65536 set for both
constants


On Tue, Mar 10, 2015 at 2:00 PM, DP Siddharth <
dp.siddharth at eng.knowlarity.com> wrote:

> Going through logs I found following commit id, where these changes were
> done. Any suggetion about having
>
> #define BUF_CHUNK 65536 * 50
> #define BUF_START 65536 * 100
>
> to some smaller values?
>
> commit 2081bf97b9836f5299c22edbb1ead077842ea2bc
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Thu Dec 16 11:33:38 2010 -0600
>
>     use a packet buffer for ESL
>
> On Tue, Mar 10, 2015 at 6:50 PM, DP Siddharth <
> dp.siddharth at eng.knowlarity.com> wrote:
>
>> Hi All,
>>
>> I am working on python/esl based server. We are seeing memory getting
>> increase by ~6.5MB when con = ESLconnection() get called.
>>
>> We tried to cleanup this object as soon calls gets complete, but somehow
>> we are not successful.
>>
>> Further looking into esl/src/include/esl.h we found
>>
>> #define BUF_CHUNK 65536 * 50
>> #define BUF_START 65536 * 100
>>
>> modifying these values help in controlling python server memory growth.
>>
>> Can someone help me in understanding why we have this buffer size of
>> 6.5MB?
>>
>>
>> --
>> Thanks & Regards,
>> D P Siddharth
>> Director (Platform)
>> Knowlarity Communications
>> Ph: +919999115231
>> dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>
>>
>> *"Come together to build a lasting world-class cloud telephony company
>> that helps businesses grow"*
>>
>
>
>
> --
> Thanks & Regards,
> D P Siddharth
> Director (Platform)
> Knowlarity Communications
> Ph: +919999115231
> dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>
>
> *"Come together to build a lasting world-class cloud telephony company
> that helps businesses grow"*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/41fc20f7/attachment.html 

From kyle.king at quentustech.com  Thu Mar 12 19:43:30 2015
From: kyle.king at quentustech.com (Kyle King)
Date: Thu, 12 Mar 2015 12:43:30 -0400
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no	challenge
In-Reply-To: <5501BEA0.4090708@telefaks.de>
References: <5501BEA0.4090708@telefaks.de>
Message-ID: <E3361EF0-3EA5-4D20-BE7F-00AB2A4A554A@quentustech.com>

Have you tried mod_fail2ban? 

On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de> wrote:
>Hello,
>
>we receive a number of Invites from certain IPs, who want to break into
>our system and call external premium rate numbers
>Unwanted registers we can block already, but we still have the issue to
>block specific invites from fraudulent IPs inside the iptables
>firewall.
>
>In the Freeswitch log we see:
>2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
>sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>signal sofia/internal/149 at 10.11.12.13 [BREAK]
>2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>signal sofia/internal/149 at 10.11.12.13 [BREAK]
>2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472
>(sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
>version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
>2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>Rejected by acl "domains". Falling back to Digest auth.
>2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491
>(sofia/internal/149 at 10.11.12.13) State NEW
>2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
>signal sofia/internal/149 at 10.11.12.13 [BREAK]
>2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>167bb9ee-c8d0-11e4-9f31-b39e581405c5
>2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>Abandoned   
>
>The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
>10.11.12.13 is the (anonymized) IP of our server.
>
>The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
>sends "authentication required". Freeswitch then logs this entry with
>"Abandoned" (see last line above) and that's it.
>
>So Is there any way to make Freeswitch show up a log line with the
>fraudulent IP 15.194.164.26 and some text like "abandonned"?
>Example for extending a current log line
>   2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>Abandoned for IP 15.194.164.26
>This would enable us to process this entry with fail2ban and block this
>IP in the Firewall.
>
>Any other hint is welcome.
>
>-- 
>With kind regards
>Marvin Keil 
>
>Telefaks Services GmbH
>mailto:lists (att) telefaks.de
>Internet: www.telefaks.de
>
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services: 
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/b4aa5fc3/attachment-0001.html 

From jason.holden at start.ca  Thu Mar 12 19:53:08 2015
From: jason.holden at start.ca (Jason Holden)
Date: Thu, 12 Mar 2015 12:53:08 -0400
Subject: [Freeswitch-users] failover in freeswitch dialplan using | or lcr
Message-ID: <185CD03D8FD84FFD9C7BDCF1D7E21D7C@bob>

Hi,

If I send an invite for a call and don't receive a response to the initial
invite message how can I force FS to try the next route right away?

 

Here is one of my lcr dialplan chunks.

    <condition field="destination_number" expression="^(.*)$">

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

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

      <action application="set" data="domain_name=$${domain}"/>

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

      <action application="ring_ready" />

      <action application="set" data="ringback=${us-ring}"/>

      <action application="set" data="transfer-ringback=${us-ring}"/>  

      <action application="lcr" data="$1 inbound"/>

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

      <action application="bridge" data="${lcr_auto_route}"/>

    </condition>

  </extension>

 

All advise is welcome, thanks.

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/9d68bacb/attachment.html 

From italorossib at gmail.com  Thu Mar 12 19:51:34 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 12 Mar 2015 13:51:34 -0300
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <E3361EF0-3EA5-4D20-BE7F-00AB2A4A554A@quentustech.com>
References: <5501BEA0.4090708@telefaks.de>
	<E3361EF0-3EA5-4D20-BE7F-00AB2A4A554A@quentustech.com>
Message-ID: <CAJR7f2Hh74Pyqo8TysVZMp=-EhQw30U9RmkuQQrdRitBcQQOog@mail.gmail.com>

Version?

I'm almost sure this is already implemented in master.
Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com> escreveu:

> Have you tried mod_fail2ban?
>
> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de>
> wrote:
>>
>> Hello,
>>
>> we receive a number of Invites from certain IPs, who want to break into
>> our system and call external premium rate numbers
>> Unwanted registers we can block already, but we still have the issue to
>> block specific invites from fraudulent IPs inside the iptables firewall.
>>
>> In the Freeswitch log we see:
>> 2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/149 at 10.11.12.13 [BREAK]
>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/149 at 10.11.12.13 [BREAK]
>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
>> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
>> version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>> Rejected by acl "domains". Falling back to Digest auth.
>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
>> sofia/internal/149 at 10.11.12.13) State NEW
>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/149 at 10.11.12.13 [BREAK]
>> 2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>> 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>> Abandoned
>>
>> The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
>> 10.11.12.13 is the (anonymized) IP of our server.
>>
>> The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
>> sends "authentication required". Freeswitch then logs this entry with
>> "Abandoned" (see last line above) and that's it.
>>
>> So Is there any way to make Freeswitch show up a log line with the
>> fraudulent IP 15.194.164.26 and some text like "abandonned"?
>> Example for extending a current log line
>>     2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>> Abandoned for IP 15.194.164.26
>> This would enable us to process this entry with fail2ban and block this
>> IP in the Firewall.
>>
>> Any other hint is welcome.
>>
>> --
>> With kind regards
>> Marvin Keil
>>
>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>> Internet: www.telefaks.de
>>
>> ------------------------------
>>
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> --
> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/cffe193f/attachment.html 

From dp.siddharth at eng.knowlarity.com  Thu Mar 12 19:53:33 2015
From: dp.siddharth at eng.knowlarity.com (DP Siddharth)
Date: Thu, 12 Mar 2015 22:23:33 +0530
Subject: [Freeswitch-users] ESL object taking large memory
In-Reply-To: <CAKbxfG8yZ+44HDNhZisEc6=xgw289LGDMp2NDFmgCoMOSXawUA@mail.gmail.com>
References: <CAFDf6-9Rn1ro4yYhhEMxGH5jeYv4zsyRS2Yw8KhaOTiYKSa4tg@mail.gmail.com>
	<CAFDf6-9wp5+V9v36ewbv=Y0WgpYmkt8SQuG8c5=FTg9OA=qw_A@mail.gmail.com>
	<CAKbxfG8yZ+44HDNhZisEc6=xgw289LGDMp2NDFmgCoMOSXawUA@mail.gmail.com>
Message-ID: <CAFDf6--JfGmaQYkxnfZxM7o+ZfBY3-RkKHOwGNbA3SdSDwgR9A@mail.gmail.com>

Thanks Anthony. Yes I tested with over a million calls & I didn't found any
issue with 64K & 128K size.

I will update same on Jira also.



On Thu, Mar 12, 2015 at 10:12 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> File a jira and confirm you can make it work with just  65536 set for both
> constants
>
>
> On Tue, Mar 10, 2015 at 2:00 PM, DP Siddharth <
> dp.siddharth at eng.knowlarity.com> wrote:
>
>> Going through logs I found following commit id, where these changes were
>> done. Any suggetion about having
>>
>> #define BUF_CHUNK 65536 * 50
>> #define BUF_START 65536 * 100
>>
>> to some smaller values?
>>
>> commit 2081bf97b9836f5299c22edbb1ead077842ea2bc
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date:   Thu Dec 16 11:33:38 2010 -0600
>>
>>     use a packet buffer for ESL
>>
>> On Tue, Mar 10, 2015 at 6:50 PM, DP Siddharth <
>> dp.siddharth at eng.knowlarity.com> wrote:
>>
>>> Hi All,
>>>
>>> I am working on python/esl based server. We are seeing memory getting
>>> increase by ~6.5MB when con = ESLconnection() get called.
>>>
>>> We tried to cleanup this object as soon calls gets complete, but somehow
>>> we are not successful.
>>>
>>> Further looking into esl/src/include/esl.h we found
>>>
>>> #define BUF_CHUNK 65536 * 50
>>> #define BUF_START 65536 * 100
>>>
>>> modifying these values help in controlling python server memory growth.
>>>
>>> Can someone help me in understanding why we have this buffer size of
>>> 6.5MB?
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> D P Siddharth
>>> Director (Platform)
>>> Knowlarity Communications
>>> Ph: +919999115231
>>> dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>
>>>
>>> *"Come together to build a lasting world-class cloud telephony company
>>> that helps businesses grow"*
>>>
>>
>>
>>
>> --
>> Thanks & Regards,
>> D P Siddharth
>> Director (Platform)
>> Knowlarity Communications
>> Ph: +919999115231
>> dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>
>>
>> *"Come together to build a lasting world-class cloud telephony company
>> that helps businesses grow"*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & Regards,
D P Siddharth
Director (Platform)
Knowlarity Communications
Ph: +919999115231
dp.siddharth at eng.knowlarity.com <emailID at knowlarity.com>

*"Come together to build a lasting world-class cloud telephony company that
helps businesses grow"*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/94ae44aa/attachment-0001.html 

From steveayre at gmail.com  Thu Mar 12 20:36:37 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 12 Mar 2015 17:36:37 +0000
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <5501BEA0.4090708@telefaks.de>
References: <5501BEA0.4090708@telefaks.de>
Message-ID: <CAFiqYunJZG7vM4NYM4GaxsVSAwkkh=9BcE=H9rsWctmy3_yYUg@mail.gmail.com>

Check <param name="log-auth-failures" value="true"/> is on your sofia
profiles. There should be a message when the challenge is sent and when the
call is abandoned. It's not the 'Abandoned' switch_core_state_machine.c one
but one coming from sofia_reg.c.

On 12 March 2015 at 16:28, Peter Steinbach <lists at telefaks.de> wrote:

>  Hello,
>
> we receive a number of Invites from certain IPs, who want to break into
> our system and call external premium rate numbers
> Unwanted registers we can block already, but we still have the issue to
> block specific invites from fraudulent IPs inside the iptables firewall.
>
> In the Freeswitch log we see:
> 2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
> version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26 Rejected
> by acl "domains". Falling back to Digest auth.
> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
> sofia/internal/149 at 10.11.12.13) State NEW
> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
> 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned
>
> The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
> 10.11.12.13 is the (anonymized) IP of our server.
>
> The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
> sends "authentication required". Freeswitch then logs this entry with
> "Abandoned" (see last line above) and that's it.
>
> So Is there any way to make Freeswitch show up a log line with the
> fraudulent IP 15.194.164.26 and some text like "abandonned"?
> Example for extending a current log line
>     2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned for IP 15.194.164.26
> This would enable us to process this entry with fail2ban and block this IP
> in the Firewall.
>
> Any other hint is welcome.
>
> --
> With kind regards
> Marvin Keil
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/b02a1e48/attachment.html 

From flokrrr at gmail.com  Thu Mar 12 21:42:50 2015
From: flokrrr at gmail.com (Florent Krieg)
Date: Thu, 12 Mar 2015 19:42:50 +0100
Subject: [Freeswitch-users] Limit application usage in Lua
Message-ID: <CAABKsuTOGeB=UfovXeZvwLyjprFC97QdyJ4Wdf72ZLmsQd0YqQ@mail.gmail.com>

Hi all,

I wanted to implement call rate limits using this Lua instruction:
session:execute("limit", "hash " .. billedcaller .. " caps 10/1
!REQUESTED_CHAN_UNAVAIL")

It is actually working, but the rest of the Lua dialplan is still
processed, which is a problem in my case.

I made a dirty but quick workaround, looking like this:
rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
if not rate_var then
    logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
aborting the dialplan.")
    return
end

How can it be done properly?
Is there a way to be able to get the result of the 'limit' app call?
Or shall I check the status of the a-leg just after to decide to process
the rest of the dialplan or not?

I know that this issue doesn't occur in a 'pure' XML dialplan, but I'm
trying to find a solution for my Lua-only dialplan.

Thanks in advance if you have any idea.
I'm willing to try any possible solution you would think about!

Florent
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/c63f867d/attachment.html 

From s.safarov at gmail.com  Thu Mar 12 21:44:50 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 12 Mar 2015 21:44:50 +0300
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <5501BEA0.4090708@telefaks.de>
References: <5501BEA0.4090708@telefaks.de>
Message-ID: <CAHtxdDejv19mdR2iozv3ciYPNX3OLGYwZXuW0njxtvKpspgYAg@mail.gmail.com>

Marvin you can use solution published at
https://freeswitch.org/jira/browse/FS-7125
https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/63a622decc0994d69a8e4ec223cb5359430f03d9

Currently I successfully block that calls

On Thu, Mar 12, 2015 at 7:28 PM, Peter Steinbach <lists at telefaks.de> wrote:

>  Hello,
>
> we receive a number of Invites from certain IPs, who want to break into
> our system and call external premium rate numbers
> Unwanted registers we can block already, but we still have the issue to
> block specific invites from fraudulent IPs inside the iptables firewall.
>
> In the Freeswitch log we see:
> 2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
> version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26 Rejected
> by acl "domains". Falling back to Digest auth.
> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
> sofia/internal/149 at 10.11.12.13) State NEW
> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
> 2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
> 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned
>
> The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
> 10.11.12.13 is the (anonymized) IP of our server.
>
> The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
> sends "authentication required". Freeswitch then logs this entry with
> "Abandoned" (see last line above) and that's it.
>
> So Is there any way to make Freeswitch show up a log line with the
> fraudulent IP 15.194.164.26 and some text like "abandonned"?
> Example for extending a current log line
>     2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned for IP 15.194.164.26
> This would enable us to process this entry with fail2ban and block this IP
> in the Firewall.
>
> Any other hint is welcome.
>
> --
> With kind regards
> Marvin Keil
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/b86ecff6/attachment.html 

From s.safarov at gmail.com  Thu Mar 12 21:51:14 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 12 Mar 2015 21:51:14 +0300
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
Message-ID: <CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>

I has used MySQL in master-master cluster and find it is unstable
configuration. Many troubles with cluster consistence.
I recommend use configuration master-slave with tools DB synchronization.

Also migration to PostgreSQL cluster will be smart decision.

On Thu, Mar 12, 2015 at 4:05 AM, Joseph Dickson <jdickson at evolvetsi.com>
wrote:

> Greetings!
>
> I'm working on putting together an HA FreeSwitch setup, and I'm having
> trouble getting sofia recovery to work.  I've got to the point where call
> state is shared successfully (master-master MySQL at the moment), and where
> the new node attempts to recover the calls.
>
> The problem seems to be in the A leg re-invite.  I've tried a few
> different A-leg user agents, and get the same issue with all... FS hangs up
> the leg with INCOMPATIBLE_DESTINATION after sending an ACK to the A leg's
> OK.  The SDP looks fine to me, and I don't see any error messages about
> codec selection etc..
>
> I turned up logging as far as I knew how (I'm still pretty new to FS, so
> if there's a better log method, let me know!)..  The output is on pastebin
> here:
>
> http://pastebin.com/WK8A7Kx7
>
> This log snippet is of the log messages generated after a sofia recover on
> my secondary node (being issued by pacemaker).  There was a single call
> connected at the time of the failover..
>
> Any ideas?  I've seen a couple JIRA entries and mailing list entries for
> things that are similar, but usually the resolution was to update to the
> newest FS..  I'm running the latest 1.4 from the freeswitch debian repo..
>
> The A leg client from this log trace is Zoiper Free.
>
> version info:
> # freeswitch -version
> FreeSWITCH version: 1.4.15-1~64bit (-1 64bit)
>
> Any pointers would be appreciated!
>
> Joe
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/b049d425/attachment-0001.html 

From s.safarov at gmail.com  Thu Mar 12 22:03:03 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 12 Mar 2015 22:03:03 +0300
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <CAJR7f2Hh74Pyqo8TysVZMp=-EhQw30U9RmkuQQrdRitBcQQOog@mail.gmail.com>
References: <5501BEA0.4090708@telefaks.de>
	<E3361EF0-3EA5-4D20-BE7F-00AB2A4A554A@quentustech.com>
	<CAJR7f2Hh74Pyqo8TysVZMp=-EhQw30U9RmkuQQrdRitBcQQOog@mail.gmail.com>
Message-ID: <CAHtxdDdq6sRPoFSYPShyi7LzU5t7P2HKMiuT3nW-CiG6YJO5Sw@mail.gmail.com>

?talo I am not rewrite patch set use network_addr in caller profile and
path not merget to master.

Sergey

On Thu, Mar 12, 2015 at 7:51 PM, ?talo Rossi <italorossib at gmail.com> wrote:

> Version?
>
> I'm almost sure this is already implemented in master.
> Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com> escreveu:
>
>> Have you tried mod_fail2ban?
>>
>> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de>
>> wrote:
>>>
>>> Hello,
>>>
>>> we receive a number of Invites from certain IPs, who want to break into
>>> our system and call external premium rate numbers
>>> Unwanted registers we can block already, but we still have the issue to
>>> block specific invites from fraudulent IPs inside the iptables firewall.
>>>
>>> In the Freeswitch log we see:
>>> 2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
>>> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
>>> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>>> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
>>> version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
>>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>>> Rejected by acl "domains". Falling back to Digest auth.
>>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
>>> sofia/internal/149 at 10.11.12.13) State NEW
>>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>> 2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>>> 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>>> Abandoned
>>>
>>> The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
>>> 10.11.12.13 is the (anonymized) IP of our server.
>>>
>>> The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
>>> sends "authentication required". Freeswitch then logs this entry with
>>> "Abandoned" (see last line above) and that's it.
>>>
>>> So Is there any way to make Freeswitch show up a log line with the
>>> fraudulent IP 15.194.164.26 and some text like "abandonned"?
>>> Example for extending a current log line
>>>     2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>>> Abandoned for IP 15.194.164.26
>>> This would enable us to process this entry with fail2ban and block this
>>> IP in the Firewall.
>>>
>>> Any other hint is welcome.
>>>
>>> --
>>> With kind regards
>>> Marvin Keil
>>>
>>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>>> Internet: www.telefaks.de
>>>
>>> ------------------------------
>>>
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> --
>> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/d4e83d4c/attachment.html 

From italorossib at gmail.com  Thu Mar 12 22:27:27 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 12 Mar 2015 16:27:27 -0300
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <CAHtxdDdq6sRPoFSYPShyi7LzU5t7P2HKMiuT3nW-CiG6YJO5Sw@mail.gmail.com>
References: <5501BEA0.4090708@telefaks.de>
	<E3361EF0-3EA5-4D20-BE7F-00AB2A4A554A@quentustech.com>
	<CAJR7f2Hh74Pyqo8TysVZMp=-EhQw30U9RmkuQQrdRitBcQQOog@mail.gmail.com>
	<CAHtxdDdq6sRPoFSYPShyi7LzU5t7P2HKMiuT3nW-CiG6YJO5Sw@mail.gmail.com>
Message-ID: <CAJR7f2GR22MXbe8fr4W2KuDDEwaDVDN8i83X6wOSaJMA-g7qhQ@mail.gmail.com>

I set the JIRA status as Needs Review, hope it get merged soon.

On Thu, Mar 12, 2015 at 4:03 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> ?talo I am not rewrite patch set use network_addr in caller profile and
> path not merget to master.
>
> Sergey
>
> On Thu, Mar 12, 2015 at 7:51 PM, ?talo Rossi <italorossib at gmail.com>
> wrote:
>
>> Version?
>>
>> I'm almost sure this is already implemented in master.
>> Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com> escreveu:
>>
>>> Have you tried mod_fail2ban?
>>>
>>> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de>
>>> wrote:
>>>>
>>>> Hello,
>>>>
>>>> we receive a number of Invites from certain IPs, who want to break into
>>>> our system and call external premium rate numbers
>>>> Unwanted registers we can block already, but we still have the issue to
>>>> block specific invites from fraudulent IPs inside the iptables firewall.
>>>>
>>>> In the Freeswitch log we see:
>>>> 2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
>>>> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>>> 2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
>>>> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>>>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>>>> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
>>>> version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
>>>> 2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
>>>> sofia/internal/149 at 10.11.12.13) State NEW
>>>> 2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/149 at 10.11.12.13 [BREAK]
>>>> 2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>>>> 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>>>> Abandoned
>>>>
>>>> The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
>>>> 10.11.12.13 is the (anonymized) IP of our server.
>>>>
>>>> The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
>>>> sends "authentication required". Freeswitch then logs this entry with
>>>> "Abandoned" (see last line above) and that's it.
>>>>
>>>> So Is there any way to make Freeswitch show up a log line with the
>>>> fraudulent IP 15.194.164.26 and some text like "abandonned"?
>>>> Example for extending a current log line
>>>>     2015-03-12 16:54:48.461568 [WARNING]
>>>> switch_core_state_machine.c:572 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>>>> sofia/internal/149 at 10.11.12.13 Abandoned for IP 15.194.164.26
>>>> This would enable us to process this entry with fail2ban and block this
>>>> IP in the Firewall.
>>>>
>>>> Any other hint is welcome.
>>>>
>>>> --
>>>> With kind regards
>>>> Marvin Keil
>>>>
>>>> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
>>>> Internet: www.telefaks.de
>>>>
>>>> ------------------------------
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> --
>>> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/a6d7fce8/attachment-0001.html 

From krice at freeswitch.org  Thu Mar 12 23:46:00 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 12 Mar 2015 14:46:00 -0600
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
 no challenge
In-Reply-To: <CAJR7f2GR22MXbe8fr4W2KuDDEwaDVDN8i83X6wOSaJMA-g7qhQ@mail.gmail.com>
Message-ID: <D1275728.9837C%krice@freeswitch.org>

Is there a pull request on that?


On 3/12/15, 1:27 PM, "?talo Rossi" <italorossib at gmail.com> wrote:

> I set the JIRA status as Needs Review, hope it get merged soon.
> 
> On Thu, Mar 12, 2015 at 4:03 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>> ?talo I am not rewrite patch set use?network_addr in caller profile and path
>> not merget to master.
>> 
>> Sergey
>> 
>> On Thu, Mar 12, 2015 at 7:51 PM, ?talo Rossi <italorossib at gmail.com> wrote:
>>> 
>>> Version?
>>> 
>>> I'm almost sure this is already implemented in master.
>>> 
>>> Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com> escreveu:
>>>> Have you tried mod_fail2ban?
>>>> 
>>>> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de>
>>>> wrote:
>>>>>      Hello,
>>>>>  
>>>>>  we receive a number of Invites from certain IPs, who want to break into
>>>>> our system and call external premium rate numbers
>>>>>  Unwanted registers we can block already, but we still have the issue to
>>>>> block specific invites from fraudulent IPs inside the iptables firewall.
>>>>>  
>>>>>  In the Freeswitch log we see:
>>>>>  2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
>>>>> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>>>>>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/internal/149 at 10.11.12.13 [BREAK]
>>>>>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/internal/149 at 10.11.12.13 [BREAK]
>>>>>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472
>>>>> (sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>>>>>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>>>>> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076
>>>>> <http://155.94.64.26:5076>  version: 1.5.15b git 82f267a 2015-02-16
>>>>> 22:59:55Z 64bit
>>>>>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26 Rejected
>>>>> by acl "domains". Falling back to Digest auth.
>>>>>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491
>>>>> (sofia/internal/149 at 10.11.12.13) State NEW
>>>>>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/internal/149 at 10.11.12.13 [BREAK]
>>>>>  2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>>>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>>>>>  2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>>>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>>>>> Abandoned??? 
>>>>>  
>>>>>  The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
>>>>> 10.11.12.13 is the (anonymized) IP of our server.
>>>>>  
>>>>>  The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
>>>>> sends "authentication required". Freeswitch then logs this entry with
>>>>> "Abandoned" (see last line above) and that's it.
>>>>>  
>>>>>  So Is there any way to make Freeswitch show up a log line with the
>>>>> fraudulent IP 15.194.164.26 and some text like "abandonned"?
>>>>>  Example for extending a current log line
>>>>>  ??? 2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
>>>>> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
>>>>> Abandoned for IP 15.194.164.26
>>>>>  This would enable us to process this entry with fail2ban and block this
>>>>> IP in the Firewall.
>>>>>  
>>>>>  Any other hint is welcome.
>>>>>  

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


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/8253e051/attachment.html 

From s.safarov at gmail.com  Thu Mar 12 23:02:38 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 12 Mar 2015 23:02:38 +0300
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <D1275728.9837C%krice@freeswitch.org>
References: <CAJR7f2GR22MXbe8fr4W2KuDDEwaDVDN8i83X6wOSaJMA-g7qhQ@mail.gmail.com>
	<D1275728.9837C%krice@freeswitch.org>
Message-ID: <CAHtxdDeNi+_tM0kjX7DQDC+W5d0Gtmj0vcD1Ley_s18FmOGZvQ@mail.gmail.com>

Ken pull request has been created
https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/159/overview
Mike rightly said that it is necessary to use a variable network_addr in
caller profile


On Thu, Mar 12, 2015 at 11:46 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Is there a pull request on that?
>
>
> On 3/12/15, 1:27 PM, "?talo Rossi" <italorossib at gmail.com> wrote:
>
> I set the JIRA status as Needs Review, hope it get merged soon.
>
> On Thu, Mar 12, 2015 at 4:03 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
> ?talo I am not rewrite patch set use network_addr in caller profile and
> path not merget to master.
>
> Sergey
>
> On Thu, Mar 12, 2015 at 7:51 PM, ?talo Rossi <italorossib at gmail.com>
> wrote:
>
>
> Version?
>
> I'm almost sure this is already implemented in master.
>
> Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com> escreveu:
>
> Have you tried mod_fail2ban?
>
> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de>
> wrote:
>
>     Hello,
>
>  we receive a number of Invites from certain IPs, who want to break into
> our system and call external premium rate numbers
>  Unwanted registers we can block already, but we still have the issue to
> block specific invites from fraudulent IPs inside the iptables firewall.
>
>  In the Freeswitch log we see:
>  2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/149 at 10.11.12.13 [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/149 at 10.11.12.13) Running State Change CS_NEW
>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
> sofia/internal/149 at 10.11.12.13 receiving invite from 155.94.64.26:5076 <
> http://155.94.64.26:5076>  version: 1.5.15b git 82f267a 2015-02-16
> 22:59:55Z 64bit
>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26 Rejected
> by acl "domains". Falling back to Digest auth.
>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (
> sofia/internal/149 at 10.11.12.13) State NEW
>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/149 at 10.11.12.13 [BREAK]
>  2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>  2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned
>
>  The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP
> 10.11.12.13 is the (anonymized) IP of our server.
>
>  The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then
> sends "authentication required". Freeswitch then logs this entry with
> "Abandoned" (see last line above) and that's it.
>
>  So Is there any way to make Freeswitch show up a log line with the
> fraudulent IP 15.194.164.26 and some text like "abandonned"?
>  Example for extending a current log line
>      2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572
> 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13
> Abandoned for IP 15.194.164.26
>  This would enable us to process this entry with fail2ban and block this
> IP in the Firewall.
>
>  Any other hint is welcome.
>
>
>
> --
> 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
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/2d4aef7e/attachment.html 

From mike at jerris.com  Thu Mar 12 23:16:36 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 Mar 2015 16:16:36 -0400
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
	no challenge
In-Reply-To: <CAHtxdDeNi+_tM0kjX7DQDC+W5d0Gtmj0vcD1Ley_s18FmOGZvQ@mail.gmail.com>
References: <CAJR7f2GR22MXbe8fr4W2KuDDEwaDVDN8i83X6wOSaJMA-g7qhQ@mail.gmail.com>
	<D1275728.9837C%krice@freeswitch.org>
	<CAHtxdDeNi+_tM0kjX7DQDC+W5d0Gtmj0vcD1Ley_s18FmOGZvQ@mail.gmail.com>
Message-ID: <718D1C6D-6B6E-4200-897F-E66C6DE5784E@jerris.com>

I want to understand why that variable is not populated and see if it can be instead of adding a new channel variable, correct.

> On Mar 12, 2015, at 4:02 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Ken pull request has been created https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/159/overview <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/159/overview>
> Mike rightly said that it is necessary to use a variable network_addr in caller profile
> 
> 
> On Thu, Mar 12, 2015 at 11:46 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> Is there a pull request on that?
> 
> 
> On 3/12/15, 1:27 PM, "?talo Rossi" <italorossib at gmail.com <http://italorossib at gmail.com/>> wrote:
> 
> I set the JIRA status as Needs Review, hope it get merged soon.
> 
> On Thu, Mar 12, 2015 at 4:03 PM, Sergey Safarov <s.safarov at gmail.com <http://s.safarov at gmail.com/>> wrote:
> ?talo I am not rewrite patch set use network_addr in caller profile and path not merget to master.
> 
> Sergey
> 
> On Thu, Mar 12, 2015 at 7:51 PM, ?talo Rossi <italorossib at gmail.com <http://italorossib at gmail.com/>> wrote:
> 
> Version?
> 
> I'm almost sure this is already implemented in master. 
> 
> Em 12/03/2015 13:43, "Kyle King" <kyle.king at quentustech.com <http://kyle.king at quentustech.com/>> escreveu:
> Have you tried mod_fail2ban? 
> 
> On March 12, 2015 12:28:16 PM EDT, Peter Steinbach <lists at telefaks.de <http://lists at telefaks.de/>> wrote:
>     Hello,
>  
>  we receive a number of Invites from certain IPs, who want to break into our system and call external premium rate numbers
>  Unwanted registers we can block already, but we still have the issue to block specific invites from fraudulent IPs inside the iptables firewall.
>  
>  In the Freeswitch log we see:
>  2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>[167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>[BREAK]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>[BREAK]
>  2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>) Running State Change CS_NEW
>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841 sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>receiving invite from 155.94.64.26:5076 <http://155.94.64.26:5076/><http://155.94.64.26:5076 <http://155.94.64.26:5076/>>  version: 1.5.15b git 82f267a 2015-02-16 22:59:55Z 64bit
>  2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26 Rejected by acl "domains". Falling back to Digest auth.
>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>) State NEW
>  2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>[BREAK]
>  2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session 167bb9ee-c8d0-11e4-9f31-b39e581405c5
>  2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>Abandoned    
>  
>  The fraudulent IP here is 15.194.164.26 (anonymized of course). The IP 10.11.12.13 is the (anonymized) IP of our server.
>  
>  The point here is: 15.194.164.26 is sending an INVITE, Freeswitch then sends "authentication required". Freeswitch then logs this entry with "Abandoned" (see last line above) and that's it. 
>  
>  So Is there any way to make Freeswitch show up a log line with the fraudulent IP 15.194.164.26 and some text like "abandonned"?
>  Example for extending a current log line
>      2015-03-12 16:54:48.461568 [WARNING] switch_core_state_machine.c:572 167bb9ee-c8d0-11e4-9f31-b39e581405c5 sofia/internal/149 at 10.11.12.13 <http://sofia/internal/149 at 10.11.12.13>Abandoned for IP 15.194.164.26 
>  This would enable us to process this entry with fail2ban and block this IP in the Firewall.
>  
>  Any other hint is welcome.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/c4bcdb15/attachment-0001.html 

From msc at freeswitch.org  Thu Mar 12 23:28:18 2015
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 12 Mar 2015 13:28:18 -0700
Subject: [Freeswitch-users] Limit application usage in Lua
In-Reply-To: <CAABKsuTOGeB=UfovXeZvwLyjprFC97QdyJ4Wdf72ZLmsQd0YqQ@mail.gmail.com>
References: <CAABKsuTOGeB=UfovXeZvwLyjprFC97QdyJ4Wdf72ZLmsQd0YqQ@mail.gmail.com>
Message-ID: <082401d05d03$14556450$3d002cf0$@freeswitch.org>

Some context here would be helpful. Can you pastebin your Lua script, or at least the relevant lines that demonstrate what is happening? Also, when you say that the rest of the Lua dialplan is still processed, what does that mean? 

 

-MC

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Florent Krieg
Sent: Thursday, March 12, 2015 11:43 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Limit application usage in Lua

 

Hi all,

I wanted to implement call rate limits using this Lua instruction:
session:execute("limit", "hash " .. billedcaller .. " caps 10/1 !REQUESTED_CHAN_UNAVAIL")

It is actually working, but the rest of the Lua dialplan is still processed, which is a problem in my case.

I made a dirty but quick workaround, looking like this:
rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
if not rate_var then
    logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ", aborting the dialplan.")
    return
end

How can it be done properly?

Is there a way to be able to get the result of the 'limit' app call?

Or shall I check the status of the a-leg just after to decide to process the rest of the dialplan or not?

I know that this issue doesn't occur in a 'pure' XML dialplan, but I'm trying to find a solution for my Lua-only dialplan.

Thanks in advance if you have any idea.

I'm willing to try any possible solution you would think about!

Florent

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/e41301c4/attachment.html 

From msc at freeswitch.org  Thu Mar 12 23:41:06 2015
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 12 Mar 2015 13:41:06 -0700
Subject: [Freeswitch-users] Realtime sip registrations
In-Reply-To: <CACOYK=gYH8EWBmv9HkYnfbQ1tj5OUjPvnALEMa_GsrghO7AT7w@mail.gmail.com>
References: <CAK+KT39rnr8hvoXrM_eyjaBnmdQTToPJshZ7oWgOKk59KJQ5jw@mail.gmail.com>	<CAPJLwtW24Bzixx8s5OFYJM22D+7nF-MvQ_V=OajpnFm8SA=x=w@mail.gmail.com>	<CAPJLwtVkBwpuWX_CpKXUq9Ebbh8jY0qtbVhaHAufwibQeV_tLw@mail.gmail.com>
	<CACOYK=gYH8EWBmv9HkYnfbQ1tj5OUjPvnALEMa_GsrghO7AT7w@mail.gmail.com>
Message-ID: <09b601d05d04$ddfafc10$99f0f430$@freeswitch.org>

+1 to sngrep. I never actually did a write-up on it but I will say that I
can highly recommend it for quickly finding the right SIP dialog on a busy
system (or in a pcap file) and does the nice ladder graph of the
communications.
-MC

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Stanislav Sinyagin
Sent: Tuesday, March 10, 2015 2:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Realtime sip registrations

there's also a nice tool called sngrep
https://github.com/irontec/sngrep
It needs a wide terminal with color support, like xterm.



On Tue, Mar 10, 2015 at 7:23 AM, Tito Cumpen <soulofmischief87 at gmail.com>
wrote:
> If you have issues getting to the console. Use ngrep or sipgrep check 
> this wiki out https://wiki.freeswitch.org/wiki/Packet_Capture.
>
> On Mar 10, 2015 2:21 AM, "Tito Cumpen" <soulofmischief87 at gmail.com> wrote:
>>
>> Richard,
>>
>> You may view registrations through the fs_cli console. You can get 
>> very insightful debug ibformation through Sofia.check out Sofia debug 
>> http://wiki.freeswitch.org/wiki/Sofia-SIP
>>
>> On Mar 10, 2015 2:18 AM, "Richard Mace" <richard.mace at gmail.com> wrote:
>>>
>>> Hi All,
>>> Is it possible to see when sip registrations happen in real time?
>>>
>>> Thanks
>>>
>>> Richard
>>>
>>> ____________________________________________________________________
>>> _____ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
>>> sers
>>> http://www.freeswitch.org
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users 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://confluence.freeswitch.org
http://www.cluecon.com

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



From lists at telefaks.de  Fri Mar 13 00:24:25 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 12 Mar 2015 22:24:25 +0100
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
 no challenge
In-Reply-To: <CAFiqYunJZG7vM4NYM4GaxsVSAwkkh=9BcE=H9rsWctmy3_yYUg@mail.gmail.com>
References: <5501BEA0.4090708@telefaks.de>
	<CAFiqYunJZG7vM4NYM4GaxsVSAwkkh=9BcE=H9rsWctmy3_yYUg@mail.gmail.com>
Message-ID: <55020409.1020003@telefaks.de>

Thanks Steven

i've put stat into the internal and external profile now. I will later
look at the output.


On 03/12/15 18:36, Steven Ayre wrote:
> Check <param name="log-auth-failures" value="true"/> is on your sofia
> profiles. There should be a message when the challenge is sent and
> when the call is abandoned. It's not the
> 'Abandoned' switch_core_state_machine.c one but one coming from
> sofia_reg.c.
>
> On 12 March 2015 at 16:28, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>     we receive a number of Invites from certain IPs, who want to break
>     into our system and call external premium rate numbers
>     Unwanted registers we can block already, but we still have the
>     issue to block specific invites from fraudulent IPs inside the
>     iptables firewall.
>
>     In the Freeswitch log we see:
>     2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New
>     Channel sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>
>     [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472
>     (sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>) Running State Change CS_NEW
>     2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> receiving invite from
>     155.94.64.26:5076 <http://155.94.64.26:5076> version: 1.5.15b git
>     82f267a 2015-02-16 22:59:55Z 64bit
>     2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>     Rejected by acl "domains". Falling back to Digest auth.
>     2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491
>     (sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>) State NEW
>     2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     2015-03-12 16:54:48.461568 [WARNING]
>     switch_core_state_machine.c:572
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> Abandoned   
>
>     The fraudulent IP here is 15.194.164.26 (anonymized of course).
>     The IP 10.11.12.13 is the (anonymized) IP of our server.
>
>     The point here is: 15.194.164.26 is sending an INVITE, Freeswitch
>     then sends "authentication required". Freeswitch then logs this
>     entry with "Abandoned" (see last line above) and that's it.
>
>     So Is there any way to make Freeswitch show up a log line with the
>     fraudulent IP 15.194.164.26 and some text like "abandonned"?
>     Example for extending a current log line
>         2015-03-12 16:54:48.461568 [WARNING]
>     switch_core_state_machine.c:572
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> Abandoned for IP
>     15.194.164.26
>     This would enable us to process this entry with fail2ban and block
>     this IP in the Firewall.
>
>     Any other hint is welcome.
>
>     -- 
>     With kind regards
>     Marvin Keil 
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>
>     _________________________________________________________________________
>     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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/7259c88f/attachment-0001.html 

From lists at telefaks.de  Fri Mar 13 00:25:53 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 12 Mar 2015 22:25:53 +0100
Subject: [Freeswitch-users] Fail to ban rule for detecting INVITES with
 no challenge
In-Reply-To: <CAHtxdDejv19mdR2iozv3ciYPNX3OLGYwZXuW0njxtvKpspgYAg@mail.gmail.com>
References: <5501BEA0.4090708@telefaks.de>
	<CAHtxdDejv19mdR2iozv3ciYPNX3OLGYwZXuW0njxtvKpspgYAg@mail.gmail.com>
Message-ID: <55020461.9010503@telefaks.de>

Thanks Sergey

i've installed it. That was rather simple.

I will look at the output in some next hours.


On 03/12/15 19:44, Sergey Safarov wrote:
> Marvin you can use solution published
> at https://freeswitch.org/jira/browse/FS-7125
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/63a622decc0994d69a8e4ec223cb5359430f03d9
>
> Currently I successfully block that calls
>
> On Thu, Mar 12, 2015 at 7:28 PM, Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>     we receive a number of Invites from certain IPs, who want to break
>     into our system and call external premium rate numbers
>     Unwanted registers we can block already, but we still have the
>     issue to block specific invites from fraudulent IPs inside the
>     iptables firewall.
>
>     In the Freeswitch log we see:
>     2015-03-12 16:54:38.381552 [NOTICE] switch_channel.c:1055 New
>     Channel sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>
>     [167bb9ee-c8d0-11e4-9f31-b39e581405c5]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.381552 [DEBUG] switch_core_state_machine.c:472
>     (sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>) Running State Change CS_NEW
>     2015-03-12 16:54:38.381552 [DEBUG] sofia.c:8841
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> receiving invite from
>     155.94.64.26:5076 <http://155.94.64.26:5076> version: 1.5.15b git
>     82f267a 2015-02-16 22:59:55Z 64bit
>     2015-03-12 16:54:38.381552 [DEBUG] sofia.c:9008 IP 15.194.164.26
>     Rejected by acl "domains". Falling back to Digest auth.
>     2015-03-12 16:54:38.441582 [DEBUG] switch_core_state_machine.c:491
>     (sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13>) State NEW
>     2015-03-12 16:54:38.441582 [DEBUG] switch_core_session.c:1061 Send
>     signal sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> [BREAK]
>     2015-03-12 16:54:38.441582 [DEBUG] sofia.c:2067 detaching session
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     2015-03-12 16:54:48.461568 [WARNING]
>     switch_core_state_machine.c:572
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> Abandoned   
>
>     The fraudulent IP here is 15.194.164.26 (anonymized of course).
>     The IP 10.11.12.13 is the (anonymized) IP of our server.
>
>     The point here is: 15.194.164.26 is sending an INVITE, Freeswitch
>     then sends "authentication required". Freeswitch then logs this
>     entry with "Abandoned" (see last line above) and that's it.
>
>     So Is there any way to make Freeswitch show up a log line with the
>     fraudulent IP 15.194.164.26 and some text like "abandonned"?
>     Example for extending a current log line
>         2015-03-12 16:54:48.461568 [WARNING]
>     switch_core_state_machine.c:572
>     167bb9ee-c8d0-11e4-9f31-b39e581405c5
>     sofia/internal/149 at 10.11.12.13
>     <mailto:sofia/internal/149 at 10.11.12.13> Abandoned for IP
>     15.194.164.26
>     This would enable us to process this entry with fail2ban and block
>     this IP in the Firewall.
>
>     Any other hint is welcome.
>
>     -- 
>     With kind regards
>     Marvin Keil 
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>
>     _________________________________________________________________________
>     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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/4bf0e60b/attachment.html 

From godson.g at gmail.com  Fri Mar 13 01:28:15 2015
From: godson.g at gmail.com (Godson Gera)
Date: Fri, 13 Mar 2015 03:58:15 +0530
Subject: [Freeswitch-users] failover in freeswitch dialplan using | or
	lcr
In-Reply-To: <185CD03D8FD84FFD9C7BDCF1D7E21D7C@bob>
References: <185CD03D8FD84FFD9C7BDCF1D7E21D7C@bob>
Message-ID: <CAPTXd0rcNo0XeEPzn9XZKQRE3dF9mEE3S97Se5Ds+2d==sJhAg@mail.gmail.com>

It would be a lot easier to do it from a script (lua, python) running in FS
or via ESL. Idea is to extract all bridge strings returned by lcr command
and pass them to bridge command with | symbol for automatic fail over.

<action application="bridge" data="bridgestring1|bridgestring2"/>

On Thu, Mar 12, 2015 at 10:23 PM, Jason Holden <jason.holden at start.ca>
wrote:

>  Hi,
>
> If I send an invite for a call and don?t receive a response to the initial
> invite message how can I force FS to try the next route right away?
>
>
>
> Here is one of my lcr dialplan chunks.
>
>     <condition field="destination_number" expression="^(.*)$">
>
>       <action application="set" data="continue_on_fail=true"/>
>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="set" data="domain_name=$${domain}"/>
>
>       <action application="set" data="instant_ringback=true"/>
>
>       <action application="ring_ready" />
>
>       <action application="set" data="ringback=${us-ring}"/>
>
>       <action application="set" data="transfer-ringback=${us-ring}"/>
>
>       <action application="lcr" data="$1 inbound"/>
>
>       <action application="set" data="bypass_media=true"/>
>
>       <action application="bridge" data="${lcr_auto_route}"/>
>
>     </condition>
>
>   </extension>
>
>
>
> All advise is welcome, thanks.
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & Regards,
Godson Gera
VoIP consultant <http://godson.in>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/abb9946c/attachment.html 

From terry at digital-outpost.com  Fri Mar 13 01:52:32 2015
From: terry at digital-outpost.com (Terry Barnum)
Date: Thu, 12 Mar 2015 15:52:32 -0700
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <CAGpk5BQP+tbnc66_u6s7-4bREbvc1XA-2LUvkNnaD=2B1KYjKg@mail.gmail.com>
References: <66B1402D-A578-47F7-BE7F-96125726E797@digital-outpost.com>
	<CACOYK=hEJNNAoHgmzJcNu_28ocSHGK5KVR-omh3CNHJy_N1x1w@mail.gmail.com>
	<8938.1426060359@ccs.covici.com>
	<6A0C6E41-561F-4C12-859D-12C6B0FCD75E@digital-outpost.com>
	<CAGpk5BQP+tbnc66_u6s7-4bREbvc1XA-2LUvkNnaD=2B1KYjKg@mail.gmail.com>
Message-ID: <08E96A76-2EEA-491C-A125-2C858EE50642@digital-outpost.com>

Thank you Shabbir. John's suggestion of just emailing is working great. In the US, ATT uses <10digitnumber>@txt.att.net to send a text and Verizon uses <10digitnumber>@vtext.com.

-Terry

> On Mar 11, 2015, at 11:28 AM, Shabbir abbasi <shabbirabbasi92 at gmail.com> wrote:
> 
> i think your solution is in  mod_gsmopen  kindly read in wiki
> 
> On Wed, Mar 11, 2015 at 10:46 PM, Terry Barnum <terry at digital-outpost.com> wrote:
> Thank you John and Stanislav. I'll have a look at mod_voicemail.
> 
> -Terry
> 
> > On Mar 11, 2015, at 12:52 AM, covici at ccs.covici.com wrote:
> >
> > There is provision in mod_voicemail to send a paging Email and so I just
> > send it as a text message to the phone's email address.
> >
> > Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> >
> >> your best choice would be an SMS gateway provider which allows you to
> >> set arbitrary sender ID. Then you would buy a subscription there and
> >> send your SMS'es via their API. You could then receive SMS
> >> notifications with the sender ID equal to the original caller ID, and
> >> your phone will automatically look up the caller in the phonebook.
> >>
> >> Here in Switzerland I'm using http://www.inetworx.ch/  for this
> >> purpose. I believe threre should also be providers in your area.
> >>
> >> Then, making a hook in the mailer program is an easy task.
> >> mod_voicemail in Confluence gives an example of such a script in
> >> Python.
> >>
> >>
> >>
> >> On Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
> >> <terry at digital-outpost.com> wrote:
> >>> When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor.
> >>>
> >>> Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.)
> >>>
> >>> I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share?
> >>>
> >>> Thanks,
> >>> -Terry
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason.holden at start.ca  Fri Mar 13 04:14:42 2015
From: jason.holden at start.ca (Jason Holden)
Date: Thu, 12 Mar 2015 21:14:42 -0400
Subject: [Freeswitch-users] failover in freeswitch dialplan using | or
	lcr
In-Reply-To: <CAPTXd0rcNo0XeEPzn9XZKQRE3dF9mEE3S97Se5Ds+2d==sJhAg@mail.gmail.com>
References: <185CD03D8FD84FFD9C7BDCF1D7E21D7C@bob>
	<CAPTXd0rcNo0XeEPzn9XZKQRE3dF9mEE3S97Se5Ds+2d==sJhAg@mail.gmail.com>
Message-ID: <ED834D1BBFF24E658A56664E6FDE85C0@bob>

It does fail over with my setup but it takes about 30 seconds to try the
next route.

So back to the question would using call time out, or progress time out as
an action application before the lcr bridge resolve the issue, if so any
recommendations on timers?

 

Thanks.

  _____  

From: Godson Gera [mailto:godson.g at gmail.com] 
Sent: Thursday, March 12, 2015 6:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] failover in freeswitch dialplan using | or
lcr

 

It would be a lot easier to do it from a script (lua, python) running in FS
or via ESL. Idea is to extract all bridge strings returned by lcr command
and pass them to bridge command with | symbol for automatic fail over. 

 

<action application="bridge" data="bridgestring1|bridgestring2"/>

 

On Thu, Mar 12, 2015 at 10:23 PM, Jason Holden <jason.holden at start.ca>
wrote:

Hi,

If I send an invite for a call and don't receive a response to the initial
invite message how can I force FS to try the next route right away?

 

Here is one of my lcr dialplan chunks.

    <condition field="destination_number" expression="^(.*)$">

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

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

      <action application="set" data="domain_name=$${domain}"/>

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

      <action application="ring_ready" />

      <action application="set" data="ringback=${us-ring}"/>

      <action application="set" data="transfer-ringback=${us-ring}"/>  

      <action application="lcr" data="$1 inbound"/>

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

      <action application="bridge" data="${lcr_auto_route}"/>

    </condition>

  </extension>

 

All advise is welcome, thanks.

 

 

 

 


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 & Regards,

Godson Gera
VoIP consultant <http://godson.in> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/2a72dd59/attachment.html 

From anthony.minessale at gmail.com  Fri Mar 13 05:32:50 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Mar 2015 21:32:50 -0500
Subject: [Freeswitch-users] Values of read_codec/write_codec on
 different FS versions...
In-Reply-To: <CAMxcU5Ls8eBW0b2J7QKFcTe43uNf-QJk6SPyPhQbE+tDb5pWUw@mail.gmail.com>
References: <CAMxcU5+1b6CMXEQLFNBaf9=CYhG0VVEno=_CFhTeJ6A80dMQyw@mail.gmail.com>
	<01aa01d052ab$64fafd00$2ef0f700$@botecomm.com>
	<CAEJMVkB4N269Co==5Gk07O70=tFjMuNVC=pA-pGowvufKh7UkA@mail.gmail.com>
	<CAMxcU5JjZbEnR34rUeZjSnTLavE3Bp60R6bxGpb0Uvmm2vZBXA@mail.gmail.com>
	<CAEJMVkCWCq16HixTfsSinOG1UbH-K-yi=GS3Vuq0=vBuRBO+Mw@mail.gmail.com>
	<CAMxcU5+_aBi+V5mpO5728_H1pGPmet8Rmj2J_-MOh-xiQFGfHw@mail.gmail.com>
	<CAEJMVkAmuKCqkXsv=6CLY7OJLBC2mrmDyKJSW3e6CNPJcBGNKg@mail.gmail.com>
	<CAMxcU5Jk2=ROwCUNPzjmGHRRikCWdiX-=zQ--NrRhpTDj11erg@mail.gmail.com>
	<CAEJMVkD=oy1FRLB5mKYBGe4S3L5eAFSFEP-9_4eZmPj80q3wBQ@mail.gmail.com>
	<CAMxcU5Ls8eBW0b2J7QKFcTe43uNf-QJk6SPyPhQbE+tDb5pWUw@mail.gmail.com>
Message-ID: <CAKbxfG-uH3h9rj22AJ3C3BuDegx2LBBeT668Tr79xrbVZ2_a4Q@mail.gmail.com>

There was a bug on older one to not do the PROXY codec right.  Don't run
old versions. Its not wise.


On Wed, Mar 11, 2015 at 3:14 PM, Vladimir Getmanshchuk <vladget at gmail.com>
wrote:

> ok, just debug log or mod_cdr_csv debug?
>
> On Wed, Mar 4, 2015 at 12:00 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Without looking a the logs I can only guess that something is triggering
>> it...happen to have a debug log of this?
>>
>> On Tue, Mar 3, 2015 at 4:30 PM, Vladimir Getmanshchuk <vladget at gmail.com>
>> wrote:
>>
>>> As I have said before configs on both FS boxes are same. So Proxy-Media
>>> enabled on both FS boxes.
>>>
>>> On Tue, Mar 3, 2015 at 7:54 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> If you enable proxy media it will say proxy too....
>>>>
>>>>
>>>> On Tuesday, March 3, 2015, Vladimir Getmanshchuk <vladget at gmail.com>
>>>> wrote:
>>>>
>>>>> Same traffic balanced between these two FS boxes and all CDRs from
>>>>> 1.4.5 came with PROXY, but all CDRs from 1.4.5 came with real codec were in
>>>>> streams.
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Mar 2, 2015 at 12:30 AM, Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> ZRTP hash in the sdp will cause it to toggle on too!
>>>>>>
>>>>>>
>>>>>> On Saturday, February 28, 2015, Vladimir Getmanshchuk <
>>>>>> vladget at gmail.com> wrote:
>>>>>>
>>>>>>> Bote,
>>>>>>> When I said identical configuration I mean files at FS
>>>>>>> configuration directory.
>>>>>>> G.729 license? No, I use proxy-media mode with no transcoding.
>>>>>>>
>>>>>>> Brian,
>>>>>>> Both FS boxes configured for proxing media:
>>>>>>> # grep inbound-proxy-media /usr/local/freeswitch/conf
>>>>>>> /sip_profiles/internal.xml
>>>>>>> <param name="inbound-proxy-media" value="true"/>
>>>>>>>
>>>>>>> I do not understand why FS version 1.4.15 trying to hide actual
>>>>>>> read/write codecs and change it by "PROXY"?
>>>>>>>
>>>>>>> Thank you.
>>>>>>>
>>>>>>> On Fri, Feb 27, 2015 at 8:02 PM, Brian West <brian at freeswitch.org>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Someone's using Proxy Media mode... Thats why the codec says PROXY.
>>>>>>>>
>>>>>>>> On Fri, Feb 27, 2015 at 10:35 AM, Bote Man <bote_radio at botecomm.com
>>>>>>>> > wrote:
>>>>>>>>
>>>>>>>>> FS1> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>>>> FS2> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>>>
>>>>>>>>> I would say that these are not "absolutely" identical. As the
>>>>>>>>> FreeSWITCH
>>>>>>>>> development team never sleeps it is likely that there are
>>>>>>>>> differences in the
>>>>>>>>> code that you now see. The first thing is to bring both machines
>>>>>>>>> up to the
>>>>>>>>> same release before comparing behaviors.
>>>>>>>>>
>>>>>>>>> Another suggestion is to confirm your G.729 license and
>>>>>>>>> configuration, if
>>>>>>>>> you are decoding that codec. Perhaps one machine has the necessary
>>>>>>>>> file(s)
>>>>>>>>> in the correct locations and the other machine does not?
>>>>>>>>>
>>>>>>>>> Hope this helps.
>>>>>>>>>
>>>>>>>>> Bote
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: Vladimir Getmanshchuk
>>>>>>>>> Sent: Friday, 27 February, 2015 07:37
>>>>>>>>> Subject: [Freeswitch-users] Values of read_codec/write_codec on
>>>>>>>>> different FS
>>>>>>>>> versions...
>>>>>>>>>
>>>>>>>>> Hello Everyone!
>>>>>>>>>
>>>>>>>>> I have two installations of FS with absolutely identical
>>>>>>>>> configurations.
>>>>>>>>> Both has SIP profiles with proxy-media enabled.
>>>>>>>>>
>>>>>>>>> But on
>>>>>>>>> freeswitch at internal> version
>>>>>>>>> FreeSWITCH Version 1.4.15~64bit ( 64bit)
>>>>>>>>>
>>>>>>>>> I have values in read_codec/write_codec variables at CDRs:
>>>>>>>>> "read_codec":"PROXY","write_codec":"PROXY"
>>>>>>>>>
>>>>>>>>> but on another one
>>>>>>>>> freeswitch at internal> version
>>>>>>>>> FreeSWITCH Version 1.4.5~64bit ( 64bit)
>>>>>>>>>
>>>>>>>>> I have:
>>>>>>>>> "read_codec":"G729","write_codec":"G729",
>>>>>>>>> "read_codec":"PCMA","write_codec":"PCMA", etc...
>>>>>>>>>
>>>>>>>>> So Why?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Yours sincerely,
>>>>>>>>> Vladimir Getmanshchuk
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> *Brian West*
>>>>>>>> brian at freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>>> http://www.freeswitchbook.com
>>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>>
>>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Yours sincerely,
>>>>>>> Vladimir Getmanshchuk
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Brian West*
>>>>>> brian at freeswitch.org
>>>>>>
>>>>>>
>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>> http://www.freeswitchbook.com
>>>>>> http://www.freeswitchcookbook.com
>>>>>>
>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Yours sincerely,
>>>>> Vladimir Getmanshchuk
>>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Yours sincerely,
>>> Vladimir Getmanshchuk
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150312/ef2d02f3/attachment-0001.html 

From osblinnikov at gmail.com  Fri Mar 13 13:24:54 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Fri, 13 Mar 2015 11:24:54 +0100
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <D766B545-6A6C-4847-A741-3777BAD3FEF7@jerris.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
	<CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
	<29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>
	<CA+M05dCo87sc3kybYtz8u=PMdBAYZ2KNtMCWEECp6aZ0_E7bsw@mail.gmail.com>
	<D766B545-6A6C-4847-A741-3777BAD3FEF7@jerris.com>
Message-ID: <CA+M05dC9082GvyawJ868D-AYcBrVdBsFodvY4QqgeJ3p8H3ypQ@mail.gmail.com>

I tried to debug again and now I noticed that the audio flow actually goes
in both directions but still no video. I discovered in FreeSwitch log that
during successful video call from Chrome to Android all 4 fingerprints are
sha-256 and all of them are shown as "verified". But during unsuccessful
call from Android I get sha-1 (from Chrome still sha-256) and for some
reason FreeSwitch shows only 3 fingerprints verified. So I guess that
something is wrong with the video fingerprint verification from my Android
Application.

Logs are attached.

Thank you Michael!


On Thu, Mar 12, 2015 at 3:57 PM, Michael Jerris <mike at jerris.com> wrote:

> its not changing it back and forth to sha-256/sha-1 those are 2 different
> channels... the leg to android is always sha-1.  That being said, Nothing
> we have tested against uses sha-1.  That could be an issue.  You should
> look at the full debug log of the call between fs and android and see if
> there is anything useful there.
>
> On Mar 12, 2015, at 8:07 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
>
> Hi,
>
> Unfortunately not everything runs smoothly. I run FreeSWITCH Version
> 1.5.15b+git~20150203T210457Z~4174fb9cbe~64bit (git 4174fb9 2015-02-03
> 21:04:57Z 64bit) with the default configuration + webrtc module + tweaked
> bridge application for 1010 - 1019 extensions where I added
> ignore_early_media=true because of chrome troubles with pranswer and
> media_webrtc=true because one of my clients is actually JAIN SIP  without
> WebSockets.
>
> Now I call from Chrome to my Android app. In Android app I receive
> modified SDP from FreeSwitch and all the media traffic goes though
> FreeSwitch. I have the audio and video in both directions.
>
> But when I createOffer in the Android application and send it to Chrome
> the media is not flowing in any directions. In case I set "<action
> application="set" data="bypass_media=true"/>" media starts flowing.
>
> PS. May be it's irrelevant  but the only strange thing I noticed is that
> Android App produces fingerprint in sha-1 then FreeSwitch changes it to
> sha-256 and sends to Chrome. Chrome responds with sha-256 then FreeSwitch
> modifies it back to sha-1.
>
> I don't know, may be I forget about some other magic options?
>
> PS: SDPs are in the attachment
>
> On Fri, Mar 6, 2015 at 4:43 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Always nice to hear that we are magic!
>>
>> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
>>
>> thank you very much Michael, it magically works.
>>
>> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> you need to tell freeswitch to send a webrtc compatible SDP.
>>>
>>> https://wiki.freeswitch.org/wiki/Variable_media_webrtc
>>>
>>>
>>> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com>
>>> wrote:
>>>
>>> Hi,
>>>
>>> I've made a simple Android Java application utilizing JAIN SIP,
>>> webrtc.org android library and connected to FreeSwitch via UDP.
>>>
>>> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client
>>> this SDP is not managed well by FreeSwitch for establishment WebRTC
>>> PeerConnection.
>>>
>>> When I call `peerConnection.setRemoteDescription(new SDPObserver(),
>>> sdp);` in my Android Application with the SDP from FreeSwitch I get:
>>>
>>> "onSetFailure Failed to set remote offer sdp: Called with SDP without
>>> DTLS fingerprint."
>>>
>>> At the same time the calls between Chrome/Firefox(
>>> http://tryit.jssip.net/) and SIP-phone (e.g. linphone) greatly managed
>>> by FreeSwitch and I have pure audio flow.
>>>
>>> Here is initial SDP from Chrome (http://tryit.jssip.net/):
>>>
>>> v=0
>>> o=- 6887715720880489867 2 IN IP4 127.0.0.1
>>> s=-
>>> t=0 0
>>> a=group:BUNDLE audio video
>>> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
>>> c=IN IP4 192.168.122.1
>>> a=rtcp:38359 IN IP4 192.168.122.1
>>> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
>>> generation 0
>>> .......
>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
>>> active generation 0
>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>> a=ice-options:google-ice
>>> a=*fingerprint*:sha-256
>>> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>> a=setup:actpass
>>> a=mid:audio
>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>> a=sendrecv
>>> a=rtcp-mux
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10
>>> a=rtpmap:103 ISAC/16000
>>> a=rtpmap:104 ISAC/32000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:106 CN/32000
>>> a=rtpmap:105 CN/16000
>>> a=rtpmap:13 CN/8000
>>> a=rtpmap:126 telephone-event/8000
>>> a=maxptime:60
>>> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
>>> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>> 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>> m=video 38359 RTP/SAVPF 100 116 117 96
>>> c=IN IP4 192.168.122.1
>>> a=rtcp:38359 IN IP4 192.168.122.1
>>> a=candidate:4062413514 1 udp 2122260223 192.168.122.1 38359 typ host
>>> generation 0
>>> ............
>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype
>>> active generation 0
>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>> a=ice-options:google-ice
>>> a=*fingerprint*:sha-256
>>> 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>> a=setup:actpass
>>> a=mid:video
>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>> a=recvonly
>>> a=rtcp-mux
>>> a=rtpmap:100 VP8/90000
>>> a=rtcp-fb:100 ccm fir
>>> a=rtcp-fb:100 nack
>>> a=rtcp-fb:100 nack pli
>>> a=rtcp-fb:100 goog-remb
>>> a=rtpmap:116 red/90000
>>> a=rtpmap:117 ulpfec/90000
>>> a=rtpmap:96 rtx/90000
>>> a=fmtp:96 apt=100
>>>
>>>
>>> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>>>
>>> v=0
>>> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.131.253
>>> t=0 0
>>> m=audio 16390 RTP/AVP 111 0 8 101 13
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> m=video 16388 RTP/AVP 100
>>> a=rtpmap:100 VP8/90000
>>>
>>>
>>> I suppose that FreeSwitch wants to see WebRTC connection only on the
>>> WebSocket ports and it doesn't know that my UDP client is actually WebRTC
>>> client.
>>>
>>> So I'm wondering if it possible to connect SIP client to the WebSocket
>>> port via TCP using standard SIP client and never upgrade connection to
>>> WebSocket?
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Oleg Blinnikov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/266bbc72/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: fail
Type: application/octet-stream
Size: 98240 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/266bbc72/attachment-0002.obj 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: success
Type: application/octet-stream
Size: 103599 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/266bbc72/attachment-0003.obj 

From godson.g at gmail.com  Fri Mar 13 14:09:50 2015
From: godson.g at gmail.com (Godson Gera)
Date: Fri, 13 Mar 2015 16:39:50 +0530
Subject: [Freeswitch-users] failover in freeswitch dialplan using | or
	lcr
In-Reply-To: <ED834D1BBFF24E658A56664E6FDE85C0@bob>
References: <185CD03D8FD84FFD9C7BDCF1D7E21D7C@bob>
	<CAPTXd0rcNo0XeEPzn9XZKQRE3dF9mEE3S97Se5Ds+2d==sJhAg@mail.gmail.com>
	<ED834D1BBFF24E658A56664E6FDE85C0@bob>
Message-ID: <CAPTXd0osode2nQfpbABd2gzHD801ynB8OnTT_=bjzx5GqEMgyg@mail.gmail.com>

There are many timeout variables available to control the call behavior
like call_timeout,leg_timeout,bridge_answer_timeout etc. Check out this
page https://freeswitch.org/confluence/display/FREESWITCH/Variables . Try
them out and pick the one suits your needs

On Fri, Mar 13, 2015 at 6:44 AM, Jason Holden <jason.holden at start.ca> wrote:

>  It does fail over with my setup but it takes about 30 seconds to try the
> next route.
>
> So back to the question would using call time out, or progress time out as
> an action application before the lcr bridge resolve the issue, if so any
> recommendations on timers?
>
>
>
> Thanks.
>   ------------------------------
>
> *From:* Godson Gera [mailto:godson.g at gmail.com]
> *Sent:* Thursday, March 12, 2015 6:28 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] failover in freeswitch dialplan using |
> or lcr
>
>
>
> It would be a lot easier to do it from a script (lua, python) running in
> FS or via ESL. Idea is to extract all bridge strings returned by lcr
> command and pass them to bridge command with | symbol for automatic fail
> over.
>
>
>
> <action application="bridge" data="bridgestring1|bridgestring2"/>
>
>
>
> On Thu, Mar 12, 2015 at 10:23 PM, Jason Holden <jason.holden at start.ca>
> wrote:
>
> Hi,
>
> If I send an invite for a call and don?t receive a response to the initial
> invite message how can I force FS to try the next route right away?
>
>
>
> Here is one of my lcr dialplan chunks.
>
>     <condition field="destination_number" expression="^(.*)$">
>
>       <action application="set" data="continue_on_fail=true"/>
>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="set" data="domain_name=$${domain}"/>
>
>       <action application="set" data="instant_ringback=true"/>
>
>       <action application="ring_ready" />
>
>       <action application="set" data="ringback=${us-ring}"/>
>
>       <action application="set" data="transfer-ringback=${us-ring}"/>
>
>       <action application="lcr" data="$1 inbound"/>
>
>       <action application="set" data="bypass_media=true"/>
>
>       <action application="bridge" data="${lcr_auto_route}"/>
>
>     </condition>
>
>   </extension>
>
>
>
> All advise is welcome, thanks.
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & Regards,
>
> Godson Gera
> VoIP consultant <http://godson.in>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & Regards,
Godson Gera
FreeSWITCH Consultant <http://godson.in>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/395c56e9/attachment.html 

From flokrrr at gmail.com  Fri Mar 13 14:31:06 2015
From: flokrrr at gmail.com (Florent Krieg)
Date: Fri, 13 Mar 2015 12:31:06 +0100
Subject: [Freeswitch-users] Limit application usage in Lua
In-Reply-To: <082401d05d03$14556450$3d002cf0$@freeswitch.org>
References: <CAABKsuTOGeB=UfovXeZvwLyjprFC97QdyJ4Wdf72ZLmsQd0YqQ@mail.gmail.com>
	<082401d05d03$14556450$3d002cf0$@freeswitch.org>
Message-ID: <CAABKsuQ3d8RALJDs_BqVpLsp_2ekFCFExQT5ynC7t8vyo3BuMg@mail.gmail.com>

Hi Michael,

Actually, here is what happens:

Context: XML dialplan with the following extensions:
        <extension name="limit_exceeded">
            <condition field="destination_number"
expression="^limit_exceeded$">
                <action application="respond" data="503"/>
            </condition>
        </extension>
        <extension name="outgoing_call">
            <condition>
                <action application="lua"
data="dialplans/caller_script.lua"/>
            </condition>
        </extension>

1/ a call is started, the second extension entry in the XML dialplan
matches and the action starts the lua script
2/ after having set some (a lot actually) variables, here is the code part
for checking call-rate limits:
-- Security (to avoid outage): limit CAPS
--   hash backend is needed when using interval
--   2 call attempts per 10 seconds here
logz("[Calls limit] Checking caps for " .. billedcaller)
session:execute("limit", "hash " .. billedcaller .. " caps 2/10
!REQUESTED_CHAN_UNAVAIL")
rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
if not rate_var then
    logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
aborting the dialplan.")
    return
end
3/ after this code part, a lot of processing is done (for instance checking
call restrictions calling external web services and so on), it's like 600
lines of lua there
4/ in the end, the session:execute bridge app is run

At 2/, when limit is exceeded, I see in the log:
2015-03-13 12:22:40.454660 [INFO] switch_cpp.cpp:1328 * (caller/0) #
7a16c525263a8bb740517f5a28552016 at trunkinge.voip * [Calls limit] Checking
caps for asteriskrd at trunkinge.voip
EXECUTE sofia/internal_auth/asteriskrd at trunkinge.voip limit(hash
asteriskrd at trunkinge.voip caps 2/10 !REQUESTED_CHAN_UNAVAIL)
2015-03-13 12:22:40.454660 [DEBUG] switch_limit.c:126 incr called:
asteriskrd at trunkinge.voip_caps max:2, interval:10
2015-03-13 12:22:40.454660 [INFO] mod_hash.c:176 Usage for
asteriskrd at trunkinge.voip_caps exceeds maximum rate of 2/10s, now at 3
2015-03-13 12:22:40.454660 [INFO] switch_cpp.cpp:1328 * (caller/0) #
7a16c525263a8bb740517f5a28552016 at trunkinge.voip * [Calls limit] CAPS limit
reached for asteriskrd at trunkinge.voip, aborting the dialplan.

and the a-leg is successfully hangup.

But here, 3/ and 4/ are aborting only because of that code part:
rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
if not rate_var then
    logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
aborting the dialplan.")
    return
end

It works, but I'm pretty sure that it is not the proper way to do it.

If I remove that code part, even though the a-leg is hangup because of the
leg being transferred to 'limit_exceeded', the rest of the dialplan is
still processed, even though the call is aborted.

I'd like to be able to catch right after doing session:execute limit app,
the return of the command.
If the call is transferred to 'limit_exceeded', then I'd like to abort the
rest of the dialplan execution.

I have to admit I might be missing something here therefore if you can help
me, that'd be amazing.
Thanks in advance
Florent


2015-03-12 21:28 GMT+01:00 Michael S Collins <msc at freeswitch.org>:

> Some context here would be helpful. Can you pastebin your Lua script, or
> at least the relevant lines that demonstrate what is happening? Also, when
> you say that the rest of the Lua dialplan is still processed, what does
> that mean?
>
>
>
> -MC
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Florent
> Krieg
> *Sent:* Thursday, March 12, 2015 11:43 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Limit application usage in Lua
>
>
>
> Hi all,
>
> I wanted to implement call rate limits using this Lua instruction:
> session:execute("limit", "hash " .. billedcaller .. " caps 10/1
> !REQUESTED_CHAN_UNAVAIL")
>
> It is actually working, but the rest of the Lua dialplan is still
> processed, which is a problem in my case.
>
> I made a dirty but quick workaround, looking like this:
> rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
> if not rate_var then
>     logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
> aborting the dialplan.")
>     return
> end
>
> How can it be done properly?
>
> Is there a way to be able to get the result of the 'limit' app call?
>
> Or shall I check the status of the a-leg just after to decide to process
> the rest of the dialplan or not?
>
> I know that this issue doesn't occur in a 'pure' XML dialplan, but I'm
> trying to find a solution for my Lua-only dialplan.
>
> Thanks in advance if you have any idea.
>
> I'm willing to try any possible solution you would think about!
>
> Florent
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/585787ad/attachment-0001.html 

From aademattia at Comcast.net  Fri Mar 13 16:04:51 2015
From: aademattia at Comcast.net (Andrew)
Date: Fri, 13 Mar 2015 09:04:51 -0400
Subject: [Freeswitch-users] illegal instruction
Message-ID: <00e401d05d8e$4eb2bc20$ec183460$@Comcast.net>

Thanks,

I looked and its set to "NOT SET".

I do know the CPU is a AMD that is having an issue.  Not sure if the other
server is an Intel or not.

 

Andrew

 

From: Andrew [mailto:aademattia at Comcast.net] 
Sent: Sunday, March 8, 2015 3:36 PM
To: 'FreeSWITCH Users Help'
Subject: illegal instruction

 

Hi,

I have an odd issue.  I was going to run out of the box FreeSWITCH on a
windows server and just by double clicking on freeswitch.exe I get a crash.
When I did a debug I found it was illegal instruction error.

 

If I remove mod_spandsp The program starts up fine.  I then try to do a tone
detection and then the program crashes again.  What would cause the same
release code to work on one server but crash on another server?

 

Andrew

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/7406086b/attachment.html 

From tfred31 at yahoo.com  Fri Mar 13 16:23:52 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Fri, 13 Mar 2015 06:23:52 -0700
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <08E96A76-2EEA-491C-A125-2C858EE50642@digital-outpost.com>
Message-ID: <1426253032.3130.YahooMailBasic@web160205.mail.bf1.yahoo.com>

Just a follow-up on sending Text messages through the email system.

If you are building a new contact list you might benefit from knowing which text gateway to use for which cell phone number.
I do this by using one of the various web reference pages - the one I use mostly is:   http://www.freecarrierlookup.com/

Next you need to know the text gateway domain for that cell phone carrier.
Again there are a number of references on the web - the one I use mostly is:   http://martinfitzpatrick.name/list-of-email-to-sms-gateways/

Also note that there is a difference between SMS and MMS
   SMS you are most likely already familiar with
   MMS is for more lengthy text messages
Often (according to the text gateway reference page) these two different 'formats' utilize different text gateways.

TF


--------------------------------------------
On Thu, 3/12/15, Terry Barnum <terry at digital-outpost.com> wrote:

 Subject: Re: [Freeswitch-users] FS voicemail notification to mobile SMS?
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Thursday, March 12, 2015, 4:52 PM
 
 Thank you Shabbir.
 John's suggestion of just emailing is working great. In
 the US, ATT uses <10digitnumber>@txt.att.net to send a
 text and Verizon uses <10digitnumber>@vtext.com.
 
 -Terry
 
 > On Mar 11, 2015, at 11:28 AM, Shabbir
 abbasi <shabbirabbasi92 at gmail.com>
 wrote:
 > 
 > i think
 your solution is in? mod_gsmopen? kindly read in wiki
 > 
 > On Wed, Mar 11, 2015
 at 10:46 PM, Terry Barnum <terry at digital-outpost.com>
 wrote:
 > Thank you John and Stanislav.
 I'll have a look at mod_voicemail.
 >
 
 > -Terry
 > 
 > > On Mar 11, 2015, at 12:52 AM, covici at ccs.covici.com
 wrote:
 > >
 > >
 There is provision in mod_voicemail to send a paging Email
 and so I just
 > > send it as a text
 message to the phone's email address.
 > >
 > > Stanislav
 Sinyagin <ssinyagin at gmail.com>
 wrote:
 > >
 >
 >> your best choice would be an SMS gateway provider
 which allows you to
 > >> set
 arbitrary sender ID. Then you would buy a subscription there
 and
 > >> send your SMS'es via
 their API. You could then receive SMS
 >
 >> notifications with the sender ID equal to the
 original caller ID, and
 > >> your
 phone will automatically look up the caller in the
 phonebook.
 > >>
 > >> Here in Switzerland I'm using
 http://www.inetworx.ch/  for this
 > >> purpose. I believe threre should
 also be providers in your area.
 >
 >>
 > >> Then, making a hook
 in the mailer program is an easy task.
 >
 >> mod_voicemail in Confluence gives an example of
 such a script in
 > >> Python.
 > >>
 > >>
 > >>
 > >> On
 Wed, Mar 11, 2015 at 12:30 AM, Terry Barnum
 > >> <terry at digital-outpost.com>
 wrote:
 > >>> When someone leaves
 a voicemail at the house, FS + FusionPBX successfully emails
 the Google transcription along with the audio attachment of
 the voicemail. It's working well. High Spousal Approval
 Factor.
 > >>>
 > >>> Now I would like to send a
 SMS text message notification via cellular to our mobile
 phones when a voicemail is left. Possible? I don't
 necessarily need the audio of the voicemail just a
 notification with the caller id in the SMS. (Though if
 it's easy I'll try it.)
 >
 >>>
 > >>> I've
 looked at the mod_sms wiki docs but all the examples appear
 to be between FS clients. How to send out to a cellular
 mobile number? Anyone have working examples they can
 share?
 > >>>
 > >>> Thanks,
 >
 >>> -Terry
 > >>>
 _________________________________________________________________________
 > >>> Professional FreeSWITCH
 Consulting Services:
 > >>> consulting at freeswitch.org
 > >>> http://www.freeswitchsolutions.com
 > >>>
 >
 >>> Official FreeSWITCH Sites
 >
 >>> http://www.freeswitch.org
 > >>> http://confluence.freeswitch.org
 > >>> http://www.cluecon.com
 > >>>
 >
 >>> FreeSWITCH-users mailing list
 > >>> FreeSWITCH-users at lists.freeswitch.org
 > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >>> http://www.freeswitch.org
 > >>
 > >>
 _________________________________________________________________________
 > >> Professional FreeSWITCH
 Consulting Services:
 > >> consulting at freeswitch.org
 > >> http://www.freeswitchsolutions.com
 > >>
 > >>
 Official FreeSWITCH Sites
 > >> http://www.freeswitch.org
 > >> http://confluence.freeswitch.org
 > >> http://www.cluecon.com
 > >>
 > >>
 FreeSWITCH-users mailing list
 > >>
 FreeSWITCH-users at 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
 > >
 > >
 _________________________________________________________________________
 > > Professional FreeSWITCH Consulting
 Services:
 > > consulting at freeswitch.org
 > > http://www.freeswitchsolutions.com
 > >
 > > Official
 FreeSWITCH Sites
 > > http://www.freeswitch.org
 > > http://confluence.freeswitch.org
 > > http://www.cluecon.com
 > >
 > >
 FreeSWITCH-users mailing list
 > > FreeSWITCH-users at lists.freeswitch.org
 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > > http://www.freeswitch.org
 > >
 > 
 > 
 >
 _________________________________________________________________________
 > Professional FreeSWITCH Consulting
 Services:
 > consulting at freeswitch.org
 > http://www.freeswitchsolutions.com
 > 
 > Official FreeSWITCH
 Sites
 > http://www.freeswitch.org
 > http://confluence.freeswitch.org
 > http://www.cluecon.com
 > 
 > FreeSWITCH-users
 mailing list
 > FreeSWITCH-users at lists.freeswitch.org
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 > 
 >
 _________________________________________________________________________
 > Professional FreeSWITCH Consulting
 Services: 
 > consulting at freeswitch.org
 > http://www.freeswitchsolutions.com
 > 
 > Official FreeSWITCH
 Sites
 > http://www.freeswitch.org
 > http://confluence.freeswitch.org
 > http://www.cluecon.com
 > 
 > FreeSWITCH-users
 mailing list
 > FreeSWITCH-users at lists.freeswitch.org
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 


From is.yaltunay at gmail.com  Fri Mar 13 16:33:43 2015
From: is.yaltunay at gmail.com (=?UTF-8?Q?Y=C3=BCcel_ALTUNAY?=)
Date: Fri, 13 Mar 2015 15:33:43 +0200
Subject: [Freeswitch-users] Using Freeswitch with TLS
Message-ID: <CANT=1X0vNkQsPZQz99iF8Bf+njJZ5JidOdGnqirbBskBWhC4_g@mail.gmail.com>

Hi,
I want to use Freeswitch with TLS and i want to learn which modules i need
mandatory.
i want to remove modules the i dont need.
can you list modules here or write any web page link?
Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/6244d4d2/attachment.html 

From ppyy at pubyun.com  Fri Mar 13 16:53:24 2015
From: ppyy at pubyun.com (=?UTF-8?B?5b2t5YuH?=)
Date: Fri, 13 Mar 2015 21:53:24 +0800
Subject: [Freeswitch-users] failover in inbound event socket mode
Message-ID: <CAB4GJpXpJm+sAbjUfWVcGEtGy_zMqQuaOYTA3u+wPbSR46LB6w@mail.gmail.com>

we would like to fail over in inbound event socket:

dial in -> park
origin -> park
uuid_bridge

originate
{continue_on_fail=^^:NO_USER_RESPONSE:UNALLOCATED_NUMBER:NORMAL_TEMPORARY_F
AILURE:NO_ROUTE_DESTINATION:CALL_REJECTED:RECOVERY_ON_TIMER_EXPIRE:NETWORK_OUT_OF_ORDER,fail_on_single_reject=^^:USER_BUSY:NO_ANSWER:ORIGINATOR_CANCEL}sofia/gateway/gw01/xxxx|sofia/gateway/gw02/xxxx
&park()

it will not hangup if the first gateway fails with ORIGINATOR_CANCEL.

how can set continue_on_fail and fail_on_single_reject?

-- 
Peng Yong
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/1ec9b575/attachment.html 

From krice at freeswitch.org  Fri Mar 13 17:00:43 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 13 Mar 2015 14:00:43 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <5502ed8c1d4_373b80d33823811@resque-worker-ip-10-231-192-240.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/640ad115/attachment.html 

From godson.g at gmail.com  Fri Mar 13 17:22:06 2015
From: godson.g at gmail.com (Godson Gera)
Date: Fri, 13 Mar 2015 19:52:06 +0530
Subject: [Freeswitch-users] bridge calls in event socket app
In-Reply-To: <CAAwzf1=5dVjMKouOoL_ijh3Gk7L0EEL=m1trnWnbje-W0rwoCA@mail.gmail.com>
References: <CAAwzf1=5dVjMKouOoL_ijh3Gk7L0EEL=m1trnWnbje-W0rwoCA@mail.gmail.com>
Message-ID: <CAPTXd0qdxeBjLLv-NfSK6GsjU=7dMbpEz-kS8dUk-OmJjo0upQ@mail.gmail.com>

Could you be more specific on what you are trying to do ? If leg A is
executing bridge command either in evensocket or in dialplan then they will
be automatically be bridged by FS unless you have set ignore_early_media
channel variable.

On Wed, Mar 11, 2015 at 6:52 AM, sukitha jayasinghe <sukithaj at gmail.com>
wrote:

> I want bridge two unanswered calls (in eventsocke) to listen to early
> media. What is the mechanism for that?
>
> Best Regards,
> Sukitha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 & Regards,
Godson Gera
VoIP Consultant <http://godson.in>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/b2433e14/attachment-0001.html 

From mike at jerris.com  Fri Mar 13 17:37:44 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 13 Mar 2015 10:37:44 -0400
Subject: [Freeswitch-users] JAVA & WebRTC & JAIN SIP - no WebSockets
In-Reply-To: <CA+M05dC9082GvyawJ868D-AYcBrVdBsFodvY4QqgeJ3p8H3ypQ@mail.gmail.com>
References: <CA+M05dAvuCsM8tL7Hawu1P26_TRe8RjnPAhsPGNKypxf08aEMw@mail.gmail.com>
	<0E66DE9E-31DF-4812-8382-769E6AD4CD37@jerris.com>
	<CA+M05dCG6TmXK6i4KexJjkJRj6obXBG_47jdRHfzz=3-aHh-vA@mail.gmail.com>
	<29F6208C-4437-4DD3-B6F0-0AE621CED630@jerris.com>
	<CA+M05dCo87sc3kybYtz8u=PMdBAYZ2KNtMCWEECp6aZ0_E7bsw@mail.gmail.com>
	<D766B545-6A6C-4847-A741-3777BAD3FEF7@jerris.com>
	<CA+M05dC9082GvyawJ868D-AYcBrVdBsFodvY4QqgeJ3p8H3ypQ@mail.gmail.com>
Message-ID: <AC9B9608-D150-4D08-8E61-4D2C34D9C9BD@jerris.com>

To be completely honest, I don't think anyone is going to look at this issue.  We try hard to make sure we work with the browsers, but have made no effort to work on interop with other devices, particularly ones that deviate behaviors with the browsers.  If you can figure it out we'd be happy to review any patches.  Might try talking to the JAIN guys and see why they are using sha-1 ?

Mike

> On Mar 13, 2015, at 6:24 AM, Oleg Blinnikov <osblinnikov at gmail.com> wrote:
> 
> I tried to debug again and now I noticed that the audio flow actually goes in both directions but still no video. I discovered in FreeSwitch log that during successful video call from Chrome to Android all 4 fingerprints are sha-256 and all of them are shown as "verified". But during unsuccessful call from Android I get sha-1 (from Chrome still sha-256) and for some reason FreeSwitch shows only 3 fingerprints verified. So I guess that something is wrong with the video fingerprint verification from my Android Application.
> 
> Logs are attached.
> 
> Thank you Michael!
> 
> 
> On Thu, Mar 12, 2015 at 3:57 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> its not changing it back and forth to sha-256/sha-1 those are 2 different channels... the leg to android is always sha-1.  That being said, Nothing we have tested against uses sha-1.  That could be an issue.  You should look at the full debug log of the call between fs and android and see if there is anything useful there.
> 
>> On Mar 12, 2015, at 8:07 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>> 
>> Hi,
>> 
>> Unfortunately not everything runs smoothly. I run FreeSWITCH Version 1.5.15b+git~20150203T210457Z~4174fb9cbe~64bit (git 4174fb9 2015-02-03 21:04:57Z 64bit) with the default configuration + webrtc module + tweaked bridge application for 1010 - 1019 extensions where I added ignore_early_media=true because of chrome troubles with pranswer and media_webrtc=true because one of my clients is actually JAIN SIP  without WebSockets.
>> 
>> Now I call from Chrome to my Android app. In Android app I receive modified SDP from FreeSwitch and all the media traffic goes though FreeSwitch. I have the audio and video in both directions. 
>> 
>> But when I createOffer in the Android application and send it to Chrome the media is not flowing in any directions. In case I set "<action application="set" data="bypass_media=true"/>" media starts flowing.
>> 
>> PS. May be it's irrelevant  but the only strange thing I noticed is that Android App produces fingerprint in sha-1 then FreeSwitch changes it to sha-256 and sends to Chrome. Chrome responds with sha-256 then FreeSwitch modifies it back to sha-1.
>> 
>> I don't know, may be I forget about some other magic options? 
>> 
>> PS: SDPs are in the attachment
>> 
>> On Fri, Mar 6, 2015 at 4:43 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> Always nice to hear that we are magic!  
>> 
>>> On Mar 6, 2015, at 5:05 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>>> 
>>> thank you very much Michael, it magically works.
>>> 
>>> On Thu, Mar 5, 2015 at 6:51 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> you need to tell freeswitch to send a webrtc compatible SDP.
>>> 
>>> https://wiki.freeswitch.org/wiki/Variable_media_webrtc <https://wiki.freeswitch.org/wiki/Variable_media_webrtc>
>>> 
>>> 
>>>> On Mar 5, 2015, at 3:51 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
>>>> 
>>>> Hi,
>>>> 
>>>> I've made a simple Android Java application utilizing JAIN SIP, webrtc.org <http://webrtc.org/> android library and connected to FreeSwitch via UDP.
>>>> 
>>>> But when I send SDP from SIP/Chrome/Firefox phone to my JAIN SIP client this SDP is not managed well by FreeSwitch for establishment WebRTC PeerConnection.
>>>> 
>>>> When I call `peerConnection.setRemoteDescription(new SDPObserver(), sdp);` in my Android Application with the SDP from FreeSwitch I get:
>>>> 
>>>> "onSetFailure Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
>>>> 
>>>> At the same time the calls between Chrome/Firefox(http://tryit.jssip.net/ <http://tryit.jssip.net/>) and SIP-phone (e.g. linphone) greatly managed by FreeSwitch and I have pure audio flow.
>>>> 
>>>> Here is initial SDP from Chrome (http://tryit.jssip.net/ <http://tryit.jssip.net/>):
>>>> 
>>>> v=0
>>>> o=- 6887715720880489867 2 IN IP4 127.0.0.1
>>>> s=-
>>>> t=0 0
>>>> a=group:BUNDLE audio video
>>>> a=msid-semantic: WMS itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>>> m=audio 38359 RTP/SAVPF 111 103 104 0 8 106 105 13 126
>>>> c=IN IP4 192.168.122.1
>>>> a=rtcp:38359 IN IP4 192.168.122.1
>>>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>>>> .......
>>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>>> a=ice-options:google-ice
>>>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>>> a=setup:actpass
>>>> a=mid:audio
>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>>> a=sendrecv
>>>> a=rtcp-mux
>>>> a=rtpmap:111 opus/48000/2
>>>> a=fmtp:111 minptime=10
>>>> a=rtpmap:103 ISAC/16000
>>>> a=rtpmap:104 ISAC/32000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:106 CN/32000
>>>> a=rtpmap:105 CN/16000
>>>> a=rtpmap:13 CN/8000
>>>> a=rtpmap:126 telephone-event/8000
>>>> a=maxptime:60
>>>> a=ssrc:1291334905 cname:ALccmKLk9bGpSGWB
>>>> a=ssrc:1291334905 msid:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU 4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>>> a=ssrc:1291334905 mslabel:itEKr0vXP6lg3KNs4kVau9aL3uAfyWOlItfU
>>>> a=ssrc:1291334905 label:4e8f212e-746a-47bb-bc62-4a42d4e9e84e
>>>> m=video 38359 RTP/SAVPF 100 116 117 96
>>>> c=IN IP4 192.168.122.1
>>>> a=rtcp:38359 IN IP4 192.168.122.1
>>>> a=candidate:4062413514 <tel:4062413514> 1 udp 2122260223 <tel:2122260223> 192.168.122.1 38359 typ host generation 0
>>>> ............
>>>> a=candidate:3741779331 2 tcp 1518018303 172.17.42.1 0 typ host tcptype active generation 0
>>>> a=ice-ufrag:bwrCv9yS8rCY12Az
>>>> a=ice-pwd:3k35jpG/i+TCbvBcJPWrw2eP
>>>> a=ice-options:google-ice
>>>> a=fingerprint:sha-256 52:8C:0F:27:C6:D6:CF:AE:F4:87:AC:AE:DF:7B:9B:B2:75:90:60:6A:2A:82:09:98:AD:04:0B:35:45:6A:13:A2
>>>> a=setup:actpass
>>>> a=mid:video
>>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>>> a=recvonly
>>>> a=rtcp-mux
>>>> a=rtpmap:100 VP8/90000
>>>> a=rtcp-fb:100 ccm fir
>>>> a=rtcp-fb:100 nack
>>>> a=rtcp-fb:100 nack pli
>>>> a=rtcp-fb:100 goog-remb
>>>> a=rtpmap:116 red/90000
>>>> a=rtpmap:117 ulpfec/90000
>>>> a=rtpmap:96 rtx/90000
>>>> a=fmtp:96 apt=100
>>>> 
>>>> 
>>>> Here is SDP received from FreeSwitch in JAIN SIP via UDP:
>>>> 
>>>> v=0
>>>> o=FreeSWITCH 1425524563 1425524564 IN IP4 192.168.131.253
>>>> s=FreeSWITCH
>>>> c=IN IP4 192.168.131.253
>>>> t=0 0
>>>> m=audio 16390 RTP/AVP 111 0 8 101 13
>>>> a=rtpmap:111 opus/48000/2
>>>> a=fmtp:111 minptime=10
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> m=video 16388 RTP/AVP 100
>>>> a=rtpmap:100 VP8/90000
>>>> 
>>>> 
>>>> I suppose that FreeSwitch wants to see WebRTC connection only on the WebSocket ports and it doesn't know that my UDP client is actually WebRTC client. 
>>>> 
>>>> So I'm wondering if it possible to connect SIP client to the WebSocket port via TCP using standard SIP client and never upgrade connection to WebSocket?
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/38b166e2/attachment.html 

From mike at jerris.com  Fri Mar 13 17:39:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 13 Mar 2015 10:39:28 -0400
Subject: [Freeswitch-users] Using Freeswitch with TLS
In-Reply-To: <CANT=1X0vNkQsPZQz99iF8Bf+njJZ5JidOdGnqirbBskBWhC4_g@mail.gmail.com>
References: <CANT=1X0vNkQsPZQz99iF8Bf+njJZ5JidOdGnqirbBskBWhC4_g@mail.gmail.com>
Message-ID: <F154C562-7FD1-4F6F-BB2C-1A7D87CCD282@jerris.com>

it depends on what exactly your needs are.  If you need sip, you need mod_sofia, you probably need mod_commands and mod_dptools, maybe mod_console, mod_logfile, maybe a bunch of others depending on what exactly you are trying to do.

> On Mar 13, 2015, at 9:33 AM, Y?cel ALTUNAY <is.yaltunay at gmail.com> wrote:
> 
> Hi,
> I want to use Freeswitch with TLS and i want to learn which modules i need mandatory.
> i want to remove modules the i dont need.
> can you list modules here or write any web page link?
> Thank you.



From bote_radio at botecomm.com  Fri Mar 13 17:47:02 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 13 Mar 2015 10:47:02 -0400
Subject: [Freeswitch-users] bridge calls in event socket app
In-Reply-To: <CAPTXd0qdxeBjLLv-NfSK6GsjU=7dMbpEz-kS8dUk-OmJjo0upQ@mail.gmail.com>
References: <CAAwzf1=5dVjMKouOoL_ijh3Gk7L0EEL=m1trnWnbje-W0rwoCA@mail.gmail.com>
	<CAPTXd0qdxeBjLLv-NfSK6GsjU=7dMbpEz-kS8dUk-OmJjo0upQ@mail.gmail.com>
Message-ID: <08ff01d05d9c$916d38c0$b447aa40$@botecomm.com>

There are some wise words about creating call legs inside a script from a core developer (anthm) on

https://freeswitch.org/confluence/display/FREESWITCH/Client+and+Developer+Interfaces

 

Bote

 

 

From: Godson Gera
Sent: Friday, 13 March, 2015 10:22
Subject: Re: [Freeswitch-users] bridge calls in event socket app

 

Could you be more specific on what you are trying to do ? If leg A is executing bridge command either in evensocket or in dialplan then they will be automatically be bridged by FS unless you have set ignore_early_media channel variable. 

 

On Wed, Mar 11, 2015 at 6:52 AM, sukitha jayasinghe <sukithaj at gmail.com> wrote:

I want bridge two unanswered calls (in eventsocke) to listen to early media. What is the mechanism for that?

 

Best Regards,

Sukitha





 

-- 

Thanks & Regards,
Godson Gera

VoIP Consultant <http://godson.in> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/61d4ebd3/attachment-0001.html 

From jdickson at evolvetsi.com  Fri Mar 13 17:48:06 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Fri, 13 Mar 2015 10:48:06 -0400
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
Message-ID: <CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>

Yes, I can definitely see where master-master mode in MySQL can put
you in a pickle consistency wise..  I just wish Postgres was easier
with regard to fail-back.. failover isn't a problem, but getting
things to fail back requires rsync of data files, etc..

Before I set that up, I'd like to see sofia recover work in my
environment..  Currently it appears that the correct information is
being retrieved from the database, but that FS is hanging up the A leg
with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
behind that hangup.

Is there better logging that I can grab that might be more helpful?

Thanks!

On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
> I has used MySQL in master-master cluster and find it is unstable
> configuration. Many troubles with cluster consistence.
> I recommend use configuration master-slave with tools DB synchronization.
>
> Also migration to PostgreSQL cluster will be smart decision.
>


From is.yaltunay at gmail.com  Fri Mar 13 18:22:16 2015
From: is.yaltunay at gmail.com (=?UTF-8?Q?Y=C3=BCcel_ALTUNAY?=)
Date: Fri, 13 Mar 2015 17:22:16 +0200
Subject: [Freeswitch-users] Using Freeswitch with TLS
In-Reply-To: <F154C562-7FD1-4F6F-BB2C-1A7D87CCD282@jerris.com>
References: <CANT=1X0vNkQsPZQz99iF8Bf+njJZ5JidOdGnqirbBskBWhC4_g@mail.gmail.com>
	<F154C562-7FD1-4F6F-BB2C-1A7D87CCD282@jerris.com>
Message-ID: <CANT=1X3TgUYe_UKuNT0i5D+SY8a6dGwsrb97Ww1PE69uegetxw@mail.gmail.com>

Thank you

2015-03-13 16:39 GMT+02:00 Michael Jerris <mike at jerris.com>:

> it depends on what exactly your needs are.  If you need sip, you need
> mod_sofia, you probably need mod_commands and mod_dptools, maybe
> mod_console, mod_logfile, maybe a bunch of others depending on what exactly
> you are trying to do.
>
> > On Mar 13, 2015, at 9:33 AM, Y?cel ALTUNAY <is.yaltunay at gmail.com>
> wrote:
> >
> > Hi,
> > I want to use Freeswitch with TLS and i want to learn which modules i
> need mandatory.
> > i want to remove modules the i dont need.
> > can you list modules here or write any web page link?
> > Thank you.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/36ddece0/attachment.html 

From areski at gmail.com  Fri Mar 13 20:14:36 2015
From: areski at gmail.com (Areski)
Date: Fri, 13 Mar 2015 18:14:36 +0100
Subject: [Freeswitch-users] Doc-Sprint Friday 13 March 2015
Message-ID: <CAMf7LBhg1_cMiZDwgbtczz1zW6ueAfVqb9+wAybxvAPyrpmsEw@mail.gmail.com>

Hi Everyone,

We are *now* running a small Sprint to work on FreeSWITCH Documentation.
It will be 2-3 hours long.

The Doc-sprint will focus on migrating the remaining pages from old FS Wiki
(https://wiki.freeswitch.org) to Confluence (
https://freeswitch.org/confluence).

We will use this IRC channel during the sprint: #freeswitch-docs
and we will be tracking our work on a spreadsheet:
https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333
to avoid working on the same content.

So during the sprint, please change the page "Status" you are working on to
"Editing" with your name next to it.

Some extra information:
- https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
-
https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation

Let us know if you have any question.

Please join in :)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/3bd30f48/attachment.html 

From brian at freeswitch.org  Fri Mar 13 20:17:51 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 13 Mar 2015 12:17:51 -0500
Subject: [Freeswitch-users] AU Ringback
Message-ID: <CAEJMVkApkLfrT7DYskRqvZ+w6LGwhKJ3ktVUW4znvhBYYjGFpA@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-7368

Anyone have the AU ringback spec handy?  I noticed in searching for it
people have asked for, but nobody has provided it.  I'll go dig it up when
I have time, but if someone has it please post it on the JIRA.

Thanks,

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/983251a0/attachment.html 

From tfred31 at yahoo.com  Sat Mar 14 00:27:29 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Fri, 13 Mar 2015 14:27:29 -0700
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAP6oyxGU7gRKG6H7UhxHiq=h6rmBd_RzEyYpYdMHRxm8FnrBYA@mail.gmail.com>
Message-ID: <1426282049.77213.YahooMailBasic@web160206.mail.bf1.yahoo.com>

I am still real new to using FreeSWITCH, but I thought that the variable   transfer_fallback_extension   was what was needed to do what you describe.

I am searching for the answer to this same question.
With that in mind, if that is the correct variable to use, then how/where do you use it to get back to the original extension which initiated the transfer (not back to the 'operator')?

I want the transfer back to execute INSTEAD of defaulting to going into Record message mode.

Thanks



From govoiper at gmail.com  Sat Mar 14 04:06:27 2015
From: govoiper at gmail.com (SamyGo)
Date: Fri, 13 Mar 2015 21:06:27 -0400
Subject: [Freeswitch-users] Perl ESL not responding !
Message-ID: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>

Hi all,

I'm feeling stupid for asking this question related to ESL. It has worked
for me always but I've a new installation where I've compiled perl ESL
library and when I try to make test scripts run they don't do anything at
all.

The perl code accepts the ESL library, executes the ESLConnection()
function but afterwards any command I send is not seen on the ESL socket !

Here is one of the test scripts from freeswitch source:

#!/usr/bin/perl
use lib '/usr/src/freeswitch/libs/esl/perl';
require ESL;

my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
if (! $con) { die "Unable to establish connection to $host:$port\n"; }
$con->events("plain","all");
while ( $con->connected() ) {
    my $e = $con->recvEventTimed(0);
    print $e->serialize;
}


It just doesn't go in the while() loop. No $con established but no error as
well !

I tried "telnet" to the 8021 port and only then I see packets on port 8021.

I know that there is something fishy with the perl ESL compilation but that
didn't gave any error either and created the required ESL.so w/o
complaining !

Can anyone help me on figuring out what I've missed ?

Best Regards,
Sammy
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/58c9d215/attachment.html 

From brian at freeswitch.org  Sat Mar 14 04:35:48 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 13 Mar 2015 20:35:48 -0500
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
Message-ID: <CAEJMVkB4XRxX1vtCU_9xPkkprutHUs16ywFYhREWX_bHOnTmxQ@mail.gmail.com>

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon") || die $@;

What distro are you using?

On Fri, Mar 13, 2015 at 8:06 PM, SamyGo <govoiper at gmail.com> wrote:

> Hi all,
>
> I'm feeling stupid for asking this question related to ESL. It has worked
> for me always but I've a new installation where I've compiled perl ESL
> library and when I try to make test scripts run they don't do anything at
> all.
>
> The perl code accepts the ESL library, executes the ESLConnection()
> function but afterwards any command I send is not seen on the ESL socket !
>
> Here is one of the test scripts from freeswitch source:
>
> #!/usr/bin/perl
> use lib '/usr/src/freeswitch/libs/esl/perl';
> require ESL;
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
> $con->events("plain","all");
> while ( $con->connected() ) {
>     my $e = $con->recvEventTimed(0);
>     print $e->serialize;
> }
>
>
> It just doesn't go in the while() loop. No $con established but no error
> as well !
>
> I tried "telnet" to the 8021 port and only then I see packets on port 8021.
>
> I know that there is something fishy with the perl ESL compilation but
> that didn't gave any error either and created the required ESL.so w/o
> complaining !
>
> Can anyone help me on figuring out what I've missed ?
>
> Best Regards,
> Sammy
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/8aa46d4c/attachment-0001.html 

From ssinyagin at gmail.com  Sat Mar 14 05:22:37 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 14 Mar 2015 03:22:37 +0100
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
Message-ID: <CACOYK=h47VpNiW=XEwnvWw4rSM9OLTTW3PAoxGwsVDyYfJA4sw@mail.gmail.com>

Just in case, here's the installation sequence for ESL Perl library on
Debian
https://github.com/voxserv/freeswitch-perf-dialer/blob/master/README.md
On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:

> Hi all,
>
> I'm feeling stupid for asking this question related to ESL. It has worked
> for me always but I've a new installation where I've compiled perl ESL
> library and when I try to make test scripts run they don't do anything at
> all.
>
> The perl code accepts the ESL library, executes the ESLConnection()
> function but afterwards any command I send is not seen on the ESL socket !
>
> Here is one of the test scripts from freeswitch source:
>
> #!/usr/bin/perl
> use lib '/usr/src/freeswitch/libs/esl/perl';
> require ESL;
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
> $con->events("plain","all");
> while ( $con->connected() ) {
>     my $e = $con->recvEventTimed(0);
>     print $e->serialize;
> }
>
>
> It just doesn't go in the while() loop. No $con established but no error
> as well !
>
> I tried "telnet" to the 8021 port and only then I see packets on port 8021.
>
> I know that there is something fishy with the perl ESL compilation but
> that didn't gave any error either and created the required ESL.so w/o
> complaining !
>
> Can anyone help me on figuring out what I've missed ?
>
> Best Regards,
> Sammy
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/dc129e4c/attachment.html 

From ssinyagin at gmail.com  Sat Mar 14 05:38:35 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 14 Mar 2015 03:38:35 +0100
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
Message-ID: <CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>

$esl->connected() or die("Cannot connect to FreeSWITCH");

That's a more correct way to do that :)
On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:

> Hi all,
>
> I'm feeling stupid for asking this question related to ESL. It has worked
> for me always but I've a new installation where I've compiled perl ESL
> library and when I try to make test scripts run they don't do anything at
> all.
>
> The perl code accepts the ESL library, executes the ESLConnection()
> function but afterwards any command I send is not seen on the ESL socket !
>
> Here is one of the test scripts from freeswitch source:
>
> #!/usr/bin/perl
> use lib '/usr/src/freeswitch/libs/esl/perl';
> require ESL;
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
> $con->events("plain","all");
> while ( $con->connected() ) {
>     my $e = $con->recvEventTimed(0);
>     print $e->serialize;
> }
>
>
> It just doesn't go in the while() loop. No $con established but no error
> as well !
>
> I tried "telnet" to the 8021 port and only then I see packets on port 8021.
>
> I know that there is something fishy with the perl ESL compilation but
> that didn't gave any error either and created the required ESL.so w/o
> complaining !
>
> Can anyone help me on figuring out what I've missed ?
>
> Best Regards,
> Sammy
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/abbc256b/attachment.html 

From govoiper at gmail.com  Sat Mar 14 05:51:41 2015
From: govoiper at gmail.com (SamyGo)
Date: Fri, 13 Mar 2015 22:51:41 -0400
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
	<CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>
Message-ID: <CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>

Brian, Im using Ubuntu 12.04 Precise Server. Should I change this ?

Thanks Stanislav I'll try that dialler code as well..but yeah did the same
Perl ESL installation. I did install the perl development libraries which
are required for the esl perlmod.

So, again I compiled as per the mentioned github link and tried running
everything over again , :( sadly no joy.



On Fri, Mar 13, 2015 at 10:38 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> $esl->connected() or die("Cannot connect to FreeSWITCH");
>
> That's a more correct way to do that :)
> On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:
>
>> Hi all,
>>
>> I'm feeling stupid for asking this question related to ESL. It has worked
>> for me always but I've a new installation where I've compiled perl ESL
>> library and when I try to make test scripts run they don't do anything at
>> all.
>>
>> The perl code accepts the ESL library, executes the ESLConnection()
>> function but afterwards any command I send is not seen on the ESL socket !
>>
>> Here is one of the test scripts from freeswitch source:
>>
>> #!/usr/bin/perl
>> use lib '/usr/src/freeswitch/libs/esl/perl';
>> require ESL;
>>
>> my $command = shift;
>> my $args = join(" ", @ARGV);
>>
>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
>> $con->events("plain","all");
>> while ( $con->connected() ) {
>>     my $e = $con->recvEventTimed(0);
>>     print $e->serialize;
>> }
>>
>>
>> It just doesn't go in the while() loop. No $con established but no error
>> as well !
>>
>> I tried "telnet" to the 8021 port and only then I see packets on port
>> 8021.
>>
>> I know that there is something fishy with the perl ESL compilation but
>> that didn't gave any error either and created the required ESL.so w/o
>> complaining !
>>
>> Can anyone help me on figuring out what I've missed ?
>>
>> Best Regards,
>> Sammy
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/09a414b9/attachment.html 

From anthony.minessale at gmail.com  Sat Mar 14 05:56:14 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 13 Mar 2015 21:56:14 -0500
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
	<CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>
	<CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>
Message-ID: <CAKbxfG8FVmNML82BWX92rvgpSUAvJBCdD=theNYVXxR_u_CfxQ@mail.gmail.com>

Ubuntu 12 is the devil..?

On Friday, March 13, 2015, SamyGo <govoiper at gmail.com> wrote:

> Brian, Im using Ubuntu 12.04 Precise Server. Should I change this ?
>
> Thanks Stanislav I'll try that dialler code as well..but yeah did the same
> Perl ESL installation. I did install the perl development libraries which
> are required for the esl perlmod.
>
> So, again I compiled as per the mentioned github link and tried running
> everything over again , :( sadly no joy.
>
>
>
> On Fri, Mar 13, 2015 at 10:38 PM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>
>> $esl->connected() or die("Cannot connect to FreeSWITCH");
>>
>> That's a more correct way to do that :)
>> On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com
>> <javascript:_e(%7B%7D,'cvml','govoiper at gmail.com');>> wrote:
>>
>>> Hi all,
>>>
>>> I'm feeling stupid for asking this question related to ESL. It has
>>> worked for me always but I've a new installation where I've compiled perl
>>> ESL library and when I try to make test scripts run they don't do anything
>>> at all.
>>>
>>> The perl code accepts the ESL library, executes the ESLConnection()
>>> function but afterwards any command I send is not seen on the ESL socket !
>>>
>>> Here is one of the test scripts from freeswitch source:
>>>
>>> #!/usr/bin/perl
>>> use lib '/usr/src/freeswitch/libs/esl/perl';
>>> require ESL;
>>>
>>> my $command = shift;
>>> my $args = join(" ", @ARGV);
>>>
>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
>>> $con->events("plain","all");
>>> while ( $con->connected() ) {
>>>     my $e = $con->recvEventTimed(0);
>>>     print $e->serialize;
>>> }
>>>
>>>
>>> It just doesn't go in the while() loop. No $con established but no error
>>> as well !
>>>
>>> I tried "telnet" to the 8021 port and only then I see packets on port
>>> 8021.
>>>
>>> I know that there is something fishy with the perl ESL compilation but
>>> that didn't gave any error either and created the required ESL.so w/o
>>> complaining !
>>>
>>> Can anyone help me on figuring out what I've missed ?
>>>
>>> Best Regards,
>>> Sammy
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>

-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/7b417af1/attachment-0001.html 

From govoiper at gmail.com  Sat Mar 14 06:00:29 2015
From: govoiper at gmail.com (SamyGo)
Date: Fri, 13 Mar 2015 23:00:29 -0400
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAKbxfG8FVmNML82BWX92rvgpSUAvJBCdD=theNYVXxR_u_CfxQ@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
	<CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>
	<CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>
	<CAKbxfG8FVmNML82BWX92rvgpSUAvJBCdD=theNYVXxR_u_CfxQ@mail.gmail.com>
Message-ID: <CAJUJwth1F3v6V1EO0M7T9GpbN_N89ag_HdMDU9wohRRtk5NUWw@mail.gmail.com>

Yup, couple years back it was the game, Im just doing the dist-upgrade to
see what happens. So, thinking that Im using old version I also tried the
v1.2.stable branch and same results.




On Fri, Mar 13, 2015 at 10:56 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Ubuntu 12 is the devil..
> [image: ?]
>
> On Friday, March 13, 2015, SamyGo <govoiper at gmail.com> wrote:
>
>> Brian, Im using Ubuntu 12.04 Precise Server. Should I change this ?
>>
>> Thanks Stanislav I'll try that dialler code as well..but yeah did the
>> same Perl ESL installation. I did install the perl development libraries
>> which are required for the esl perlmod.
>>
>> So, again I compiled as per the mentioned github link and tried running
>> everything over again , :( sadly no joy.
>>
>>
>>
>> On Fri, Mar 13, 2015 at 10:38 PM, Stanislav Sinyagin <ssinyagin at gmail.com
>> > wrote:
>>
>>> $esl->connected() or die("Cannot connect to FreeSWITCH");
>>>
>>> That's a more correct way to do that :)
>>> On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:
>>>
>>>> Hi all,
>>>>
>>>> I'm feeling stupid for asking this question related to ESL. It has
>>>> worked for me always but I've a new installation where I've compiled perl
>>>> ESL library and when I try to make test scripts run they don't do anything
>>>> at all.
>>>>
>>>> The perl code accepts the ESL library, executes the ESLConnection()
>>>> function but afterwards any command I send is not seen on the ESL socket !
>>>>
>>>> Here is one of the test scripts from freeswitch source:
>>>>
>>>> #!/usr/bin/perl
>>>> use lib '/usr/src/freeswitch/libs/esl/perl';
>>>> require ESL;
>>>>
>>>> my $command = shift;
>>>> my $args = join(" ", @ARGV);
>>>>
>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
>>>> $con->events("plain","all");
>>>> while ( $con->connected() ) {
>>>>     my $e = $con->recvEventTimed(0);
>>>>     print $e->serialize;
>>>> }
>>>>
>>>>
>>>> It just doesn't go in the while() loop. No $con established but no
>>>> error as well !
>>>>
>>>> I tried "telnet" to the 8021 port and only then I see packets on port
>>>> 8021.
>>>>
>>>> I know that there is something fishy with the perl ESL compilation but
>>>> that didn't gave any error either and created the required ESL.so w/o
>>>> complaining !
>>>>
>>>> Can anyone help me on figuring out what I've missed ?
>>>>
>>>> Best Regards,
>>>> Sammy
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/4274b83f/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 1616 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/4274b83f/attachment.png 

From bote_radio at botecomm.com  Sat Mar 14 06:50:56 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 13 Mar 2015 23:50:56 -0400
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <CAJUJwth1F3v6V1EO0M7T9GpbN_N89ag_HdMDU9wohRRtk5NUWw@mail.gmail.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>	<CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>	<CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>	<CAKbxfG8FVmNML82BWX92rvgpSUAvJBCdD=theNYVXxR_u_CfxQ@mail.gmail.com>
	<CAJUJwth1F3v6V1EO0M7T9GpbN_N89ag_HdMDU9wohRRtk5NUWw@mail.gmail.com>
Message-ID: <09ba01d05e0a$14370b60$3ca52220$@botecomm.com>

I?m surprised that FS 1.2 would even compile with all those cobwebs J

 

Bote

 

 

From: SamyGo
Sent: Friday, 13 March, 2015 23:00
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Perl ESL not responding !

 

Yup, couple years back it was the game, Im just doing the dist-upgrade to see what happens. So, thinking that Im using old version I also tried the v1.2.stable branch and same results.

 

 

 

 

On Fri, Mar 13, 2015 at 10:56 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

Ubuntu 12 is the devil..

??

On Friday, March 13, 2015, SamyGo <govoiper at gmail.com> wrote:

Brian, Im using Ubuntu 12.04 Precise Server. Should I change this ?

 

Thanks Stanislav I'll try that dialler code as well..but yeah did the same Perl ESL installation. I did install the perl development libraries which are required for the esl perlmod.

 

So, again I compiled as per the mentioned github link and tried running everything over again , :( sadly no joy.

 

 

 

On Fri, Mar 13, 2015 at 10:38 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

$esl->connected() or die("Cannot connect to FreeSWITCH");

That's a more correct way to do that :)

On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:

Hi all,

 

I'm feeling stupid for asking this question related to ESL. It has worked for me always but I've a new installation where I've compiled perl ESL library and when I try to make test scripts run they don't do anything at all.

 

The perl code accepts the ESL library, executes the ESLConnection() function but afterwards any command I send is not seen on the ESL socket !

 

Here is one of the test scripts from freeswitch source:

 

#!/usr/bin/perl

use lib '/usr/src/freeswitch/libs/esl/perl';

require ESL;

 

my $command = shift;

my $args = join(" ", @ARGV);

 

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");

if (! $con) { die "Unable to establish connection to $host:$port\n"; }

$con->events("plain","all");

while ( $con->connected() ) {

    my $e = $con->recvEventTimed(0);

    print $e->serialize;

}

 

 

It just doesn't go in the while() loop. No $con established but no error as well !

I tried "telnet" to the 8021 port and only then I see packets on port 8021.

I know that there is something fishy with the perl ESL compilation but that didn't gave any error either and created the required ESL.so w/o complaining !

Can anyone help me on figuring out what I've missed ?

 

Best Regards,

Sammy

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/5167eb5a/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/png
Size: 1616 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150313/5167eb5a/attachment-0001.png 

From kamil.nigmatullin at gmail.com  Sat Mar 14 17:19:49 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Sat, 14 Mar 2015 20:19:49 +0600
Subject: [Freeswitch-users] Limit application usage in Lua
In-Reply-To: <CAABKsuQ3d8RALJDs_BqVpLsp_2ekFCFExQT5ynC7t8vyo3BuMg@mail.gmail.com>
References: <CAABKsuTOGeB=UfovXeZvwLyjprFC97QdyJ4Wdf72ZLmsQd0YqQ@mail.gmail.com>
	<082401d05d03$14556450$3d002cf0$@freeswitch.org>
	<CAABKsuQ3d8RALJDs_BqVpLsp_2ekFCFExQT5ynC7t8vyo3BuMg@mail.gmail.com>
Message-ID: <CA+XtieeF-rQoTkJRvBCXtxVw7ShQAF2E1Cg-uqPNcH78qRQ-ZQ@mail.gmail.com>

I think what you did is not a dirty hack but normal scripting. You cannot
exit or stop in the middle of lua script. Just let it finish its job
13 ????? 2015 ?. 17:32 ???????????? "Florent Krieg" <flokrrr at gmail.com>
???????:

> Hi Michael,
>
> Actually, here is what happens:
>
> Context: XML dialplan with the following extensions:
>         <extension name="limit_exceeded">
>             <condition field="destination_number"
> expression="^limit_exceeded$">
>                 <action application="respond" data="503"/>
>             </condition>
>         </extension>
>         <extension name="outgoing_call">
>             <condition>
>                 <action application="lua"
> data="dialplans/caller_script.lua"/>
>             </condition>
>         </extension>
>
> 1/ a call is started, the second extension entry in the XML dialplan
> matches and the action starts the lua script
> 2/ after having set some (a lot actually) variables, here is the code part
> for checking call-rate limits:
> -- Security (to avoid outage): limit CAPS
> --   hash backend is needed when using interval
> --   2 call attempts per 10 seconds here
> logz("[Calls limit] Checking caps for " .. billedcaller)
> session:execute("limit", "hash " .. billedcaller .. " caps 2/10
> !REQUESTED_CHAN_UNAVAIL")
> rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
> if not rate_var then
>     logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
> aborting the dialplan.")
>     return
> end
> 3/ after this code part, a lot of processing is done (for instance
> checking call restrictions calling external web services and so on), it's
> like 600 lines of lua there
> 4/ in the end, the session:execute bridge app is run
>
> At 2/, when limit is exceeded, I see in the log:
> 2015-03-13 12:22:40.454660 [INFO] switch_cpp.cpp:1328 * (caller/0) #
> 7a16c525263a8bb740517f5a28552016 at trunkinge.voip * [Calls limit] Checking
> caps for asteriskrd at trunkinge.voip
> EXECUTE sofia/internal_auth/asteriskrd at trunkinge.voip limit(hash
> asteriskrd at trunkinge.voip caps 2/10 !REQUESTED_CHAN_UNAVAIL)
> 2015-03-13 12:22:40.454660 [DEBUG] switch_limit.c:126 incr called:
> asteriskrd at trunkinge.voip_caps max:2, interval:10
> 2015-03-13 12:22:40.454660 [INFO] mod_hash.c:176 Usage for
> asteriskrd at trunkinge.voip_caps exceeds maximum rate of 2/10s, now at 3
> 2015-03-13 12:22:40.454660 [INFO] switch_cpp.cpp:1328 * (caller/0) #
> 7a16c525263a8bb740517f5a28552016 at trunkinge.voip * [Calls limit] CAPS
> limit reached for asteriskrd at trunkinge.voip, aborting the dialplan.
>
> and the a-leg is successfully hangup.
>
> But here, 3/ and 4/ are aborting only because of that code part:
> rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
> if not rate_var then
>     logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
> aborting the dialplan.")
>     return
> end
>
> It works, but I'm pretty sure that it is not the proper way to do it.
>
> If I remove that code part, even though the a-leg is hangup because of the
> leg being transferred to 'limit_exceeded', the rest of the dialplan is
> still processed, even though the call is aborted.
>
> I'd like to be able to catch right after doing session:execute limit app,
> the return of the command.
> If the call is transferred to 'limit_exceeded', then I'd like to abort the
> rest of the dialplan execution.
>
> I have to admit I might be missing something here therefore if you can
> help me, that'd be amazing.
> Thanks in advance
> Florent
>
>
> 2015-03-12 21:28 GMT+01:00 Michael S Collins <msc at freeswitch.org>:
>
>> Some context here would be helpful. Can you pastebin your Lua script, or
>> at least the relevant lines that demonstrate what is happening? Also, when
>> you say that the rest of the Lua dialplan is still processed, what does
>> that mean?
>>
>>
>>
>> -MC
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Florent
>> Krieg
>> *Sent:* Thursday, March 12, 2015 11:43 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Limit application usage in Lua
>>
>>
>>
>> Hi all,
>>
>> I wanted to implement call rate limits using this Lua instruction:
>> session:execute("limit", "hash " .. billedcaller .. " caps 10/1
>> !REQUESTED_CHAN_UNAVAIL")
>>
>> It is actually working, but the rest of the Lua dialplan is still
>> processed, which is a problem in my case.
>>
>> I made a dirty but quick workaround, looking like this:
>> rate_var = session:getVariable("limit_rate_" .. billedcaller .. "_caps")
>> if not rate_var then
>>     logz("[Calls limit] CAPS limit reached for " .. billedcaller .. ",
>> aborting the dialplan.")
>>     return
>> end
>>
>> How can it be done properly?
>>
>> Is there a way to be able to get the result of the 'limit' app call?
>>
>> Or shall I check the status of the a-leg just after to decide to process
>> the rest of the dialplan or not?
>>
>> I know that this issue doesn't occur in a 'pure' XML dialplan, but I'm
>> trying to find a solution for my Lua-only dialplan.
>>
>> Thanks in advance if you have any idea.
>>
>> I'm willing to try any possible solution you would think about!
>>
>> Florent
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/98d03932/attachment.html 

From kamil.nigmatullin at gmail.com  Sat Mar 14 17:24:18 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Sat, 14 Mar 2015 20:24:18 +0600
Subject: [Freeswitch-users] Perl ESL not responding !
In-Reply-To: <09ba01d05e0a$14370b60$3ca52220$@botecomm.com>
References: <CAJUJwtjaHLgzW9vjXBkkRaeh_NWdB=iUNBuq0yzRYEjOCJWccw@mail.gmail.com>
	<CACOYK=hVEdF3xNkpStGu9SWFLNxXCTU6pXGZd=XqP0CY+Dky8Q@mail.gmail.com>
	<CAJUJwth323ZoV4RSJnJA+TtpLNZWc2grrd9U-Wtahh0SQF8ePA@mail.gmail.com>
	<CAKbxfG8FVmNML82BWX92rvgpSUAvJBCdD=theNYVXxR_u_CfxQ@mail.gmail.com>
	<CAJUJwth1F3v6V1EO0M7T9GpbN_N89ag_HdMDU9wohRRtk5NUWw@mail.gmail.com>
	<09ba01d05e0a$14370b60$3ca52220$@botecomm.com>
Message-ID: <CA+XtiediaVBeAy+P50GFsyPCeNjtHeCHqLozivicXEqyDQfUXw@mail.gmail.com>

To get warnings in perl you have to use warnings library or -w flag
14 ????? 2015 ?. 9:51 ???????????? "Bote Man" <bote_radio at botecomm.com>
???????:

> I?m surprised that FS 1.2 would even compile with all those cobwebs J
>
>
>
> Bote
>
>
>
>
>
> *From:* SamyGo
> *Sent:* Friday, 13 March, 2015 23:00
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Perl ESL not responding !
>
>
>
> Yup, couple years back it was the game, Im just doing the dist-upgrade to
> see what happens. So, thinking that Im using old version I also tried the
> v1.2.stable branch and same results.
>
>
>
>
>
>
>
>
>
> On Fri, Mar 13, 2015 at 10:56 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> Ubuntu 12 is the devil..
>
> [image: ?]
>
> On Friday, March 13, 2015, SamyGo <govoiper at gmail.com> wrote:
>
> Brian, Im using Ubuntu 12.04 Precise Server. Should I change this ?
>
>
>
> Thanks Stanislav I'll try that dialler code as well..but yeah did the same
> Perl ESL installation. I did install the perl development libraries which
> are required for the esl perlmod.
>
>
>
> So, again I compiled as per the mentioned github link and tried running
> everything over again , :( sadly no joy.
>
>
>
>
>
>
>
> On Fri, Mar 13, 2015 at 10:38 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
> $esl->connected() or die("Cannot connect to FreeSWITCH");
>
> That's a more correct way to do that :)
>
> On Mar 14, 2015 2:07 AM, "SamyGo" <govoiper at gmail.com> wrote:
>
> Hi all,
>
>
>
> I'm feeling stupid for asking this question related to ESL. It has worked
> for me always but I've a new installation where I've compiled perl ESL
> library and when I try to make test scripts run they don't do anything at
> all.
>
>
>
> The perl code accepts the ESL library, executes the ESLConnection()
> function but afterwards any command I send is not seen on the ESL socket !
>
>
>
> Here is one of the test scripts from freeswitch source:
>
>
>
> #!/usr/bin/perl
>
> use lib '/usr/src/freeswitch/libs/esl/perl';
>
> require ESL;
>
>
>
> my $command = shift;
>
> my $args = join(" ", @ARGV);
>
>
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>
> if (! $con) { die "Unable to establish connection to $host:$port\n"; }
>
> $con->events("plain","all");
>
> while ( $con->connected() ) {
>
>     my $e = $con->recvEventTimed(0);
>
>     print $e->serialize;
>
> }
>
>
>
>
>
> It just doesn't go in the while() loop. No $con established but no error
> as well !
>
> I tried "telnet" to the 8021 port and only then I see packets on port 8021.
>
> I know that there is something fishy with the perl ESL compilation but
> that didn't gave any error either and created the required ESL.so w/o
> complaining !
>
> Can anyone help me on figuring out what I've missed ?
>
>
>
> Best Regards,
>
> Sammy
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/086db5ec/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 1616 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/086db5ec/attachment-0001.png 

From m.stgeorges at csl-technologies.com  Sat Mar 14 05:18:41 2015
From: m.stgeorges at csl-technologies.com (Michael St-Georges)
Date: Sat, 14 Mar 2015 02:18:41 +0000 (UTC)
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
	after 200
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
Message-ID: <loom.20150314T031658-719@post.gmane.org>

Steven Ayre <steveayre at ...> writes:

> 
> 
> This sounds like it belongs on Jira so the issue can be tracked.
> 
> On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
> Hello,
> The following is on?FreeSWITCH Version 
1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 
21:57:33Z 64bit).
> 
> The design goal for this configuration is that of a simple transcoding 
SBC.? SIP calls arrive with various supported codecs, SIP calls bridge 
out on PCMU.? No users, no auth, etc.? Overall, it seems to work but 
there is one call flow I'm struggling with.
> 
> The B-leg of calls are bridged to a PSTN gateway.? If the gateway's 
signaling follows 100, 183, 200, all is well.? But if the gateway sends 
multiple 183s with different RTP ports, there is no audio when the call 
goes to 200.? See the following example:
> 
> ?- Gateway signals 183 with SDP indicating audio on port 16384.?- 
Gateway signals 183 with SDP indicating audio on port 16386.?- Gateway 
signals 200 with SDP indicating audio on port 16384 (same as original 
183).
> 
> In the debug I see where it detects the port change from 183 #1 to 183 
#2.? As such, I hear early media from both until the 200 OK.? When the 
call connects, I see no such port change in the debug, and since it's 
still listening on the wrong port (from 183 #2), there is no audio.
> 
> I've seen some older posts where Anthony seemed against even the port 
change from 183 #1 to 183 #2, yet that seems to work okay today.? It 
just doesn't sense the port change from 183 --> 200.? I don't know if 
this is a feature, bug, or misconfigured option.? Thoughts are welcome!
> 
> 
> - Jeff
> 
> 
> 
> 
________________________________________________________________________
_
> Professional FreeSWITCH Consulting Services:consulting-
YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.co
m
> Official FreeSWITCH 
Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
.cluecon.com
> FreeSWITCH-users mailing listFreeSWITCH-users <at> 
lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi
tch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
usershttp://www.freeswitch.org
> 
> 
> 
> 
> 
> 
> 
________________________________________________________________________
_
> Professional FreeSWITCH Consulting Services: 
> consulting at ...
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at ...
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
users
> http://www.freeswitch.org

Was this ever fixed? I have a similar problem with the behavior of a 
Samsung PBX that connects to a FreeSwitch.

From tomp at tomp.uk  Sat Mar 14 15:56:20 2015
From: tomp at tomp.uk (Tom Parrott)
Date: Sat, 14 Mar 2015 12:56:20 +0000
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <55042EFB.8090500@tomp.uk>
References: <55042EFB.8090500@tomp.uk>
Message-ID: <55042FF4.808@tomp.uk>

Hi,

There is a tarball of freeswitch 1.4.17 here:

http://files.freeswitch.org/freeswitch-1.4.17.tar.gz

I cannot seem to find a release notes doc for this version, and weirdly 
it doesn't appear in the git tag list.

That only goes up to 1.4.15.

Jira shows several unreleased versions with no notes from .16 to .18

https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-projects-plugin:summary-panel 


But the presence of the .17 tarballs suggests it has been released, but 
what happened to 1.6?

Really confused right now.

Thanks
Tom


-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 4201 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/f44214e6/attachment.bin 

From krice at freeswitch.org  Sat Mar 14 18:51:29 2015
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 14 Mar 2015 09:51:29 -0600
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <55042FF4.808@tomp.uk>
Message-ID: <D129B521.984BD%krice@freeswitch.org>

This is because we are in the middle of changing around the release process
in prep for the future and will be releasing documentation on the processes
in the near future.

Nothing has happened to 1.6 at this point as its not been released, At this
point there is no time frame for the release of 1.6 other then it will be
released when its ready.



On 3/14/15, 6:56 AM, "Tom Parrott" <tomp at tomp.uk> wrote:

> Hi,
> 
> There is a tarball of freeswitch 1.4.17 here:
> 
> http://files.freeswitch.org/freeswitch-1.4.17.tar.gz
> 
> I cannot seem to find a release notes doc for this version, and weirdly
> it doesn't appear in the git tag list.
> 
> That only goes up to 1.4.15.
> 
> Jira shows several unreleased versions with no notes from .16 to .18
> 
> https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-pro
> jects-plugin:summary-panel
> 
> 
> But the presence of the .17 tarballs suggests it has been released, but
> what happened to 1.6?
> 
> Really confused right now.
> 
> Thanks
> Tom
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH





From tomp at tomp.uk  Sat Mar 14 19:00:59 2015
From: tomp at tomp.uk (Tom Parrott)
Date: Sat, 14 Mar 2015 16:00:59 +0000
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <D129B521.984BD%krice@freeswitch.org>
References: <D129B521.984BD%krice@freeswitch.org>
Message-ID: <8CDD005E-EE7F-4E8D-A168-A27F268483EF@tomp.uk>

Sorry I made a typo, I meant 1.4.16

On 14 March 2015 15:51:29 GMT, Ken Rice <krice at freeswitch.org> wrote:
>This is because we are in the middle of changing around the release
>process
>in prep for the future and will be releasing documentation on the
>processes
>in the near future.
>
>Nothing has happened to 1.6 at this point as its not been released, At
>this
>point there is no time frame for the release of 1.6 other then it will
>be
>released when its ready.
>
>
>
>On 3/14/15, 6:56 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>
>> Hi,
>> 
>> There is a tarball of freeswitch 1.4.17 here:
>> 
>> http://files.freeswitch.org/freeswitch-1.4.17.tar.gz
>> 
>> I cannot seem to find a release notes doc for this version, and
>weirdly
>> it doesn't appear in the git tag list.
>> 
>> That only goes up to 1.4.15.
>> 
>> Jira shows several unreleased versions with no notes from .16 to .18
>> 
>>
>https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-pro
>> jects-plugin:summary-panel
>> 
>> 
>> But the presence of the .17 tarballs suggests it has been released,
>but
>> what happened to 1.6?
>> 
>> Really confused right now.
>> 
>> Thanks
>> Tom
>> 
>> 
>>
>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>Twitter: @FreeSWITCH
>
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services: 
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/4a50f682/attachment.html 

From krice at freeswitch.org  Sat Mar 14 20:07:22 2015
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 14 Mar 2015 11:07:22 -0600
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <8CDD005E-EE7F-4E8D-A168-A27F268483EF@tomp.uk>
Message-ID: <D129C6EA.984C3%krice@freeswitch.org>

We started to release it but we found issues with it, so on to the next
version number...

Keep in mind version numbers are cheap


On 3/14/15, 10:00 AM, "Tom Parrott" <tomp at tomp.uk> wrote:

> Sorry I made a typo, I meant 1.4.16
> 
> On 14 March 2015 15:51:29 GMT, Ken Rice <krice at freeswitch.org> wrote:
>> This is because we are in the middle of changing around the release process
>> in prep for the future and will be releasing documentation on the processes
>> in the near future.
>> 
>> Nothing has happened to 1.6 at this point as its not been released, At this
>> point there is no time frame for the release of 1.6 other then it will be
>> released when its ready.
>> 
>> 
>> 
>> On 3/14/15, 6:56 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>> 
>>>  Hi,
>>>  
>>>  There is a tarball of freeswitch 1.4.17 here:
>>>  
>>>  http://files.freeswitch.org/freeswitch-1.4.17.tar.gz
>>>  
>>>  I cannot seem to find a release notes doc for this version, and weirdly
>>>  it doesn't appear in the git tag list.
>>>  
>>>  That only goes up to 1.4.15.
>>>  
>>> 
>>> Jira shows several unreleased versions with no notes from .16 to .18
>>>  
>>>  
>>> https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-p
>>> ro
>>>  jects-plugin:summary-panel
>>>  
>>>  
>>>  But the presence of the .17 tarballs suggests it has been released, but
>>>  what happened to 1.6?
>>>  
>>>  Really confused right now.
>>>  
>>>  Thanks
>>>  Tom
>>>  
>>>  
>>> 
>>> 
>>>  Professional FreeSWITCH Consulting Services:
>>>  consulting at freeswitch.org
>>>  http://www.freeswitchsolutions.com
>>>  
>>>  Official FreeSWITCH Sites
>>>  http://www.freeswitch.org
>>>  http://confluence.freeswitch.org
>>>  http://www.cluecon.com
>>>  
>>>  FreeSWITCH-users mailing list
>>> 
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/0d9f9023/attachment-0001.html 

From tomp at tomp.uk  Sat Mar 14 19:15:00 2015
From: tomp at tomp.uk (Tom Parrott)
Date: Sat, 14 Mar 2015 16:15:00 +0000
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <D129C6EA.984C3%krice@freeswitch.org>
References: <D129C6EA.984C3%krice@freeswitch.org>
Message-ID: <3C268046-A2ED-4B1A-B86C-3F348E0E052F@tomp.uk>

Cool no problem, just wanted to figure out which version is latest to build 

On 14 March 2015 17:07:22 GMT, Ken Rice <krice at freeswitch.org> wrote:
>We started to release it but we found issues with it, so on to the next
>version number...
>
>Keep in mind version numbers are cheap
>
>
>On 3/14/15, 10:00 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>
>> Sorry I made a typo, I meant 1.4.16
>> 
>> On 14 March 2015 15:51:29 GMT, Ken Rice <krice at freeswitch.org> wrote:
>>> This is because we are in the middle of changing around the release
>process
>>> in prep for the future and will be releasing documentation on the
>processes
>>> in the near future.
>>> 
>>> Nothing has happened to 1.6 at this point as its not been released,
>At this
>>> point there is no time frame for the release of 1.6 other then it
>will be
>>> released when its ready.
>>> 
>>> 
>>> 
>>> On 3/14/15, 6:56 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>>> 
>>>>  Hi,
>>>>  
>>>>  There is a tarball of freeswitch 1.4.17 here:
>>>>  
>>>>  http://files.freeswitch.org/freeswitch-1.4.17.tar.gz
>>>>  
>>>>  I cannot seem to find a release notes doc for this version, and
>weirdly
>>>>  it doesn't appear in the git tag list.
>>>>  
>>>>  That only goes up to 1.4.15.
>>>>  
>>>> 
>>>> Jira shows several unreleased versions with no notes from .16 to
>.18
>>>>  
>>>>  
>>>>
>https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-p
>>>> ro
>>>>  jects-plugin:summary-panel
>>>>  
>>>>  
>>>>  But the presence of the .17 tarballs suggests it has been
>released, but
>>>>  what happened to 1.6?
>>>>  
>>>>  Really confused right now.
>>>>  
>>>>  Thanks
>>>>  Tom
>>>>  
>>>>  
>>>> 
>>>> 
>>>>  Professional FreeSWITCH Consulting Services:
>>>>  consulting at freeswitch.org
>>>>  http://www.freeswitchsolutions.com
>>>>  
>>>>  Official FreeSWITCH Sites
>>>>  http://www.freeswitch.org
>>>>  http://confluence.freeswitch.org
>>>>  http://www.cluecon.com
>>>>  
>>>>  FreeSWITCH-users mailing list
>>>> 
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  http://www.freeswitch.org
>> 
>> 
>>
>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>Twitter: @FreeSWITCH
>
>
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services: 
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150314/6dc8bd63/attachment.html 

From paul.atreides83 at googlemail.com  Sun Mar 15 18:43:27 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Sun, 15 Mar 2015 16:43:27 +0100
Subject: [Freeswitch-users] Channel Variable:
	variable_pre_transfer_caller_id_number
Message-ID: <CAP6oyxHHt9Dn3jM=5j0hqQ-mG9fBbrnG3Q14qEsWBFzvdcu4wg@mail.gmail.com>

Hi,

<action application="info"/>
when I call the info app in my dial plan then I can see the following vars
in my log

variable_pre_transfer_caller_id_name: []
variable_pre_transfer_caller_id_number: [18]

But when I try to access one single channel variable its always empty. Why?
${variable_pre_transfer_caller_id_number}

Thank you for your help.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150315/fcd1547f/attachment.html 

From brian at freeswitch.org  Sun Mar 15 23:12:02 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 15 Mar 2015 15:12:02 -0500
Subject: [Freeswitch-users] Freeswitch 1.4.17
In-Reply-To: <3C268046-A2ED-4B1A-B86C-3F348E0E052F@tomp.uk>
References: <D129C6EA.984C3%krice@freeswitch.org>
	<3C268046-A2ED-4B1A-B86C-3F348E0E052F@tomp.uk>
Message-ID: <CAEJMVkA6MDye8NYg2aHp580-6GYAGG2wMMuObJ2yS6jO1AEkFA@mail.gmail.com>

Master! :) LOL

On Saturday, March 14, 2015, Tom Parrott <tomp at tomp.uk> wrote:

> Cool no problem, just wanted to figure out which version is latest to
> build
>
> On 14 March 2015 17:07:22 GMT, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>> wrote:
>>
>> We started to release it but we found issues with it, so on to the next
>> version number...
>>
>> Keep in mind version numbers are cheap
>>
>>
>> On 3/14/15, 10:00 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>>
>> Sorry I made a typo, I meant 1.4.16
>>
>> On 14 March 2015 15:51:29 GMT, Ken Rice <krice at freeswitch.org> wrote:
>>
>> This is because we are in the middle of changing around the release
>> process
>> in prep for the future and will be releasing documentation on the
>> processes
>> in the near future.
>>
>> Nothing has happened to 1.6 at this point as its not been released, At
>> this
>> point there is no time frame for the release of 1.6 other then it will be
>> released when its ready.
>>
>>
>>
>> On 3/14/15, 6:56 AM, "Tom Parrott" <tomp at tomp.uk> wrote:
>>
>>  Hi,
>>
>>  There is a tarball of freeswitch 1.4.17 here:
>>
>>  http://files.freeswitch.org/freeswitch-1.4.17.tar.gz
>>
>>  I cannot seem to find a release notes doc for this version, and weirdly
>>  it doesn't appear in the git tag list.
>>
>>  That only goes up to 1.4.15.
>>
>>
>> Jira shows several unreleased versions with no notes from .16 to .18
>>
>>
>> https://freeswitch.org/jira/browse/FS/?selectedTab=com.atlassian.jira.jira-pro
>>  jects-plugin:summary-panel
>>
>>
>>  But the presence of the .17 tarballs suggests it has been released, but
>>  what happened to 1.6?
>>
>>  Really confused right now.
>>
>>  Thanks
>>  Tom
>>
>>
>> ------------------------------
>>
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>>
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://confluence.freeswitch.org
>>  http://www.cluecon.com
>>
>>  FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>> Twitter: @FreeSWITCH
>>
>>  ------------------------------
>>
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150315/38e49490/attachment-0001.html 

From bote_radio at botecomm.com  Mon Mar 16 00:25:51 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sun, 15 Mar 2015 17:25:51 -0400
Subject: [Freeswitch-users] Channel
	Variable:	variable_pre_transfer_caller_id_number
In-Reply-To: <CAP6oyxHHt9Dn3jM=5j0hqQ-mG9fBbrnG3Q14qEsWBFzvdcu4wg@mail.gmail.com>
References: <CAP6oyxHHt9Dn3jM=5j0hqQ-mG9fBbrnG3Q14qEsWBFzvdcu4wg@mail.gmail.com>
Message-ID: <009e01d05f66$9d4d82f0$d7e888d0$@botecomm.com>

I believe the ?variable_? prefix only applies to CDRs and logs. When processing them in the dialplan omit the ?variable_? prefix and just use

pre_transfer_caller_id_name

 

 

Bote

 

FreeSWITCH Docs Janitor

http://freeswitch.org/confluence

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Atreides
Sent: Sunday, 15 March, 2015 11:43
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Channel Variable: variable_pre_transfer_caller_id_number

 

Hi,

<action application="info"/>
when I call the info app in my dial plan then I can see the following vars in my log

variable_pre_transfer_caller_id_name: []
variable_pre_transfer_caller_id_number: [18]

But when I try to access one single channel variable its always empty. Why?
${variable_pre_transfer_caller_id_number}

Thank you for your help.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150315/577f1172/attachment.html 

From dujinfang at gmail.com  Mon Mar 16 05:26:06 2015
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 16 Mar 2015 10:26:06 +0800
Subject: [Freeswitch-users] failover in inbound event socket mode
In-Reply-To: <CAB4GJpXpJm+sAbjUfWVcGEtGy_zMqQuaOYTA3u+wPbSR46LB6w@mail.gmail.com>
References: <CAB4GJpXpJm+sAbjUfWVcGEtGy_zMqQuaOYTA3u+wPbSR46LB6w@mail.gmail.com>
Message-ID: <CAJs+-5qV4RnL9nUviL-+-OU9vjnFXtVKgeOq-O2zp2MDW0W6hA@mail.gmail.com>

The only reason ORIGINATOR_CANCEL is that FS canceled the call and moving
forward to the next  after the `|' so there's no reason it work as you
thought. So I would completely remove everything including and after `|' if
I really don't want to call over the next gateway.



On Fri, Mar 13, 2015 at 9:53 PM, ?? <ppyy at pubyun.com> wrote:

>
> we would like to fail over in inbound event socket:
>
> dial in -> park
> origin -> park
> uuid_bridge
>
> originate
> {continue_on_fail=^^:NO_USER_RESPONSE:UNALLOCATED_NUMBER:NORMAL_TEMPORARY_F
> AILURE:NO_ROUTE_DESTINATION:CALL_REJECTED:RECOVERY_ON_TIMER_EXPIRE:NETWORK_OUT_OF_ORDER,fail_on_single_reject=^^:USER_BUSY:NO_ANSWER:ORIGINATOR_CANCEL}sofia/gateway/gw01/xxxx|sofia/gateway/gw02/xxxx
> &park()
>
> it will not hangup if the first gateway fails with ORIGINATOR_CANCEL.
>
> how can set continue_on_fail and fail_on_single_reject?
>
> --
> Peng Yong
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/0b887dd1/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar 16 13:05:21 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Mon, 16 Mar 2015 11:05:21 +0100
Subject: [Freeswitch-users] Channel Variable:
	variable_pre_transfer_caller_id_number
In-Reply-To: <009e01d05f66$9d4d82f0$d7e888d0$@botecomm.com>
References: <CAP6oyxHHt9Dn3jM=5j0hqQ-mG9fBbrnG3Q14qEsWBFzvdcu4wg@mail.gmail.com>
	<009e01d05f66$9d4d82f0$d7e888d0$@botecomm.com>
Message-ID: <CAP6oyxFEezejXBEHG+iam7OT7btXj6OKAtiU0rXjT8QoETX7EA@mail.gmail.com>

You are right, thank you

On Sun, Mar 15, 2015 at 10:25 PM, Bote Man <bote_radio at botecomm.com> wrote:

> I believe the ?variable_? prefix only applies to CDRs and logs. When
> processing them in the dialplan omit the ?variable_? prefix and just use
>
> pre_transfer_caller_id_name
>
>
>
>
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Paul
> Atreides
> *Sent:* Sunday, 15 March, 2015 11:43
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Channel Variable:
> variable_pre_transfer_caller_id_number
>
>
>
> Hi,
>
> <action application="info"/>
> when I call the info app in my dial plan then I can see the following vars
> in my log
>
> variable_pre_transfer_caller_id_name: []
> variable_pre_transfer_caller_id_number: [18]
>
> But when I try to access one single channel variable its always empty. Why?
> ${variable_pre_transfer_caller_id_number}
>
> Thank you for your help.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/7be0f9cd/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar 16 13:11:23 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Mon, 16 Mar 2015 11:11:23 +0100
Subject: [Freeswitch-users] Transfer back to origin on failure
In-Reply-To: <CAP6oyxGU7gRKG6H7UhxHiq=h6rmBd_RzEyYpYdMHRxm8FnrBYA@mail.gmail.com>
References: <CAP6oyxGknDiMEjPebxYeoE6OKHdSJ9oz45pjX4KtZjFEyi0ciA@mail.gmail.com>
	<CAKzWOxXAnfh47aLdL0OWik2jYeRH6c7tO5zCSOwq3YZCrcQBDA@mail.gmail.com>
	<CAP6oyxGU7gRKG6H7UhxHiq=h6rmBd_RzEyYpYdMHRxm8FnrBYA@mail.gmail.com>
Message-ID: <CAP6oyxGTSU2KtzwFGFb-mn_6a4wn+N5U5DxhZb6NqtqA5rS02w@mail.gmail.com>

hi,

can someone tell me what is happening here. I figured that freeswitch is
not leaving the extension,
it just rexecutes it. I would like it do another action when there is a
value inside pre_transfer_caller_id_number

[INFO] switch_channel.c:3062 sofia/internal/sip:14 at 10.0.200.14:5060
Flipping CID from "" <18> to "Outbound Call" <14>
[DEBUG] switch_core_state_machine.c:216 (sofia/internal/
sip:14 at 10.0.200.14:5060) State Change CS_ROUTING -> CS_EXECUTE
[DEBUG] switch_core_session.c:1388 Send signal sofia/internal/
sip:14 at 10.0.200.14:5060 [BREAK]
[DEBUG] switch_core_state_machine.c:528 (sofia/internal/
sip:14 at 10.0.200.14:5060) State ROUTING going to sleep
[DEBUG] switch_core_state_machine.c:472 (sofia/internal/
sip:14 at 10.0.200.14:5060) Running State Change CS_EXECUTE
[DEBUG] switch_core_state_machine.c:535 (sofia/internal/
sip:14 at 10.0.200.14:5060) State EXECUTE
[DEBUG] mod_sofia.c:178 sofia/internal/sip:14 at 10.0.200.14:5060 SOFIA EXECUTE
[DEBUG] switch_core_state_machine.c:258 sofia/internal/
sip:14 at 10.0.200.14:5060 Standard EXECUTE


 <extension name="Extension">
   <condition field="destination_number" expression="^([0-9][0-9])$">
       <action application="export" data="dialed_extension=$1"/>
       <action application="set" data="call_timeout=90"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>

       <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
       <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
       <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>

       <action application="bridge"
data="[leg_timeout=20]user/${dialed_extension}"/>

      </condition>
 </extension>

On Tue, Mar 10, 2015 at 4:57 PM, Paul Atreides <
paul.atreides83 at googlemail.com> wrote:

> >In other words, if user A did a blind x-fer of caller C to user B and
> user B doesn't answer (for whatever failure reason) then >caller C would
> start ringing back to user A? Just making sure we understand the scope of
> the feature you're implementing.
>
> Exactly, at the moment the GXP disconnects A and makes a new call to B by
> using the default dial plan. So freeswitch thinks its a normal call and
> precedes as usual. I need a condition where I can check if its a transfer /
> attended call or not
>
> In the log below I found the channel variable
> variable_pre_transfer_caller_id_number and variable_sip_refer_to but these
> are are only set when it is a blind transfer. But for some reason there is
> no match for the condition.
>
> Here is my default dial plan
> http://pastebin.freeswitch.org/23993
>
> Here is the debug from the console when I start the transfer.
> http://pastebin.freeswitch.org/23994
>
>
>
> On Mon, Mar 9, 2015 at 7:18 PM, Michael Collins <msc at freeswitch.org>
> wrote:
>
>>
>>
>> On Sat, Mar 7, 2015 at 9:03 AM, Paul Atreides <
>> paul.atreides83 at googlemail.com> wrote:
>>
>>> When I do a blind transfer then I want freeswitch to call back the
>>> origin who initiated the call.
>>> But I am not able the capture the transfer event?
>>>
>>
>> In other words, if user A did a blind x-fer of caller C to user B and
>> user B doesn't answer (for whatever failure reason) then caller C would
>> start ringing back to user A? Just making sure we understand the scope of
>> the feature you're implementing.
>>
>> How does the GXP do the x-fer? Some kind of hook-flash and DTMF code? Can
>> you pastebin the dialplan that the transferor uses when sending the call?
>>
>> -MC
>>
>>
>>> <condition field="destination_number" expression="^att_xfer$">
>>> <condition field="destination_number" expression="^is_transfer$">
>>>
>>> They seem do be ignored by the dialplan. Is there a list what kind of
>>> values destionation_number can have besides the called numbers?
>>>
>>>
>>> I am doing the transfer with a grandstream gxp2140
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/e7146df9/attachment-0001.html 

From alex.bldck at gmail.com  Mon Mar 16 09:02:32 2015
From: alex.bldck at gmail.com (Alexander Baldeck)
Date: Mon, 16 Mar 2015 14:02:32 +0800
Subject: [Freeswitch-users] Caller ID not logged in CDR properly
Message-ID: <550671F8.7030605@gmail.com>

Hello,

we are using Freeswitch 1.5.12b+git~20140324T234505Z~4dd0a5848f~64bit
interfacing it with Plivo and experience an issue where our call detail
records list the destination_number twice when someone picks up a call
like so:

"Outbound Call","99901637777777","99901637777777","default","2015-02-28
13:15:00","2015-02-28 13:15:08","2015-02-28
13:16:32","92","84","NORMAL_CLEARING","24fdbb20-32a3-11e4-b883-e58f73531fe4","","","PCMA","PCMA"


If the call isn't picked up upon though, everything is alright and
caller_id_number is logged alongside destination_number like so:

"","0123456789","99901637777777","default","2015-02-28
13:11:00","","2015-02-28
13:12:00","60","0","NO_ANSWER","95b050f4-32a2-11e4-b87f-e58f73531fe4","","","",""

The used template in cdr_csv.xml looks like this and I made sure it is
being used:

<template
name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>

I'm at a loss what's going on here, has anyone got insight on where to look?


Best,
Alex


From jpyle at fidelityvoice.com  Mon Mar 16 17:18:34 2015
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Mon, 16 Mar 2015 10:18:34 -0400
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
 after 200
In-Reply-To: <loom.20150314T031658-719@post.gmane.org>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
Message-ID: <CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>

Michael,

I don't believe so.  I haven't seen any activity in the Jira
<https://freeswitch.org/jira/browse/FS-7207>.


- Jeff


On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
m.stgeorges at csl-technologies.com> wrote:

> Steven Ayre <steveayre at ...> writes:
>
> >
> >
> > This sounds like it belongs on Jira so the issue can be tracked.
> >
> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
> > Hello,
> > The following is on FreeSWITCH Version
> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
> 21:57:33Z 64bit).
> >
> > The design goal for this configuration is that of a simple transcoding
> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
> there is one call flow I'm struggling with.
> >
> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
> signaling follows 100, 183, 200, all is well.  But if the gateway sends
> multiple 183s with different RTP ports, there is no audio when the call
> goes to 200.  See the following example:
> >
> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
> signals 200 with SDP indicating audio on port 16384 (same as original
> 183).
> >
> > In the debug I see where it detects the port change from 183 #1 to 183
> #2.  As such, I hear early media from both until the 200 OK.  When the
> call connects, I see no such port change in the debug, and since it's
> still listening on the wrong port (from 183 #2), there is no audio.
> >
> > I've seen some older posts where Anthony seemed against even the port
> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
> just doesn't sense the port change from 183 --> 200.  I don't know if
> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
> >
> >
> > - Jeff
> >
> >
> >
> >
> ________________________________________________________________________
> _
> > Professional FreeSWITCH Consulting Services:consulting-
> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.co
> m
> > Official FreeSWITCH
> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
> .cluecon.com
> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi
> tch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> usershttp://www.freeswitch.org
> >
> >
> >
> >
> >
> >
> >
> ________________________________________________________________________
> _
> > Professional FreeSWITCH Consulting Services:
> > consulting at ...
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at ...
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
>
> Was this ever fixed? I have a similar problem with the behavior of a
> Samsung PBX that connects to a FreeSwitch.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/b1193c44/attachment.html 

From s.safarov at gmail.com  Mon Mar 16 17:26:15 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 16 Mar 2015 14:26:15 +0000
Subject: [Freeswitch-users] Caller ID not logged in CDR properly
References: <550671F8.7030605@gmail.com>
Message-ID: <CAHtxdDd2=AHJGyyuTxK2m0hotW81_qzF6-2DHEBta_pKEN0YmQ@mail.gmail.com>

I believe that if you do the analysis xml_cdr file you can find the cause.
For example, when make attendant transfer then destinational number is
present in b-leg file

On Mon, Mar 16, 2015, 15:28 Alexander Baldeck <alex.bldck at gmail.com> wrote:

> Hello,
>
> we are using Freeswitch 1.5.12b+git~20140324T234505Z~4dd0a5848f~64bit
> interfacing it with Plivo and experience an issue where our call detail
> records list the destination_number twice when someone picks up a call
> like so:
>
> "Outbound Call","99901637777777","99901637777777","default","2015-02-28
> 13:15:00","2015-02-28 13:15:08","2015-02-28
> 13:16:32","92","84","NORMAL_CLEARING","24fdbb20-32a3-11e4-
> b883-e58f73531fe4","","","PCMA","PCMA"
>
>
> If the call isn't picked up upon though, everything is alright and
> caller_id_number is logged alongside destination_number like so:
>
> "","0123456789","99901637777777","default","2015-02-28
> 13:11:00","","2015-02-28
> 13:12:00","60","0","NO_ANSWER","95b050f4-32a2-11e4-b87f-
> e58f73531fe4","","","",""
>
> The used template in cdr_csv.xml looks like this and I made sure it is
> being used:
>
> <template
> name="example">"${caller_id_name}","${caller_id_number}","
> ${destination_number}","${context}","${start_stamp}","${
> answer_stamp}","${end_stamp}","${duration}","${billsec}","${
> hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","
> ${read_codec}","${write_codec}"</template>
>
> I'm at a loss what's going on here, has anyone got insight on where to
> look?
>
>
> Best,
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/2767798c/attachment.html 

From steveayre at gmail.com  Mon Mar 16 18:43:06 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 16 Mar 2015 15:43:06 +0000
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
 after 200
In-Reply-To: <CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
	<CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
Message-ID: <CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>

Just an observation, but what you wrote in the Jira appears to be different
from your original email.

'In the debug I see where it detects the port change from 183 #1 to 183
#2.  As such, I hear early media from both until the 200 OK.  When the call
connects, I see no such port change in the debug, and since it's still
listening on the wrong port (from 183 #2), there is no audio. I've seen
some older posts where Anthony seemed against even the port change from 183
#1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the
port change from 183 --> 200.'

So it sounded like you heard both ringbacks and nothing after 200. In the
Jira you don't talk about the 200 at all, just issues before then. Do you
have audio working on receiving 200 now?



On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> Michael,
>
> I don't believe so.  I haven't seen any activity in the Jira
> <https://freeswitch.org/jira/browse/FS-7207>.
>
>
> - Jeff
>
>
> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
> m.stgeorges at csl-technologies.com> wrote:
>
>> Steven Ayre <steveayre at ...> writes:
>>
>> >
>> >
>> > This sounds like it belongs on Jira so the issue can be tracked.
>> >
>> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
>> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
>> > Hello,
>> > The following is on FreeSWITCH Version
>> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
>> 21:57:33Z 64bit).
>> >
>> > The design goal for this configuration is that of a simple transcoding
>> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
>> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
>> there is one call flow I'm struggling with.
>> >
>> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
>> signaling follows 100, 183, 200, all is well.  But if the gateway sends
>> multiple 183s with different RTP ports, there is no audio when the call
>> goes to 200.  See the following example:
>> >
>> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
>> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
>> signals 200 with SDP indicating audio on port 16384 (same as original
>> 183).
>> >
>> > In the debug I see where it detects the port change from 183 #1 to 183
>> #2.  As such, I hear early media from both until the 200 OK.  When the
>> call connects, I see no such port change in the debug, and since it's
>> still listening on the wrong port (from 183 #2), there is no audio.
>> >
>> > I've seen some older posts where Anthony seemed against even the port
>> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
>> just doesn't sense the port change from 183 --> 200.  I don't know if
>> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
>> >
>> >
>> > - Jeff
>> >
>> >
>> >
>> >
>> ________________________________________________________________________
>> _
>> > Professional FreeSWITCH Consulting Services:consulting-
>> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.co
>> m
>> > Official FreeSWITCH
>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
>> .cluecon.com
>> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
>> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi
>> tch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> usershttp://www.freeswitch.org
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> ________________________________________________________________________
>> _
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at ...
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at ...
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> > http://www.freeswitch.org
>>
>> Was this ever fixed? I have a similar problem with the behavior of a
>> Samsung PBX that connects to a FreeSwitch.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/e10eda28/attachment-0001.html 

From brian at freeswitch.org  Mon Mar 16 18:47:59 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Mar 2015 10:47:59 -0500
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
 after 200
In-Reply-To: <CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
	<CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
	<CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>
Message-ID: <CAEJMVkBj-06nPxCVZkpPv3K-m=H7=9XuaNOWf3jxnWc16gDnOg@mail.gmail.com>

If that is what is going on, then thats a critical detail that wasn't
outlined in the JIRA, maybe we can get a new test on the latest code to
verify?

On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Just an observation, but what you wrote in the Jira appears to be
> different from your original email.
>
> 'In the debug I see where it detects the port change from 183 #1 to 183
> #2.  As such, I hear early media from both until the 200 OK.  When the call
> connects, I see no such port change in the debug, and since it's still
> listening on the wrong port (from 183 #2), there is no audio. I've seen
> some older posts where Anthony seemed against even the port change from 183
> #1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the
> port change from 183 --> 200.'
>
> So it sounded like you heard both ringbacks and nothing after 200. In the
> Jira you don't talk about the 200 at all, just issues before then. Do you
> have audio working on receiving 200 now?
>
>
>
> On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>
>> Michael,
>>
>> I don't believe so.  I haven't seen any activity in the Jira
>> <https://freeswitch.org/jira/browse/FS-7207>.
>>
>>
>> - Jeff
>>
>>
>> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
>> m.stgeorges at csl-technologies.com> wrote:
>>
>>> Steven Ayre <steveayre at ...> writes:
>>>
>>> >
>>> >
>>> > This sounds like it belongs on Jira so the issue can be tracked.
>>> >
>>> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
>>> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
>>> > Hello,
>>> > The following is on FreeSWITCH Version
>>> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
>>> 21:57:33Z 64bit).
>>> >
>>> > The design goal for this configuration is that of a simple transcoding
>>> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
>>> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
>>> there is one call flow I'm struggling with.
>>> >
>>> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
>>> signaling follows 100, 183, 200, all is well.  But if the gateway sends
>>> multiple 183s with different RTP ports, there is no audio when the call
>>> goes to 200.  See the following example:
>>> >
>>> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
>>> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
>>> signals 200 with SDP indicating audio on port 16384 (same as original
>>> 183).
>>> >
>>> > In the debug I see where it detects the port change from 183 #1 to 183
>>> #2.  As such, I hear early media from both until the 200 OK.  When the
>>> call connects, I see no such port change in the debug, and since it's
>>> still listening on the wrong port (from 183 #2), there is no audio.
>>> >
>>> > I've seen some older posts where Anthony seemed against even the port
>>> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
>>> just doesn't sense the port change from 183 --> 200.  I don't know if
>>> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
>>> >
>>> >
>>> > - Jeff
>>> >
>>> >
>>> >
>>> >
>>> ________________________________________________________________________
>>> _
>>> > Professional FreeSWITCH Consulting Services:consulting-
>>> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.co
>>> m
>>> > Official FreeSWITCH
>>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
>>> .cluecon.com
>>> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
>>> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi
>>> tch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> usershttp://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> ________________________________________________________________________
>>> _
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at ...
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at ...
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> > http://www.freeswitch.org
>>>
>>> Was this ever fixed? I have a similar problem with the behavior of a
>>> Samsung PBX that connects to a FreeSwitch.
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/fcacccf5/attachment.html 

From jpyle at fidelityvoice.com  Mon Mar 16 18:58:40 2015
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Mon, 16 Mar 2015 11:58:40 -0400
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
 after 200
In-Reply-To: <CAEJMVkBj-06nPxCVZkpPv3K-m=H7=9XuaNOWf3jxnWc16gDnOg@mail.gmail.com>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
	<CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
	<CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>
	<CAEJMVkBj-06nPxCVZkpPv3K-m=H7=9XuaNOWf3jxnWc16gDnOg@mail.gmail.com>
Message-ID: <CACYJG3J9cauAROt3645JZNXXhMnbs1ewNoZyEYfE96N3uyMWwg@mail.gmail.com>

They're two separate issues.  It looks like I referenced the wrong Jira in
my reply to Michael's comment.  The correct Jira for this issue, involving
the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>.
Oops.  I'll retest shortly.

Agreed, the 180-after-183 no audio issue described in FS-7207
<https://freeswitch.org/jira/browse/FS-7207> is clearly different.  Sorry
for the confusion.



- Jeff

On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian at freeswitch.org> wrote:

> If that is what is going on, then thats a critical detail that wasn't
> outlined in the JIRA, maybe we can get a new test on the latest code to
> verify?
>
> On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Just an observation, but what you wrote in the Jira appears to be
>> different from your original email.
>>
>> 'In the debug I see where it detects the port change from 183 #1 to 183
>> #2.  As such, I hear early media from both until the 200 OK.  When the call
>> connects, I see no such port change in the debug, and since it's still
>> listening on the wrong port (from 183 #2), there is no audio. I've seen
>> some older posts where Anthony seemed against even the port change from 183
>> #1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the
>> port change from 183 --> 200.'
>>
>> So it sounded like you heard both ringbacks and nothing after 200. In the
>> Jira you don't talk about the 200 at all, just issues before then. Do you
>> have audio working on receiving 200 now?
>>
>>
>>
>> On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>>
>>> Michael,
>>>
>>> I don't believe so.  I haven't seen any activity in the Jira
>>> <https://freeswitch.org/jira/browse/FS-7207>.
>>>
>>>
>>> - Jeff
>>>
>>>
>>> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
>>> m.stgeorges at csl-technologies.com> wrote:
>>>
>>>> Steven Ayre <steveayre at ...> writes:
>>>>
>>>> >
>>>> >
>>>> > This sounds like it belongs on Jira so the issue can be tracked.
>>>> >
>>>> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
>>>> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
>>>> > Hello,
>>>> > The following is on FreeSWITCH Version
>>>> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
>>>> 21:57:33Z 64bit).
>>>> >
>>>> > The design goal for this configuration is that of a simple transcoding
>>>> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
>>>> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
>>>> there is one call flow I'm struggling with.
>>>> >
>>>> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
>>>> signaling follows 100, 183, 200, all is well.  But if the gateway sends
>>>> multiple 183s with different RTP ports, there is no audio when the call
>>>> goes to 200.  See the following example:
>>>> >
>>>> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
>>>> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
>>>> signals 200 with SDP indicating audio on port 16384 (same as original
>>>> 183).
>>>> >
>>>> > In the debug I see where it detects the port change from 183 #1 to 183
>>>> #2.  As such, I hear early media from both until the 200 OK.  When the
>>>> call connects, I see no such port change in the debug, and since it's
>>>> still listening on the wrong port (from 183 #2), there is no audio.
>>>> >
>>>> > I've seen some older posts where Anthony seemed against even the port
>>>> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
>>>> just doesn't sense the port change from 183 --> 200.  I don't know if
>>>> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
>>>> >
>>>> >
>>>> > - Jeff
>>>> >
>>>> >
>>>> >
>>>> >
>>>> ________________________________________________________________________
>>>> _
>>>> > Professional FreeSWITCH Consulting Services:consulting-
>>>> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://
>>>> www.freeswitchsolutions.co
>>>> m
>>>> > Official FreeSWITCH
>>>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
>>>> .cluecon.com
>>>> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
>>>> lists.freeswitch.orghttp://
>>>> lists.freeswitch.org/mailman/listinfo/freeswi
>>>> tch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> usershttp://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> >
>>>> >
>>>> >
>>>> >
>>>> ________________________________________________________________________
>>>> _
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at ...
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at ...
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> > http://www.freeswitch.org
>>>>
>>>> Was this ever fixed? I have a similar problem with the behavior of a
>>>> Samsung PBX that connects to a FreeSwitch.
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/fbfcfcac/attachment-0001.html 

From krice at freeswitch.org  Mon Mar 16 21:54:17 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 16 Mar 2015 18:54:17 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March
	7th-14th
Message-ID: <550726d9b2c75_78ee49f32c61621@resque-worker-ip-10-31-77-28.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1CmA0mR
FreeSWITCH Week in Review (Master Branch) March 7th-14th
Hello, again. This passed week in the FreeSWITCH master branch we had 13 commits. The features for this week include: the addition of Australian ringback, a minimal configuration for configuring FreeSWITCH from scratch, and the filter feature ported from mod_event_socket to mod_erlang_event .

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7354 Filter feature ported from mod_event_socket to mod_erlang_event

FS-7362 Add minimal configuration for configuring FreeSWITCH from scratch

FS-7368 Added Australian ringback

The following bugs were squashed:

FS-7355 Fix rpl_realloc symbol missing link error that can occur when using clang.

FS-7300 Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs

?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/a18b172a/attachment.html 

From ali.jibran44 at gmail.com  Tue Mar 17 00:00:18 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Mon, 16 Mar 2015 21:00:18 +0000
Subject: [Freeswitch-users] DTMF (Mod-Call Center)
Message-ID: <CAP9PqkRHqu14387aDPwgb4wEtWCWYwMfdtt4Wdg1Gp2Qy+Zp6Q@mail.gmail.com>

I wanted to know if there is some way we could find out what DTMF was
pressed when agent picks up the the call.
Like if there is a queue say support at default on the same Extension but
differnet numbers..so how can I know what DTMF was pressed to enter it?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150316/6d144120/attachment.html 

From davidwaf at gmail.com  Tue Mar 17 02:29:36 2015
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 17 Mar 2015 01:29:36 +0200
Subject: [Freeswitch-users] Intercept chat message before forwarding it to
	receiver
Message-ID: <CAK9Ws2dK54T8TiRuAU29W8tqWpFU841Dk_den5YcnHV5JCmqpg@mail.gmail.com>

Dear all,
We are implementing custom messaging system that is using freeswitch.
I wish to intercept a chat message, prepend Country Code (or whatever i
want)  of the sender, then forward it to receiver.
How does one intercept a chat message ?

Regards,
-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/e784bb11/attachment.html 

From terry at digital-outpost.com  Tue Mar 17 05:49:41 2015
From: terry at digital-outpost.com (Terry Barnum)
Date: Mon, 16 Mar 2015 19:49:41 -0700
Subject: [Freeswitch-users] FS voicemail notification to mobile SMS?
In-Reply-To: <1426253032.3130.YahooMailBasic@web160205.mail.bf1.yahoo.com>
References: <1426253032.3130.YahooMailBasic@web160205.mail.bf1.yahoo.com>
Message-ID: <0FC1BEB2-22D4-4207-8641-2BD662C7C502@digital-outpost.com>

Thank you TF. Switching to the ATT and Verizon mms gateways solved a base64 decoding problem I was seeing on the sms gateways.

-Terry

> On Mar 13, 2015, at 6:23 AM, T Fred Farmington <tfred31 at yahoo.com> wrote:
> 
> Just a follow-up on sending Text messages through the email system.
> 
> If you are building a new contact list you might benefit from knowing which text gateway to use for which cell phone number.
> I do this by using one of the various web reference pages - the one I use mostly is:   http://www.freecarrierlookup.com/
> 
> Next you need to know the text gateway domain for that cell phone carrier.
> Again there are a number of references on the web - the one I use mostly is:   http://martinfitzpatrick.name/list-of-email-to-sms-gateways/
> 
> Also note that there is a difference between SMS and MMS
>   SMS you are most likely already familiar with
>   MMS is for more lengthy text messages
> Often (according to the text gateway reference page) these two different 'formats' utilize different text gateways.
> 
> TF
> 
> 
> --------------------------------------------
> On Thu, 3/12/15, Terry Barnum <terry at digital-outpost.com> wrote:
> 
> Subject: Re: [Freeswitch-users] FS voicemail notification to mobile SMS?
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, March 12, 2015, 4:52 PM
> 
> Thank you Shabbir.
> John's suggestion of just emailing is working great. In
> the US, ATT uses <10digitnumber>@txt.att.net to send a
> text and Verizon uses <10digitnumber>@vtext.com.
> 
> -Terry



From jack at yosin.com.tw  Tue Mar 17 08:24:34 2015
From: jack at yosin.com.tw (jack)
Date: Tue, 17 Mar 2015 13:24:34 +0800
Subject: [Freeswitch-users] read application in dialplan is notworking when
	I use bind_meta_app application ( FXS, TDM 400P with DAHDI )
Message-ID: <E897E82766CC4BA9A1158E828E07D74A@pca022>

To whom it may concern,

I just hit the wall when I test the atended transfer.
Here's my simple dialplan:

test.xml:

  <extension name="DID-Call">
      <condition field="destination_number" expression="^(382)$">    
          <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
          <action application="bridge" data="freetdm/myFXS/1"/>
      </condition>
  </extension>

features.xml:

  <extension name="att_xfer">
      <condition field="destination_number" expression="^att_xfer$">
           <action application="answer"/>
           <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/>
           <action application="att_xfer" data="user/${digits}"/>
      </condition>
  </extension>

And the test environment configuration is like this:

 

As you can see, when you dial number "382" from phone 282, the Diguim telephony card will get ring signal at once.
Then the freeSWITCH server starts a thread to handle dialplan. While in the execute-state, it will get in the test.xml and match the condition,
so FreeSwitch will help bridging two channels together (freetdm/myFXO and freetdm/myFXS/1).

After bridging, you'll hear the ring from phone 4001 and pick up the phone, then you can dial "*1" to activate the action and transfer to call featurs.xml immediately.
You're now in features.xml and matched the exact condition "att_xfer", so you will hear the generated dial tone by tone_stream and need to dial 4 digits in the moment.
The BIG problem is, I can't get any DTMF tone while hitting the digit keys on the phone 4001. But when I change the dialplan like this:

test.xml
  <extension name="DID-Call">
      <condition field="destination_number" expression="^(382)$">    
          <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
          <action application="bridge" data="user/1001"/>
      </condition>
  </extension>

The "read" works just fine and really do the addended transfer as expect.

And I do some tests to make sure the read function work well in normal context,
which means, I discard the "bind_meta_app" application, just use "read" when phone has been answered as follow:


test.xml
  <extension name="DID-Call">
      <condition field="destination_number" expression="^(382)$">    
          <action application="answer"/>
          <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/>
          <action application="log" data="The number you dialed is : ${digits}"/> 
      </condition>
  </extension>

and it works well when I dialed 382 and reach out the freeswitch server, I could see the DTMF tone show on console.
After I dialed "4001", the "log" application shows "The number you dialed is : 4001", just likes I expected.

So, why would I get involved with situation like this? I was checked the mod_dptools.c and nothing is found in there, since the arguments are not the trouble maker.
Maybe the critical part came from FreeTDM( samgoma ) Module, but I gotta figure out which direction of thought is the most probable. 
If you guys could give me some advice I'll be very thankful, thanks in advance!


Kind Regard,
Jack Huang
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/79f5782a/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/bmp
Size: 772182 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/79f5782a/attachment-0001.bmp 

From mbodbg at gmx.net  Tue Mar 17 11:37:58 2015
From: mbodbg at gmx.net (mbo)
Date: Tue, 17 Mar 2015 09:37:58 +0100
Subject: [Freeswitch-users] freeswitch sip trace to separate file
Message-ID: <94D0A0EC-89C6-4E8E-9048-E579FE195C89@gmx.net>

Hello,

is it possible to configure freeswitch to write the SIP Trace to a separate file, preferable different files per sip profile?  If no, is there already a feature request for that, otherwise I create one? 

Thanks and Regards

Markus




From avi at avimarcus.net  Tue Mar 17 11:51:58 2015
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 17 Mar 2015 08:51:58 +0000
Subject: [Freeswitch-users] freeswitch sip trace to separate file
In-Reply-To: <94D0A0EC-89C6-4E8E-9048-E579FE195C89@gmx.net>
References: <94D0A0EC-89C6-4E8E-9048-E579FE195C89@gmx.net>
Message-ID: <0000014c26eeaca2-fe1c97ae-e14e-4a9d-b57e-56d097623306-000000@email.amazonses.com>

I don't think there's anything native, unless you want to log it with the
normal fs_cli log.

See here: http://wiki.freeswitch.org/wiki/Packet_Capture (better formatting
than confluence).

Perhaps you'd be interested in ngrep or homer. Pcapsipdump is great if you
want the RTP, too.

-Avi

On Tue, Mar 17, 2015 at 10:37 AM, mbo <mbodbg at gmx.net> wrote:

> Hello,
>
> is it possible to configure freeswitch to write the SIP Trace to a
> separate file, preferable different files per sip profile?  If no, is there
> already a feature request for that, otherwise I create one?
>
> Thanks and Regards
>
> Markus
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/8f30240c/attachment.html 

From brian at freeswitch.org  Tue Mar 17 16:06:18 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 17 Mar 2015 08:06:18 -0500
Subject: [Freeswitch-users] Intercept chat message before forwarding it
	to receiver
In-Reply-To: <CAK9Ws2dK54T8TiRuAU29W8tqWpFU841Dk_den5YcnHV5JCmqpg@mail.gmail.com>
References: <CAK9Ws2dK54T8TiRuAU29W8tqWpFU841Dk_den5YcnHV5JCmqpg@mail.gmail.com>
Message-ID: <CAEJMVkBWbk67OVYdFyHW2Q=S1JkEuDagrCB5shdS0SU0qT0m1g@mail.gmail.com>

You would use the Chat Plan to accomplish this, I'm pretty sure examples
are on confluence or wiki.

On Mon, Mar 16, 2015 at 6:29 PM, David Wafula <davidwaf at gmail.com> wrote:

> Dear all,
> We are implementing custom messaging system that is using freeswitch.
> I wish to intercept a chat message, prepend Country Code (or whatever i
> want)  of the sender, then forward it to receiver.
> How does one intercept a chat message ?
>
> Regards,
> --
> David Wafula
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/d6cf8aa8/attachment.html 

From mike at jerris.com  Tue Mar 17 17:34:42 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Mar 2015 10:34:42 -0400
Subject: [Freeswitch-users] read application in dialplan is notworking
	when I use bind_meta_app application ( FXS, TDM 400P with DAHDI )
In-Reply-To: <E897E82766CC4BA9A1158E828E07D74A@pca022>
References: <E897E82766CC4BA9A1158E828E07D74A@pca022>
Message-ID: <D66FEBC9-27B2-4CC1-AF0F-C684236912ED@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA <https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA>

> On Mar 17, 2015, at 1:24 AM, jack <jack at yosin.com.tw> wrote:
> 
> To whom it may concern,
>  
> I just hit the wall when I test the atended transfer.
> Here's my simple dialplan:
>  
> test.xml:
>  
>   <extension name="DID-Call">
>       <condition field="destination_number" expression="^(382)$">    
>           <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
>           <action application="bridge" data="freetdm/myFXS/1"/>
>       </condition>
>   </extension>
>  
> features.xml:
>  
>   <extension name="att_xfer">
>       <condition field="destination_number" expression="^att_xfer$">
>            <action application="answer"/>
>            <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/>
>            <action application="att_xfer" data="user/${digits}"/>
>       </condition>
>   </extension>
>  
> And the test environment configuration is like this:
>  
>  <test configuration.bmp>
>  
> As you can see, when you dial number "382" from phone 282, the Diguim telephony card will get ring signal at once.
> Then the freeSWITCH server starts a thread to handle dialplan. While in the execute-state, it will get in the test.xml and match the condition,
> so FreeSwitch will help bridging two channels together (freetdm/myFXO and freetdm/myFXS/1).
>  
> After bridging, you'll hear the ring from phone 4001 and pick up the phone, then you can dial "*1" to activate the action and transfer to call featurs.xml immediately.
> You're now in features.xml and matched the exact condition "att_xfer", so you will hear the generated dial tone by tone_stream and need to dial 4 digits in the moment.
> The BIG problem is, I can't get any DTMF tone while hitting the digit keys on the phone 4001. But when I change the dialplan like this:
>  
> test.xml
>   <extension name="DID-Call">
>       <condition field="destination_number" expression="^(382)$">    
>           <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/>
>           <action application="bridge" data="user/1001"/>
>       </condition>
>   </extension>
>  
> The "read" works just fine and really do the addended transfer as expect.
>  
> And I do some tests to make sure the read function work well in normal context,
> which means, I discard the "bind_meta_app" application, just use "read" when phone has been answered as follow:
>  
>  
> test.xml
>   <extension name="DID-Call">
>       <condition field="destination_number" expression="^(382)$">    
>           <action application="answer"/>
>           <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/>
>           <action application="log" data="The number you dialed is : ${digits}"/> 
>       </condition>
>   </extension>
>  
> and it works well when I dialed 382 and reach out the freeswitch server, I could see the DTMF tone show on console.
> After I dialed "4001", the "log" application shows "The number you dialed is : 4001", just likes I expected.
>  
> So, why would I get involved with situation like this? I was checked the mod_dptools.c and nothing is found in there, since the arguments are not the trouble maker.
> Maybe the critical part came from FreeTDM( samgoma ) Module, but I gotta figure out which direction of thought is the most probable. 
> If you guys could give me some advice I'll be very thankful, thanks in advance!
>  
>  
> Kind Regard,
> Jack Huang
>  
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/bde7a4a9/attachment-0001.html 

From jpyle at fidelityvoice.com  Tue Mar 17 17:50:55 2015
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Tue, 17 Mar 2015 10:50:55 -0400
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
 after 200
In-Reply-To: <CACYJG3J9cauAROt3645JZNXXhMnbs1ewNoZyEYfE96N3uyMWwg@mail.gmail.com>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
	<CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
	<CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>
	<CAEJMVkBj-06nPxCVZkpPv3K-m=H7=9XuaNOWf3jxnWc16gDnOg@mail.gmail.com>
	<CACYJG3J9cauAROt3645JZNXXhMnbs1ewNoZyEYfE96N3uyMWwg@mail.gmail.com>
Message-ID: <CACYJG3LeCSrWEG7=KF8dveO_YUA3A=sxxFCEEwd8Pq6ZRLhEsw@mail.gmail.com>

Confirmed, this issue is still present in FreeSWITCH Version
1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13
21:50:13Z 64bit).


- Jeff

On Mon, Mar 16, 2015 at 11:58 AM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> They're two separate issues.  It looks like I referenced the wrong Jira in
> my reply to Michael's comment.  The correct Jira for this issue, involving
> the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>.
> Oops.  I'll retest shortly.
>
> Agreed, the 180-after-183 no audio issue described in FS-7207
> <https://freeswitch.org/jira/browse/FS-7207> is clearly different.  Sorry
> for the confusion.
>
>
>
> - Jeff
>
> On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian at freeswitch.org> wrote:
>
>> If that is what is going on, then thats a critical detail that wasn't
>> outlined in the JIRA, maybe we can get a new test on the latest code to
>> verify?
>>
>> On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>>
>>> Just an observation, but what you wrote in the Jira appears to be
>>> different from your original email.
>>>
>>> 'In the debug I see where it detects the port change from 183 #1 to 183
>>> #2.  As such, I hear early media from both until the 200 OK.  When the call
>>> connects, I see no such port change in the debug, and since it's still
>>> listening on the wrong port (from 183 #2), there is no audio. I've seen
>>> some older posts where Anthony seemed against even the port change from 183
>>> #1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the
>>> port change from 183 --> 200.'
>>>
>>> So it sounded like you heard both ringbacks and nothing after 200. In
>>> the Jira you don't talk about the 200 at all, just issues before then. Do
>>> you have audio working on receiving 200 now?
>>>
>>>
>>>
>>> On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>>>
>>>> Michael,
>>>>
>>>> I don't believe so.  I haven't seen any activity in the Jira
>>>> <https://freeswitch.org/jira/browse/FS-7207>.
>>>>
>>>>
>>>> - Jeff
>>>>
>>>>
>>>> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <
>>>> m.stgeorges at csl-technologies.com> wrote:
>>>>
>>>>> Steven Ayre <steveayre at ...> writes:
>>>>>
>>>>> >
>>>>> >
>>>>> > This sounds like it belongs on Jira so the issue can be tracked.
>>>>> >
>>>>> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
>>>>> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org> wrote:
>>>>> > Hello,
>>>>> > The following is on FreeSWITCH Version
>>>>> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
>>>>> 21:57:33Z 64bit).
>>>>> >
>>>>> > The design goal for this configuration is that of a simple
>>>>> transcoding
>>>>> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
>>>>> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
>>>>> there is one call flow I'm struggling with.
>>>>> >
>>>>> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
>>>>> signaling follows 100, 183, 200, all is well.  But if the gateway sends
>>>>> multiple 183s with different RTP ports, there is no audio when the call
>>>>> goes to 200.  See the following example:
>>>>> >
>>>>> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
>>>>> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
>>>>> signals 200 with SDP indicating audio on port 16384 (same as original
>>>>> 183).
>>>>> >
>>>>> > In the debug I see where it detects the port change from 183 #1 to
>>>>> 183
>>>>> #2.  As such, I hear early media from both until the 200 OK.  When the
>>>>> call connects, I see no such port change in the debug, and since it's
>>>>> still listening on the wrong port (from 183 #2), there is no audio.
>>>>> >
>>>>> > I've seen some older posts where Anthony seemed against even the port
>>>>> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
>>>>> just doesn't sense the port change from 183 --> 200.  I don't know if
>>>>> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
>>>>> >
>>>>> >
>>>>> > - Jeff
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>>
>>>>> ________________________________________________________________________
>>>>> _
>>>>> > Professional FreeSWITCH Consulting Services:consulting-
>>>>> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://
>>>>> www.freeswitchsolutions.co
>>>>> m
>>>>> > Official FreeSWITCH
>>>>>
>>>>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
>>>>> .cluecon.com
>>>>> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
>>>>> lists.freeswitch.orghttp://
>>>>> lists.freeswitch.org/mailman/listinfo/freeswi
>>>>> tch-users
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> usershttp://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>>
>>>>> ________________________________________________________________________
>>>>> _
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at ...
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://confluence.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at ...
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>> Was this ever fixed? I have a similar problem with the behavior of a
>>>>> Samsung PBX that connects to a FreeSwitch.
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/7ddafe51/attachment.html 

From mike at jerris.com  Tue Mar 17 18:04:58 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Mar 2015 11:04:58 -0400
Subject: [Freeswitch-users] Port changes in multiple 183s cause no audio
	after 200
In-Reply-To: <CACYJG3LeCSrWEG7=KF8dveO_YUA3A=sxxFCEEwd8Pq6ZRLhEsw@mail.gmail.com>
References: <CACYJG3L3_dA7ZhR0F79LEwz7Fx+FD1FgBSWST+y4vA2_KDChkA@mail.gmail.com>
	<CAFiqYunJQQdU+_r1_RAUuT7oRVdE-dCCAnZoCzoJS-QeYbkn6Q@mail.gmail.com>
	<loom.20150314T031658-719@post.gmane.org>
	<CACYJG3JotqSgsBzyn5P4QsTn_hv2A3-hT_rCobG-pBfanmv+0Q@mail.gmail.com>
	<CAFiqYum=dB_Enpot=y4KUvBt-Q4L2cmHKy1mudvT_wQSqB1hPw@mail.gmail.com>
	<CAEJMVkBj-06nPxCVZkpPv3K-m=H7=9XuaNOWf3jxnWc16gDnOg@mail.gmail.com>
	<CACYJG3J9cauAROt3645JZNXXhMnbs1ewNoZyEYfE96N3uyMWwg@mail.gmail.com>
	<CACYJG3LeCSrWEG7=KF8dveO_YUA3A=sxxFCEEwd8Pq6ZRLhEsw@mail.gmail.com>
Message-ID: <72FF9A17-031A-4643-BA0A-4A056C126DEB@jerris.com>

please follow up on the jira, not here.

Thanks
Mike

> On Mar 17, 2015, at 10:50 AM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
> 
> Confirmed, this issue is still present in FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13 21:50:13Z 64bit).
> 
> 
> - Jeff
> 
> On Mon, Mar 16, 2015 at 11:58 AM, Jeff Pyle <jpyle at fidelityvoice.com <mailto:jpyle at fidelityvoice.com>> wrote:
> They're two separate issues.  It looks like I referenced the wrong Jira in my reply to Michael's comment.  The correct Jira for this issue, involving the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>.  Oops.  I'll retest shortly.
> 
> Agreed, the 180-after-183 no audio issue described in FS-7207 <https://freeswitch.org/jira/browse/FS-7207> is clearly different.  Sorry for the confusion.
> 
> 
> 
> - Jeff
> 
> On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
> If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify?
> 
> On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
> Just an observation, but what you wrote in the Jira appears to be different from your original email.
> 
> 'In the debug I see where it detects the port change from 183 #1 to 183 #2.  As such, I hear early media from both until the 200 OK.  When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today.  It just doesn't sense the port change from 183 --> 200.'
> 
> So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?
> 
> 
> 
> On 16 March 2015 at 14:18, Jeff Pyle <jpyle at fidelityvoice.com <mailto:jpyle at fidelityvoice.com>> wrote:
> Michael,
> 
> I don't believe so.  I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.  
> 
> 
> - Jeff
> 
> 
> On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <m.stgeorges at csl-technologies.com <mailto:m.stgeorges at csl-technologies.com>> wrote:
> Steven Ayre <steveayre at ...> writes:
> 
> >
> >
> > This sounds like it belongs on Jira so the issue can be tracked.
> >
> > On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
> eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf at public.gmane.org <mailto:G2K4zDHf at public.gmane.org>> wrote:
> > Hello,
> > The following is on FreeSWITCH Version
> 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26
> 21:57:33Z 64bit).
> >
> > The design goal for this configuration is that of a simple transcoding
> SBC.  SIP calls arrive with various supported codecs, SIP calls bridge
> out on PCMU.  No users, no auth, etc.  Overall, it seems to work but
> there is one call flow I'm struggling with.
> >
> > The B-leg of calls are bridged to a PSTN gateway.  If the gateway's
> signaling follows 100, 183, 200, all is well.  But if the gateway sends
> multiple 183s with different RTP ports, there is no audio when the call
> goes to 200.  See the following example:
> >
> >  - Gateway signals 183 with SDP indicating audio on port 16384. -
> Gateway signals 183 with SDP indicating audio on port 16386. - Gateway
> signals 200 with SDP indicating audio on port 16384 (same as original
> 183).
> >
> > In the debug I see where it detects the port change from 183 #1 to 183
> #2.  As such, I hear early media from both until the 200 OK.  When the
> call connects, I see no such port change in the debug, and since it's
> still listening on the wrong port (from 183 #2), there is no audio.
> >
> > I've seen some older posts where Anthony seemed against even the port
> change from 183 #1 to 183 #2, yet that seems to work okay today.  It
> just doesn't sense the port change from 183 --> 200.  I don't know if
> this is a feature, bug, or misconfigured option.  Thoughts are welcome!
> >
> >
> > - Jeff
> >
> >
> >
> >
> ________________________________________________________________________
> _
> > Professional FreeSWITCH Consulting Services:consulting-
> YF8E+gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.co <http://www.freeswitchsolutions.co/>
> m
> > Official FreeSWITCH
> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www
> .cluecon.com <http://cluecon.com/>
> > FreeSWITCH-users mailing listFreeSWITCH-users <at>
> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi <http://lists.freeswitch.org/mailman/listinfo/freeswi>
> tch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- <http://lists.freeswitch.org/mailman/options/freeswitch->
> usershttp://www.freeswitch.org <http://www.freeswitch.org/>
> >
> >
> >
> >
> >
> >
> >
> ________________________________________________________________________
> _
> > Professional FreeSWITCH Consulting Services:
> > consulting at ...
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at ...
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- <http://lists.freeswitch.org/mailman/options/freeswitch->
> users
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> Was this ever fixed? I have a similar problem with the behavior of a
> Samsung PBX that connects to a 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/a0b06345/attachment-0001.html 

From jpyle at fidelityvoice.com  Tue Mar 17 18:10:44 2015
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Tue, 17 Mar 2015 11:10:44 -0400
Subject: [Freeswitch-users] delay before ringback at 183 to 180 transition
Message-ID: <CACYJG3KiyfTnjGwqpP0XcG1fNeQM4O_z9vFk_8Mef1ss=rQ1Dg@mail.gmail.com>

Hello,

FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git
9872c52 2015-03-13 21:50:13Z 64bit)

I have a situation where a gateway sends a 183 with media, then a 180.
Following Anthony's advice from FS-3859 I have the following:

  <extension name="sendit">
    <condition field="destination_number" expression="(.*)">
      <action application="set"
data="ringback=/usr/local/freeswitch/sounds/ringtester.wav"/>
      <action application="bridge"
data="{bridge_early_media=true}sofia/public/$1@${host2}"/>
    </condition>
  </extension>

On the a-leg I hear media from the 183 until host2 stops it and sends the
180.  Then there's a 4.8 second delay of no RTP from FS before I hear my
ringtester.wav file.  This occurs with a tone stream as well.

The last event on the console is "Callstate Change EARLY -> RINGING" when
the 180 comes in, then "nua.c:366 nua_handle_magic() nua: nua_handle_magic:
entering" when the 180 arrives.  Then 4.8s of nothing.  Then FS resumes the
RTP with the ringback file.  There is nothing new displayed when the RTP
resumes with the ringback file.

Is this a bug, or a misconfiguration?


- Jeff
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/00534d59/attachment.html 

From anthony.minessale at gmail.com  Tue Mar 17 18:56:46 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 17 Mar 2015 10:56:46 -0500
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
Message-ID: <CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>

Start by always testing on master.  Today's master is tomorrows release.
Todays release is yesterdays problem that won't be solved.
secondly, report issues to JIRA (after reproducing on master)


On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson <jdickson at evolvetsi.com>
wrote:

> Yes, I can definitely see where master-master mode in MySQL can put
> you in a pickle consistency wise..  I just wish Postgres was easier
> with regard to fail-back.. failover isn't a problem, but getting
> things to fail back requires rsync of data files, etc..
>
> Before I set that up, I'd like to see sofia recover work in my
> environment..  Currently it appears that the correct information is
> being retrieved from the database, but that FS is hanging up the A leg
> with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
> behind that hangup.
>
> Is there better logging that I can grab that might be more helpful?
>
> Thanks!
>
> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
> > I has used MySQL in master-master cluster and find it is unstable
> > configuration. Many troubles with cluster consistence.
> > I recommend use configuration master-slave with tools DB synchronization.
> >
> > Also migration to PostgreSQL cluster will be smart decision.
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/16ba7811/attachment.html 

From jdickson at evolvetsi.com  Tue Mar 17 19:01:54 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Tue, 17 Mar 2015 12:01:54 -0400
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
	<CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>
Message-ID: <CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>

Understood.. I'll spend some time this week getting things compiled from
source instead of using the binary packages, and see if behavior is the
same.

On Tue, Mar 17, 2015 at 11:56 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Start by always testing on master.  Today's master is tomorrows release.
> Todays release is yesterdays problem that won't be solved.
> secondly, report issues to JIRA (after reproducing on master)
>
>
> On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson <jdickson at evolvetsi.com>
> wrote:
>
>> Yes, I can definitely see where master-master mode in MySQL can put
>> you in a pickle consistency wise..  I just wish Postgres was easier
>> with regard to fail-back.. failover isn't a problem, but getting
>> things to fail back requires rsync of data files, etc..
>>
>> Before I set that up, I'd like to see sofia recover work in my
>> environment..  Currently it appears that the correct information is
>> being retrieved from the database, but that FS is hanging up the A leg
>> with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
>> behind that hangup.
>>
>> Is there better logging that I can grab that might be more helpful?
>>
>> Thanks!
>>
>> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>> > I has used MySQL in master-master cluster and find it is unstable
>> > configuration. Many troubles with cluster consistence.
>> > I recommend use configuration master-slave with tools DB
>> synchronization.
>> >
>> > Also migration to PostgreSQL cluster will be smart decision.
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/4dde42ad/attachment.html 

From alipey at gmail.com  Tue Mar 17 19:04:34 2015
From: alipey at gmail.com (Ali Pey)
Date: Tue, 17 Mar 2015 12:04:34 -0400
Subject: [Freeswitch-users] park_after_bridge does not work if far end does
	180 Ringing
Message-ID: <CA+q4kRJCNT7LKSMVGKh7fL73vk_OdT_fwqX6ocGgZvTQkb1wDQ@mail.gmail.com>

Hello,

I see a different behaviour for 183 Ringing vs 180 Ringing and a 408
Request Timeout response message with regards to park_after_bridge:

1) I have "park_after_bridge=true".
2) Do a bridge to a PSTN number (SIP Trunk)
3) After sometime freeswitch receives 408 Request Timeout message:
    a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg
is parked after the 408 timeout message.
    b) if carrier had sent 180 Ringing, the aleg is hungup after timeout
(408 message)


Is this a bug?
Is there a work around?

Thanks,
Ali Pey
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/e44224e9/attachment.html 

From alhakeem at gmail.com  Tue Mar 17 19:15:39 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Tue, 17 Mar 2015 16:15:39 -0000
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>

Hello,

Is there any way to override FS behaviour when no ACK is received by FS from
registered clients?
I'm hoping there might be a way for FS to continue processing rather than end
the call.

Cheers,
Abdul Hakeem




From mike at jerris.com  Tue Mar 17 20:00:36 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Mar 2015 13:00:36 -0400
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
Message-ID: <A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>

No, and there shouldn't/can't be.  If you try to continue a call that has nat issues keeping responses from going through, there will be no way to end the call.

> On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello,
> 
> Is there any way to override FS behaviour when no ACK is received by FS from
> registered clients?
> I'm hoping there might be a way for FS to continue processing rather than end
> the call.
> 
> Cheers,
> Abdul Hakeem



From italorossib at gmail.com  Tue Mar 17 20:34:18 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 17 Mar 2015 14:34:18 -0300
Subject: [Freeswitch-users] park_after_bridge does not work if far end
 does 180 Ringing
In-Reply-To: <CA+q4kRJCNT7LKSMVGKh7fL73vk_OdT_fwqX6ocGgZvTQkb1wDQ@mail.gmail.com>
References: <CA+q4kRJCNT7LKSMVGKh7fL73vk_OdT_fwqX6ocGgZvTQkb1wDQ@mail.gmail.com>
Message-ID: <CAJR7f2F581qLKmc8UjJ46Co_9XAEe8C0KjH9yK-6yzq5ZR5kVA@mail.gmail.com>

183 means early media and the bridge by default is considered successful by
FreeSWITCH -->
https://freeswitch.org/confluence/display/FREESWITCH/Early+Media#EarlyMedia-A
"Successful"CallAttempt

So park_after_bridge will only park the call if FreeSWITCH has received 183
or 200 from far end.

Not a bug, expected behavior.

Let me know if I'm wrong

On Tue, Mar 17, 2015 at 1:04 PM, Ali Pey <alipey at gmail.com> wrote:

> Hello,
>
> I see a different behaviour for 183 Ringing vs 180 Ringing and a 408
> Request Timeout response message with regards to park_after_bridge:
>
> 1) I have "park_after_bridge=true".
> 2) Do a bridge to a PSTN number (SIP Trunk)
> 3) After sometime freeswitch receives 408 Request Timeout message:
>     a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg
> is parked after the 408 timeout message.
>     b) if carrier had sent 180 Ringing, the aleg is hungup after timeout
> (408 message)
>
>
> Is this a bug?
> Is there a work around?
>
> Thanks,
> Ali Pey
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/e306d5e7/attachment.html 

From patrick.shea at citrix.com  Tue Mar 17 20:02:39 2015
From: patrick.shea at citrix.com (Patrick Shea)
Date: Tue, 17 Mar 2015 17:02:39 +0000
Subject: [Freeswitch-users] Disable Re-INVITE on REFER ?
Message-ID: <D2117D269EE53A49B1CB699FE2E7B3241F2BBC28@SJCPEX01CL01.citrite.net>

?I am seeing 2 problems that I can't seem to find the answers to. I've scoured the Freeswitch Wiki (old), Confluence, and the users list.

I have 2 SIP calls bridge through Freeswitch - they are CONTROL channels, so no media is negotiated. A Leg is SIP over Websockets, B Leg is SIP.

SIPjs (WS) ?<---- A Leg ---> Freeswitch ?<---- B Leg ---> EP

The problem I have is in some cases I get a REFER from the EP to transfer the call somewhere else, and this results in a Re-INVITE from Freeswitch to the SIPjs client. (Ideally, I'd like to "skip" this re-Invite and just bridge the A-leg to the new AP).

Problem 1.?
Freeswitch sends media in the SDP of the Re-INVITE. This doesn't look right to me, but I assume using some local codecs specified somewhere.?
      m=audio 28692 RTP/AVP 0 8
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
Can I disable these and just send the original m= line received for the initial call?
      m=application 28118 udp as

Problem 2.
SIPjs doesn't support Re-INVITE, yet, so it responds with 488 Unacceptable Here. At this point, Freeswitch, drops the A leg and the transfer fails. I can't find a solution to work around this - any ideas?

Like I said, Ideally I'd like to skip the re-INVITE altogether. Currently debugging SOFIA to see how this can be done - is there a parameter already somewhere that I am overlooking?

Thanks
Patrick


From mike at jerris.com  Tue Mar 17 21:11:41 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Mar 2015 14:11:41 -0400
Subject: [Freeswitch-users] park_after_bridge does not work if far end
	does 180 Ringing
In-Reply-To: <CAJR7f2F581qLKmc8UjJ46Co_9XAEe8C0KjH9yK-6yzq5ZR5kVA@mail.gmail.com>
References: <CA+q4kRJCNT7LKSMVGKh7fL73vk_OdT_fwqX6ocGgZvTQkb1wDQ@mail.gmail.com>
	<CAJR7f2F581qLKmc8UjJ46Co_9XAEe8C0KjH9yK-6yzq5ZR5kVA@mail.gmail.com>
Message-ID: <3CF03FA5-ECDC-41E0-A8EF-E88A84F51E20@jerris.com>

if you ignore_early_media you can make 180 and 183 act the same.

> On Mar 17, 2015, at 1:34 PM, ?talo Rossi <italorossib at gmail.com> wrote:
> 
> 183 means early media and the bridge by default is considered successful by FreeSWITCH -->  https://freeswitch.org/confluence/display/FREESWITCH/Early+Media#EarlyMedia-A <https://freeswitch.org/confluence/display/FREESWITCH/Early+Media#EarlyMedia-A>"Successful"CallAttempt
> 
> So park_after_bridge will only park the call if FreeSWITCH has received 183 or 200 from far end.
> 
> Not a bug, expected behavior.
> 
> Let me know if I'm wrong
> 
> On Tue, Mar 17, 2015 at 1:04 PM, Ali Pey <alipey at gmail.com <mailto:alipey at gmail.com>> wrote:
> Hello,
> 
> I see a different behaviour for 183 Ringing vs 180 Ringing and a 408 Request Timeout response message with regards to park_after_bridge:
> 
> 1) I have "park_after_bridge=true".
> 2) Do a bridge to a PSTN number (SIP Trunk)
> 3) After sometime freeswitch receives 408 Request Timeout message:
>     a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg is parked after the 408 timeout message.
>     b) if carrier had sent 180 Ringing, the aleg is hungup after timeout (408 message)
> 
> 
> Is this a bug?
> Is there a work around?
> 
> Thanks,
> Ali Pey
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/bf0d07bd/attachment.html 

From pjintheusa at gmail.com  Tue Mar 17 22:05:10 2015
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 17 Mar 2015 15:05:10 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
Message-ID: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>

Hi All,

The following dial plan detects RFC 2833 and if it is not present, runs the
"start_dtmf" application.

<condition field="${switch_r_sdp}"
expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
<action application="set" data="rtp_payload_number=$1"/>
<anti-action application="start_dtmf"/>
</condition>

I need to do this on the outbound leg - using Lua. Can anyone point me to
some code example that would achieve this or get me started?

Any help appreciated.

Thanks



Phil
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/184a9fe5/attachment.html 

From cmrienzo at gmail.com  Tue Mar 17 22:10:46 2015
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 17 Mar 2015 15:10:46 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
Message-ID: <CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>

You could set the "execute_on_answer" variable on the b-leg dialstring to
execute the lua script.  In the lua script, you'd use
session:getVariable("switch_r_sdp") and then some pattern matching to
figure out if "telephone-event" is in the SDP.  If not, then do
session:execute("start_dtmf", "")

Chris


On Tue, Mar 17, 2015 at 3:05 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi All,
>
> The following dial plan detects RFC 2833 and if it is not present, runs
> the "start_dtmf" application.
>
> <condition field="${switch_r_sdp}"
> expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
> <action application="set" data="rtp_payload_number=$1"/>
> <anti-action application="start_dtmf"/>
> </condition>
>
> I need to do this on the outbound leg - using Lua. Can anyone point me to
> some code example that would achieve this or get me started?
>
> Any help appreciated.
>
> Thanks
>
>
>
> Phil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/f0b695cd/attachment-0001.html 

From pjintheusa at gmail.com  Tue Mar 17 22:29:43 2015
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 17 Mar 2015 15:29:43 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
	<CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
Message-ID: <CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>

Thanks Chris.

So something like.

s = session:getVariable("switch_r_sdp")

if string.find(s, "telephone-event/8000") == nil then

    session.execute("start_dtfm")

end

Is telephone-event/8000 the correct thing to look for? I guess it is from
the dial plan example.


On Tue, Mar 17, 2015 at 3:10 PM, Christopher Rienzo <cmrienzo at gmail.com>
wrote:

> You could set the "execute_on_answer" variable on the b-leg dialstring to
> execute the lua script.  In the lua script, you'd use
> session:getVariable("switch_r_sdp") and then some pattern matching to
> figure out if "telephone-event" is in the SDP.  If not, then do
> session:execute("start_dtmf", "")
>
> Chris
>
>
> On Tue, Mar 17, 2015 at 3:05 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
>
>> Hi All,
>>
>> The following dial plan detects RFC 2833 and if it is not present, runs
>> the "start_dtmf" application.
>>
>> <condition field="${switch_r_sdp}"
>> expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
>> <action application="set" data="rtp_payload_number=$1"/>
>> <anti-action application="start_dtmf"/>
>> </condition>
>>
>> I need to do this on the outbound leg - using Lua. Can anyone point me to
>> some code example that would achieve this or get me started?
>>
>> Any help appreciated.
>>
>> Thanks
>>
>>
>>
>> Phil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/d89e134a/attachment.html 

From cmrienzo at gmail.com  Tue Mar 17 22:44:09 2015
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 17 Mar 2015 15:44:09 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
	<CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
	<CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
Message-ID: <CAD-qRkhiYkVws0hFO4h8m0E=isqf4B+Df_=eJ8PKWdi9zG_L=Q@mail.gmail.com>

Use session:execute("start_dtmf", "") and it should work.

Chris



On Tue, Mar 17, 2015 at 3:29 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Thanks Chris.
>
> So something like.
>
> s = session:getVariable("switch_r_sdp")
>
> if string.find(s, "telephone-event/8000") == nil then
>
>     session.execute("start_dtfm")
>
> end
>
> Is telephone-event/8000 the correct thing to look for? I guess it is from
> the dial plan example.
>
>
> On Tue, Mar 17, 2015 at 3:10 PM, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
>
>> You could set the "execute_on_answer" variable on the b-leg dialstring to
>> execute the lua script.  In the lua script, you'd use
>> session:getVariable("switch_r_sdp") and then some pattern matching to
>> figure out if "telephone-event" is in the SDP.  If not, then do
>> session:execute("start_dtmf", "")
>>
>> Chris
>>
>>
>> On Tue, Mar 17, 2015 at 3:05 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>
>>> Hi All,
>>>
>>> The following dial plan detects RFC 2833 and if it is not present, runs
>>> the "start_dtmf" application.
>>>
>>> <condition field="${switch_r_sdp}"
>>> expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
>>> <action application="set" data="rtp_payload_number=$1"/>
>>> <anti-action application="start_dtmf"/>
>>> </condition>
>>>
>>> I need to do this on the outbound leg - using Lua. Can anyone point me
>>> to some code example that would achieve this or get me started?
>>>
>>> Any help appreciated.
>>>
>>> Thanks
>>>
>>>
>>>
>>> Phil
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/5be3ec24/attachment.html 

From alipey at gmail.com  Tue Mar 17 23:33:47 2015
From: alipey at gmail.com (Ali Pey)
Date: Tue, 17 Mar 2015 16:33:47 -0400
Subject: [Freeswitch-users] park_after_bridge does not work if far end
 does 180 Ringing
In-Reply-To: <3CF03FA5-ECDC-41E0-A8EF-E88A84F51E20@jerris.com>
References: <CA+q4kRJCNT7LKSMVGKh7fL73vk_OdT_fwqX6ocGgZvTQkb1wDQ@mail.gmail.com>
	<CAJR7f2F581qLKmc8UjJ46Co_9XAEe8C0KjH9yK-6yzq5ZR5kVA@mail.gmail.com>
	<3CF03FA5-ECDC-41E0-A8EF-E88A84F51E20@jerris.com>
Message-ID: <CA+q4kRKNLJiXr+Kk9vDgovJ8ibiYQkXasc=2ZcMH1+cTQ265Cw@mail.gmail.com>

I use uuid_bridge to connect two uuids. I have set park_after_bridge=true.

No matter if I set ignore_early_media to true or false, if b-leg does 183
Ringing and then times out (408 timeout response), a-leg will be parked. If
b-leg does 180 Ringing and then times out, a-leg is dropped.

How can I make sure a-leg will always be parked if b-leg is parked?

This seems a bug to me.

Regards,
Ali Pey


On Tue, Mar 17, 2015 at 2:11 PM, Michael Jerris <mike at jerris.com> wrote:

> if you ignore_early_media you can make 180 and 183 act the same.
>
> On Mar 17, 2015, at 1:34 PM, ?talo Rossi <italorossib at gmail.com> wrote:
>
> 183 means early media and the bridge by default is considered successful
> by FreeSWITCH -->
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media#EarlyMedia-A
> "Successful"CallAttempt
>
> So park_after_bridge will only park the call if FreeSWITCH has received
> 183 or 200 from far end.
>
> Not a bug, expected behavior.
>
> Let me know if I'm wrong
>
> On Tue, Mar 17, 2015 at 1:04 PM, Ali Pey <alipey at gmail.com> wrote:
>
>> Hello,
>>
>> I see a different behaviour for 183 Ringing vs 180 Ringing and a 408
>> Request Timeout response message with regards to park_after_bridge:
>>
>> 1) I have "park_after_bridge=true".
>> 2) Do a bridge to a PSTN number (SIP Trunk)
>> 3) After sometime freeswitch receives 408 Request Timeout message:
>>     a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg
>> is parked after the 408 timeout message.
>>     b) if carrier had sent 180 Ringing, the aleg is hungup after timeout
>> (408 message)
>>
>>
>> Is this a bug?
>> Is there a work around?
>>
>> Thanks,
>> Ali Pey
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/73d83867/attachment-0001.html 

From steveayre at gmail.com  Tue Mar 17 23:38:40 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 17 Mar 2015 20:38:40 +0000
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
	<CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
	<CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
Message-ID: <CAFiqYumQo2V=b0eDwNXfzm0EU9EZ=qN+m2_6HtfTWzB4=Hk-+w@mail.gmail.com>

It is. The regex checking a=rtpmap too would be more 'correct', but in
practice you'll only see that string on that line.

Since telephone-event has no static payload type assigned you'll always see
the a=rtpmap line for it, so you don't need to worry about non-verbose SDP.


On 17 March 2015 at 19:29, Phillip Jones <pjintheusa at gmail.com> wrote:

> Thanks Chris.
>
> So something like.
>
> s = session:getVariable("switch_r_sdp")
>
> if string.find(s, "telephone-event/8000") == nil then
>
>     session.execute("start_dtfm")
>
> end
>
> Is telephone-event/8000 the correct thing to look for? I guess it is from
> the dial plan example.
>
>
> On Tue, Mar 17, 2015 at 3:10 PM, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
>
>> You could set the "execute_on_answer" variable on the b-leg dialstring to
>> execute the lua script.  In the lua script, you'd use
>> session:getVariable("switch_r_sdp") and then some pattern matching to
>> figure out if "telephone-event" is in the SDP.  If not, then do
>> session:execute("start_dtmf", "")
>>
>> Chris
>>
>>
>> On Tue, Mar 17, 2015 at 3:05 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>
>>> Hi All,
>>>
>>> The following dial plan detects RFC 2833 and if it is not present, runs
>>> the "start_dtmf" application.
>>>
>>> <condition field="${switch_r_sdp}"
>>> expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
>>> <action application="set" data="rtp_payload_number=$1"/>
>>> <anti-action application="start_dtmf"/>
>>> </condition>
>>>
>>> I need to do this on the outbound leg - using Lua. Can anyone point me
>>> to some code example that would achieve this or get me started?
>>>
>>> Any help appreciated.
>>>
>>> Thanks
>>>
>>>
>>>
>>> Phil
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/a93a58ac/attachment.html 

From davidwaf at gmail.com  Tue Mar 17 23:58:58 2015
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 17 Mar 2015 22:58:58 +0200
Subject: [Freeswitch-users] Intercept chat message before forwarding it
	to receiver
In-Reply-To: <CAEJMVkBWbk67OVYdFyHW2Q=S1JkEuDagrCB5shdS0SU0qT0m1g@mail.gmail.com>
References: <CAK9Ws2dK54T8TiRuAU29W8tqWpFU841Dk_den5YcnHV5JCmqpg@mail.gmail.com>
	<CAEJMVkBWbk67OVYdFyHW2Q=S1JkEuDagrCB5shdS0SU0qT0m1g@mail.gmail.com>
Message-ID: <CAK9Ws2fW41hjiMmYiXA1gyPRFY9Av+jMx1eHYWFCs2OjsLRFww@mail.gmail.com>

OH cool. Chatplan it is.
Thanks!

On Tue, Mar 17, 2015 at 3:06 PM, Brian West <brian at freeswitch.org> wrote:

> You would use the Chat Plan to accomplish this, I'm pretty sure examples
> are on confluence or wiki.
>
> On Mon, Mar 16, 2015 at 6:29 PM, David Wafula <davidwaf at gmail.com> wrote:
>
>> Dear all,
>> We are implementing custom messaging system that is using freeswitch.
>> I wish to intercept a chat message, prepend Country Code (or whatever i
>> want)  of the sender, then forward it to receiver.
>> How does one intercept a chat message ?
>>
>> Regards,
>> --
>> David Wafula
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David Wafula
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/d8208cd9/attachment.html 

From pjintheusa at gmail.com  Tue Mar 17 23:59:37 2015
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 17 Mar 2015 16:59:37 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAFiqYumQo2V=b0eDwNXfzm0EU9EZ=qN+m2_6HtfTWzB4=Hk-+w@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
	<CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
	<CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
	<CAFiqYumQo2V=b0eDwNXfzm0EU9EZ=qN+m2_6HtfTWzB4=Hk-+w@mail.gmail.com>
Message-ID: <CAGK4cYyNxr9WVwmBP0=Wv4QkCJ2-7WGBafgSvWahbhH3MATb4g@mail.gmail.com>

Appreciated. Thanks

On Tue, Mar 17, 2015 at 4:38 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It is. The regex checking a=rtpmap too would be more 'correct', but in
> practice you'll only see that string on that line.
>
> Since telephone-event has no static payload type assigned you'll always
> see the a=rtpmap line for it, so you don't need to worry about non-verbose
> SDP.
>
>
> On 17 March 2015 at 19:29, Phillip Jones <pjintheusa at gmail.com> wrote:
>
>> Thanks Chris.
>>
>> So something like.
>>
>> s = session:getVariable("switch_r_sdp")
>>
>> if string.find(s, "telephone-event/8000") == nil then
>>
>>     session.execute("start_dtfm")
>>
>> end
>>
>> Is telephone-event/8000 the correct thing to look for? I guess it is
>> from the dial plan example.
>>
>>
>> On Tue, Mar 17, 2015 at 3:10 PM, Christopher Rienzo <cmrienzo at gmail.com>
>> wrote:
>>
>>> You could set the "execute_on_answer" variable on the b-leg dialstring
>>> to execute the lua script.  In the lua script, you'd use
>>> session:getVariable("switch_r_sdp") and then some pattern matching to
>>> figure out if "telephone-event" is in the SDP.  If not, then do
>>> session:execute("start_dtmf", "")
>>>
>>> Chris
>>>
>>>
>>> On Tue, Mar 17, 2015 at 3:05 PM, Phillip Jones <pjintheusa at gmail.com>
>>> wrote:
>>>
>>>> Hi All,
>>>>
>>>> The following dial plan detects RFC 2833 and if it is not present, runs
>>>> the "start_dtmf" application.
>>>>
>>>> <condition field="${switch_r_sdp}"
>>>> expression="a=rtpmap:(\d+)\stelephone-event/8000" break="never">
>>>> <action application="set" data="rtp_payload_number=$1"/>
>>>> <anti-action application="start_dtmf"/>
>>>> </condition>
>>>>
>>>> I need to do this on the outbound leg - using Lua. Can anyone point me
>>>> to some code example that would achieve this or get me started?
>>>>
>>>> Any help appreciated.
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> Phil
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/297692bf/attachment-0001.html 

From grcamauer at gmail.com  Wed Mar 18 00:20:14 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 17 Mar 2015 18:20:14 -0300
Subject: [Freeswitch-users] ASR and PocketSphinx
Message-ID: <CAPNTnnoSquK6DdZ3ej_9uCrgoPyO4eu3n470W5b3W1KeVRTNsw@mail.gmail.com>

I am looking to experiment with some simple ASR. Small vocabulary (yes, No,
one, two.. zero, repeat).  I found mod_pocketsphinx in the old WIKI.  I am
wondering if this module is being maintained or if it has fallen into
disuse.  If the latter, can anyone share their experiences with other FREE
ASRs?  Are there any free still out there or must we go the commercial
route?  By the way, I am looking for both English and Spanish ASR.

Thanks,

-- 
Guillermo Ruiz Camauer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150317/899f30eb/attachment.html 

From ssinyagin at gmail.com  Wed Mar 18 00:35:15 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 17 Mar 2015 22:35:15 +0100
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
	<CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>
	<CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>
Message-ID: <CACOYK=iDD35byh3xM+LSZeniX-=Tjyd_7E2i41iDZEQSvsQLXw@mail.gmail.com>

http://files.freeswitch.org/repo/deb-master/debian/
seems to be quite fresh. How often is it updated?

On Tue, Mar 17, 2015 at 5:01 PM, Joseph Dickson <jdickson at evolvetsi.com> wrote:
> Understood.. I'll spend some time this week getting things compiled from
> source instead of using the binary packages, and see if behavior is the
> same.
>
>
> On Tue, Mar 17, 2015 at 11:56 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Start by always testing on master.  Today's master is tomorrows release.
>> Todays release is yesterdays problem that won't be solved.
>> secondly, report issues to JIRA (after reproducing on master)
>>
>>
>> On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson <jdickson at evolvetsi.com>
>> wrote:
>>>
>>> Yes, I can definitely see where master-master mode in MySQL can put
>>> you in a pickle consistency wise..  I just wish Postgres was easier
>>> with regard to fail-back.. failover isn't a problem, but getting
>>> things to fail back requires rsync of data files, etc..
>>>
>>> Before I set that up, I'd like to see sofia recover work in my
>>> environment..  Currently it appears that the correct information is
>>> being retrieved from the database, but that FS is hanging up the A leg
>>> with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
>>> behind that hangup.
>>>
>>> Is there better logging that I can grab that might be more helpful?
>>>
>>> Thanks!
>>>
>>> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>> > I has used MySQL in master-master cluster and find it is unstable
>>> > configuration. Many troubles with cluster consistence.
>>> > I recommend use configuration master-slave with tools DB
>>> > synchronization.
>>> >
>>> > Also migration to PostgreSQL cluster will be smart decision.
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 gmail.com  Wed Mar 18 00:44:10 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 17 Mar 2015 22:44:10 +0100
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CACOYK=iDD35byh3xM+LSZeniX-=Tjyd_7E2i41iDZEQSvsQLXw@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
	<CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>
	<CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>
	<CACOYK=iDD35byh3xM+LSZeniX-=Tjyd_7E2i41iDZEQSvsQLXw@mail.gmail.com>
Message-ID: <CACOYK=h8h=L6r3Y2UNSmsLNtT=U02eb43Fehe=aGWMJMKfpgqQ@mail.gmail.com>

duh, deb-master repo is about a month old.

On Tue, Mar 17, 2015 at 10:35 PM, Stanislav Sinyagin
<ssinyagin at gmail.com> wrote:
> http://files.freeswitch.org/repo/deb-master/debian/
> seems to be quite fresh. How often is it updated?
>
> On Tue, Mar 17, 2015 at 5:01 PM, Joseph Dickson <jdickson at evolvetsi.com> wrote:
>> Understood.. I'll spend some time this week getting things compiled from
>> source instead of using the binary packages, and see if behavior is the
>> same.
>>
>>
>> On Tue, Mar 17, 2015 at 11:56 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> Start by always testing on master.  Today's master is tomorrows release.
>>> Todays release is yesterdays problem that won't be solved.
>>> secondly, report issues to JIRA (after reproducing on master)
>>>
>>>
>>> On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson <jdickson at evolvetsi.com>
>>> wrote:
>>>>
>>>> Yes, I can definitely see where master-master mode in MySQL can put
>>>> you in a pickle consistency wise..  I just wish Postgres was easier
>>>> with regard to fail-back.. failover isn't a problem, but getting
>>>> things to fail back requires rsync of data files, etc..
>>>>
>>>> Before I set that up, I'd like to see sofia recover work in my
>>>> environment..  Currently it appears that the correct information is
>>>> being retrieved from the database, but that FS is hanging up the A leg
>>>> with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
>>>> behind that hangup.
>>>>
>>>> Is there better logging that I can grab that might be more helpful?
>>>>
>>>> Thanks!
>>>>
>>>> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>> > I has used MySQL in master-master cluster and find it is unstable
>>>> > configuration. Many troubles with cluster consistence.
>>>> > I recommend use configuration master-slave with tools DB
>>>> > synchronization.
>>>> >
>>>> > Also migration to PostgreSQL cluster will be smart decision.
>>>> >
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>
>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>> http://twitter.com/FreeSWITCH
>>> ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+
>>>
>>> ClueCon Weekly Development Call
>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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 Mar 18 08:20:06 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 18 Mar 2015 00:20:06 -0500
Subject: [Freeswitch-users] Channel
	Variable:	variable_pre_transfer_caller_id_number
In-Reply-To: <009e01d05f66$9d4d82f0$d7e888d0$@botecomm.com>
References: <CAP6oyxHHt9Dn3jM=5j0hqQ-mG9fBbrnG3Q14qEsWBFzvdcu4wg@mail.gmail.com>
	<009e01d05f66$9d4d82f0$d7e888d0$@botecomm.com>
Message-ID: <55090B06.8020504@freeswitch.org>

To clarify, variable_ might be prepended in *some* of the CDR loggers, 
but it is in no way a standard function of our loggers.  Events from the 
event socket will prepend "variable_" to channel variables to indicate 
that these are channel variables that are being listed within the event 
itself.

-- 
Yossi Neiman

On 03/15/2015 04:25 PM, Bote Man wrote:
>
> I believe the ?variable_? prefix only applies to CDRs and logs. When 
> processing them in the dialplan omit the ?variable_? prefix and just use
>
> pre_transfer_caller_id_name
>
> Bote
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Paul Atreides
> *Sent:* Sunday, 15 March, 2015 11:43
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Channel Variable: 
> variable_pre_transfer_caller_id_number
>
> Hi,
>
> <action application="info"/>
> when I call the info app in my dial plan then I can see the following 
> vars in my log
>
> variable_pre_transfer_caller_id_name: []
> variable_pre_transfer_caller_id_number: [18]
>
> But when I try to access one single channel variable its always empty. 
> Why?
> ${variable_pre_transfer_caller_id_number}
>
> Thank you for your help.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/8c4e0f2a/attachment.html 

From ali.jibran44 at gmail.com  Wed Mar 18 13:51:26 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 15:51:26 +0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
Message-ID: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>

Hi all.
I make a call to FS callcenter using webRTC. Now I would like to add some
additional data to the header.
That I did by using extraheaders. Now all that is good. I can see the
custom header in the sip trace on freeswitch.

For e.g in sip.js I add:
var options = {
  extraHeaders: [ 'a: foo' ]
};


on FS i get:
 INVITE sip:900 at anonymous.invalid SIP/2.0
   Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
   Max-Forwards: 70
   To: <sip:900 at anonymous.invalid>
   From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
   Call-ID: l5pieavpjguoset7ugvd
   CSeq: 1458 INVITE
  * a: foo*
   Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
   Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
   Content-Type: application/sdp
   Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
   Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
   Content-Type: application/sdp
   Supported: outbound
   User-Agent: SIP.js/0.5.0
   Content-Length: 3636

How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
work. Help please?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/fe3ecd57/attachment-0001.html 

From alhakeem at gmail.com  Wed Mar 18 14:41:37 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Wed, 18 Mar 2015 11:41:37 -0000
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
In-Reply-To: <A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
	<A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>

Hello,
The lost or delayed ACK are periodic. Sometime it disappear within minutes, but
of course after the call has been dropped.
IS there any way to send fake ACK packet to FS via ESL  before CONNECT ?
Cheers

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, March 17, 2015 5:01 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received

No, and there shouldn't/can't be.  If you try to continue a call that has nat
issues keeping responses from going through, there will be no way to end the
call.

> On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>
> Hello,
>
> Is there any way to override FS behaviour when no ACK is received by FS from
> registered clients?
> I'm hoping there might be a way for FS to continue processing rather than end
> the call.
>
> Cheers,
> Abdul Hakeem


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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




From s.safarov at gmail.com  Wed Mar 18 15:26:47 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 18 Mar 2015 12:26:47 +0000
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
	<A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>
Message-ID: <CAHtxdDe5CfjveLWCY9Y4sLHJhaMfQrWP3i3=1sogds79+T=1fA@mail.gmail.com>

I will recommend you switch to use TCP or TLS transport. If packet is lost
it auto retransmitted by TCP stack protocol.
It help me make call through nat devices with bad implementation of nat.

On Wed, Mar 18, 2015, 13:42 Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
> The lost or delayed ACK are periodic. Sometime it disappear within
> minutes, but
> of course after the call has been dropped.
> IS there any way to send fake ACK packet to FS via ESL  before CONNECT ?
> Cheers
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael
> Jerris
> Sent: Tuesday, March 17, 2015 5:01 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received
>
> No, and there shouldn't/can't be.  If you try to continue a call that has
> nat
> issues keeping responses from going through, there will be no way to end
> the
> call.
>
> > On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> >
> > Hello,
> >
> > Is there any way to override FS behaviour when no ACK is received by FS
> from
> > registered clients?
> > I'm hoping there might be a way for FS to continue processing rather
> than end
> > the call.
> >
> > Cheers,
> > Abdul Hakeem
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/08570e2e/attachment.html 

From mike at jerris.com  Wed Mar 18 15:27:37 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Mar 2015 08:27:37 -0400
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
Message-ID: <F5A8CA3B-C54F-4520-8CF4-DC9E7C93F25A@jerris.com>

That would have worked if the variable you put into the message was X-a instead of a

> On Mar 18, 2015, at 6:51 AM, Ali Jibran <ali.jibran44 at gmail.com> wrote:
> 
> Hi all.
> I make a call to FS callcenter using webRTC. Now I would like to add some additional data to the header. 
> That I did by using extraheaders. Now all that is good. I can see the custom header in the sip trace on freeswitch.
> 
> For e.g in sip.js I add:
> var options = {
>   extraHeaders: [ 'a: foo' ]
> };
> 
> 
> on FS i get:
>  INVITE sip:900 at anonymous.invalid SIP/2.0
>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>    Max-Forwards: 70
>    To: <sip:900 at anonymous.invalid>
>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>    Call-ID: l5pieavpjguoset7ugvd
>    CSeq: 1458 INVITE
>    a: foo
>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>    Content-Type: application/sdp
>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>    Content-Type: application/sdp
>    Supported: outbound
>    User-Agent: SIP.js/0.5.0
>    Content-Length: 3636
> 
> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't work. Help please?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/d3131ac6/attachment.html 

From ben at langfeld.co.uk  Wed Mar 18 15:28:37 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 18 Mar 2015 09:28:37 -0300
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
Message-ID: <CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>

Prefix the header with X-. Anything else is invalid.

On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> Hi all.
> I make a call to FS callcenter using webRTC. Now I would like to add some
> additional data to the header.
> That I did by using extraheaders. Now all that is good. I can see the
> custom header in the sip trace on freeswitch.
>
> For e.g in sip.js I add:
> var options = {
>   extraHeaders: [ 'a: foo' ]
> };
>
>
> on FS i get:
>  INVITE sip:900 at anonymous.invalid SIP/2.0
>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>    Max-Forwards: 70
>    To: <sip:900 at anonymous.invalid>
>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>    Call-ID: l5pieavpjguoset7ugvd
>    CSeq: 1458 INVITE
>   * a: foo*
>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>    Content-Type: application/sdp
>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>    Content-Type: application/sdp
>    Supported: outbound
>    User-Agent: SIP.js/0.5.0
>    Content-Length: 3636
>
> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
> work. Help please?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/10bd2874/attachment.html 

From mike at jerris.com  Wed Mar 18 15:29:54 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Mar 2015 08:29:54 -0400
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
	<A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>
Message-ID: <B0200099-E5A6-42C3-BE38-429B80CDE3C8@jerris.com>

I don't understand what you are saying?  What is disappearing?  No, you never want to send a fake packet.. there is a message flow that includes ack for a reason.  You need to address whatever is causing lost packets or you can't hope to have coherent conversations over sip.

> On Mar 18, 2015, at 7:41 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello,
> The lost or delayed ACK are periodic. Sometime it disappear within minutes, but
> of course after the call has been dropped.
> IS there any way to send fake ACK packet to FS via ESL  before CONNECT ?
> Cheers
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Jerris
> Sent: Tuesday, March 17, 2015 5:01 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received
> 
> No, and there shouldn't/can't be.  If you try to continue a call that has nat
> issues keeping responses from going through, there will be no way to end the
> call.
> 
>> On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>> 
>> Hello,
>> 
>> Is there any way to override FS behaviour when no ACK is received by FS from
>> registered clients?
>> I'm hoping there might be a way for FS to continue processing rather than end
>> the call.
>> 
>> Cheers,
>> Abdul Hakeem
> 



From ali.jibran44 at gmail.com  Wed Mar 18 15:47:39 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 17:47:39 +0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
Message-ID: <CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>

Ok yeah :) thank you that worked.

One more thing. I can access the Header now in my dialplan. But I pass the
call into call Center.
Now how do I pass the header into call Center so that it pops up to which
ever agent picked it up.

Like
Web->fs->call center->agent

I can get the header to Freeswitch but it's not showing after the call is
bridged to agent.
I tried sip_copy_custom_headers=true but it didn't work

On Wednesday, March 18, 2015, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Prefix the header with X-. Anything else is invalid.
>
> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com
> <javascript:_e(%7B%7D,'cvml','ali.jibran44 at gmail.com');>> wrote:
>
>> Hi all.
>> I make a call to FS callcenter using webRTC. Now I would like to add some
>> additional data to the header.
>> That I did by using extraheaders. Now all that is good. I can see the
>> custom header in the sip trace on freeswitch.
>>
>> For e.g in sip.js I add:
>> var options = {
>>   extraHeaders: [ 'a: foo' ]
>> };
>>
>>
>> on FS i get:
>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>    Max-Forwards: 70
>>    To: <sip:900 at anonymous.invalid>
>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>    Call-ID: l5pieavpjguoset7ugvd
>>    CSeq: 1458 INVITE
>>   * a: foo*
>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>    Content-Type: application/sdp
>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>    Content-Type: application/sdp
>>    Supported: outbound
>>    User-Agent: SIP.js/0.5.0
>>    Content-Length: 3636
>>
>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
>> work. Help please?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/21c02882/attachment.html 

From brian at freeswitch.org  Wed Mar 18 16:15:49 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 08:15:49 -0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
Message-ID: <CAEJMVkBxdeGyVrv6kbxw1ticNhqQN77mp7H1ReotsB3zuin6uw@mail.gmail.com>

Sadly thats no longer true, Crazy right?

On Wed, Mar 18, 2015 at 7:28 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Prefix the header with X-. Anything else is invalid.
>
> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>
>> Hi all.
>> I make a call to FS callcenter using webRTC. Now I would like to add some
>> additional data to the header.
>> That I did by using extraheaders. Now all that is good. I can see the
>> custom header in the sip trace on freeswitch.
>>
>> For e.g in sip.js I add:
>> var options = {
>>   extraHeaders: [ 'a: foo' ]
>> };
>>
>>
>> on FS i get:
>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>    Max-Forwards: 70
>>    To: <sip:900 at anonymous.invalid>
>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>    Call-ID: l5pieavpjguoset7ugvd
>>    CSeq: 1458 INVITE
>>   * a: foo*
>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>    Content-Type: application/sdp
>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>    Content-Type: application/sdp
>>    Supported: outbound
>>    User-Agent: SIP.js/0.5.0
>>    Content-Length: 3636
>>
>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
>> work. Help please?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/dddeaf4c/attachment.html 

From brian at freeswitch.org  Wed Mar 18 16:16:41 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 08:16:41 -0500
Subject: [Freeswitch-users] ASR and PocketSphinx
In-Reply-To: <CAPNTnnoSquK6DdZ3ej_9uCrgoPyO4eu3n470W5b3W1KeVRTNsw@mail.gmail.com>
References: <CAPNTnnoSquK6DdZ3ej_9uCrgoPyO4eu3n470W5b3W1KeVRTNsw@mail.gmail.com>
Message-ID: <CAEJMVkDA9ah3jWE7PfuVDb+CTE9Npf2=F8CHg4sA7CPWM4nQng@mail.gmail.com>

It should work per the examples on the wiki.  But sadly there are no
acoustical models for any other language but English.

On Tue, Mar 17, 2015 at 4:20 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> I am looking to experiment with some simple ASR. Small vocabulary (yes,
> No, one, two.. zero, repeat).  I found mod_pocketsphinx in the old WIKI.  I
> am wondering if this module is being maintained or if it has fallen into
> disuse.  If the latter, can anyone share their experiences with other FREE
> ASRs?  Are there any free still out there or must we go the commercial
> route?  By the way, I am looking for both English and Spanish ASR.
>
> Thanks,
>
> --
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/959d317c/attachment.html 

From denis at ringme.ru  Wed Mar 18 16:49:03 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Wed, 18 Mar 2015 16:49:03 +0300
Subject: [Freeswitch-users] bug with events?
Message-ID: <5509824F.6090608@ringme.ru>

Hi.

Possible, we found a bug.
If we call to freeswitch with redirect to mobile phone and hangup before 
state "ringing" - ESL events show leg B without Call-ID (i.e. 
CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...)

example (json view)
https://pastebin.freeswitch.org/24022


From mike at jerris.com  Wed Mar 18 17:04:55 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Mar 2015 10:04:55 -0400
Subject: [Freeswitch-users] bug with events?
In-Reply-To: <5509824F.6090608@ringme.ru>
References: <5509824F.6090608@ringme.ru>
Message-ID: <E501824C-F192-4335-85E0-C240679F433A@jerris.com>

This is probably expected, although with so little information as provided I can't say for sure.  You might have luck finding it from the other leg's cdr, we might not even have that information yet.

> On Mar 18, 2015, at 9:49 AM, ????? <denis at ringme.ru> wrote:
> 
> Hi.
> 
> Possible, we found a bug.
> If we call to freeswitch with redirect to mobile phone and hangup before 
> state "ringing" - ESL events show leg B without Call-ID (i.e. 
> CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...)
> 
> example (json view)
> https://pastebin.freeswitch.org/24022
> 



From alhakeem at gmail.com  Wed Mar 18 17:05:12 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Wed, 18 Mar 2015 14:05:12 -0000
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
In-Reply-To: <B0200099-E5A6-42C3-BE38-429B80CDE3C8@jerris.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>	<A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>
	<B0200099-E5A6-42C3-BE38-429B80CDE3C8@jerris.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAJmF+vZxNElOqOj5UNc+uqkBAAAAAA==@gmail.com>

Hello,
What I meant by disappearing is: an inbound call might fail now & 5 minutes
later NAT is fine.
There's no firewall so, I think it might be the OPTIONS pings.
Anyway, I will change the router just to see any improvement.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Wednesday, March 18, 2015 12:30 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received

I don't understand what you are saying?  What is disappearing?  No, you never
want to send a fake packet.. there is a message flow that includes ack for a
reason.  You need to address whatever is causing lost packets or you can't hope
to have coherent conversations over sip.

> On Mar 18, 2015, at 7:41 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>
> Hello,
> The lost or delayed ACK are periodic. Sometime it disappear within minutes,
but
> of course after the call has been dropped.
> IS there any way to send fake ACK packet to FS via ESL  before CONNECT ?
> Cheers
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Jerris
> Sent: Tuesday, March 17, 2015 5:01 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received
>
> No, and there shouldn't/can't be.  If you try to continue a call that has nat
> issues keeping responses from going through, there will be no way to end the
> call.
>
>> On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>>
>> Hello,
>>
>> Is there any way to override FS behaviour when no ACK is received by FS from
>> registered clients?
>> I'm hoping there might be a way for FS to continue processing rather than end
>> the call.
>>
>> Cheers,
>> Abdul Hakeem
>


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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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




From italorossib at gmail.com  Wed Mar 18 17:25:01 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 18 Mar 2015 11:25:01 -0300
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
Message-ID: <CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>

using mod_callcenter? try setting cc_export_vars

On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> Ok yeah :) thank you that worked.
>
> One more thing. I can access the Header now in my dialplan. But I pass the
> call into call Center.
> Now how do I pass the header into call Center so that it pops up to which
> ever agent picked it up.
>
> Like
> Web->fs->call center->agent
>
> I can get the header to Freeswitch but it's not showing after the call is
> bridged to agent.
> I tried sip_copy_custom_headers=true but it didn't work
>
> On Wednesday, March 18, 2015, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Prefix the header with X-. Anything else is invalid.
>>
>> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>>
>>> Hi all.
>>> I make a call to FS callcenter using webRTC. Now I would like to add
>>> some additional data to the header.
>>> That I did by using extraheaders. Now all that is good. I can see the
>>> custom header in the sip trace on freeswitch.
>>>
>>> For e.g in sip.js I add:
>>> var options = {
>>>   extraHeaders: [ 'a: foo' ]
>>> };
>>>
>>>
>>> on FS i get:
>>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>>    Max-Forwards: 70
>>>    To: <sip:900 at anonymous.invalid>
>>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>>    Call-ID: l5pieavpjguoset7ugvd
>>>    CSeq: 1458 INVITE
>>>   * a: foo*
>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>    Content-Type: application/sdp
>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>    Content-Type: application/sdp
>>>    Supported: outbound
>>>    User-Agent: SIP.js/0.5.0
>>>    Content-Length: 3636
>>>
>>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
>>> work. Help please?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/2087106e/attachment.html 

From mike at jerris.com  Wed Mar 18 17:52:17 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Mar 2015 10:52:17 -0400
Subject: [Freeswitch-users] FS default behaiour when no ACK is received
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAJmF+vZxNElOqOj5UNc+uqkBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAFR118RNspVDndpYL5GnFkYBAAAAAA==@gmail.com>
	<A5138C34-63E2-464E-81B3-AF72BE5814D0@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEAo7hJTMDRJoBZRKCgAcqMBAAAAAA==@gmail.com>
	<B0200099-E5A6-42C3-BE38-429B80CDE3C8@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAJmF+vZxNElOqOj5UNc+uqkBAAAAAA==@gmail.com>
Message-ID: <2A48C05D-AD18-452D-B336-FFE5C02E1A8A@jerris.com>

The answer is to fix your router issues or nat configuration so you don't have these issues in the first place.

> On Mar 18, 2015, at 10:05 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello,
> What I meant by disappearing is: an inbound call might fail now & 5 minutes
> later NAT is fine.
> There's no firewall so, I think it might be the OPTIONS pings.
> Anyway, I will change the router just to see any improvement.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Jerris
> Sent: Wednesday, March 18, 2015 12:30 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received
> 
> I don't understand what you are saying?  What is disappearing?  No, you never
> want to send a fake packet.. there is a message flow that includes ack for a
> reason.  You need to address whatever is causing lost packets or you can't hope
> to have coherent conversations over sip.
> 
>> On Mar 18, 2015, at 7:41 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>> 
>> Hello,
>> The lost or delayed ACK are periodic. Sometime it disappear within minutes,
> but
>> of course after the call has been dropped.
>> IS there any way to send fake ACK packet to FS via ESL  before CONNECT ?
>> Cheers
>> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
>> Jerris
>> Sent: Tuesday, March 17, 2015 5:01 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] FS default behaiour when no ACK is received
>> 
>> No, and there shouldn't/can't be.  If you try to continue a call that has nat
>> issues keeping responses from going through, there will be no way to end the
>> call.
>> 
>>> On Mar 17, 2015, at 12:15 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>>> 
>>> Hello,
>>> 
>>> Is there any way to override FS behaviour when no ACK is received by FS from
>>> registered clients?
>>> I'm hoping there might be a way for FS to continue processing rather than end
>>> the call.
>>> 
>>> Cheers,
>>> Abdul Hakeem
>> 
> 



From ali.jibran44 at gmail.com  Wed Mar 18 18:10:26 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 20:10:26 +0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
	<CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>
Message-ID: <CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>

Didn't work :/

So there's no way I can export data to mod_callcenter?

On Wednesday, March 18, 2015, ?talo Rossi <italorossib at gmail.com> wrote:

> using mod_callcenter? try setting cc_export_vars
>
> On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44 at gmail.com
> <javascript:_e(%7B%7D,'cvml','ali.jibran44 at gmail.com');>> wrote:
>
>> Ok yeah :) thank you that worked.
>>
>> One more thing. I can access the Header now in my dialplan. But I pass
>> the call into call Center.
>> Now how do I pass the header into call Center so that it pops up to which
>> ever agent picked it up.
>>
>> Like
>> Web->fs->call center->agent
>>
>> I can get the header to Freeswitch but it's not showing after the call is
>> bridged to agent.
>> I tried sip_copy_custom_headers=true but it didn't work
>>
>> On Wednesday, March 18, 2015, Ben Langfeld <ben at langfeld.co.uk
>> <javascript:_e(%7B%7D,'cvml','ben at langfeld.co.uk');>> wrote:
>>
>>> Prefix the header with X-. Anything else is invalid.
>>>
>>> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>>>
>>>> Hi all.
>>>> I make a call to FS callcenter using webRTC. Now I would like to add
>>>> some additional data to the header.
>>>> That I did by using extraheaders. Now all that is good. I can see the
>>>> custom header in the sip trace on freeswitch.
>>>>
>>>> For e.g in sip.js I add:
>>>> var options = {
>>>>   extraHeaders: [ 'a: foo' ]
>>>> };
>>>>
>>>>
>>>> on FS i get:
>>>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>>>    Max-Forwards: 70
>>>>    To: <sip:900 at anonymous.invalid>
>>>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>>>    Call-ID: l5pieavpjguoset7ugvd
>>>>    CSeq: 1458 INVITE
>>>>   * a: foo*
>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>    Content-Type: application/sdp
>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>    Content-Type: application/sdp
>>>>    Supported: outbound
>>>>    User-Agent: SIP.js/0.5.0
>>>>    Content-Length: 3636
>>>>
>>>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it didnt't
>>>> work. Help please?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/113ba159/attachment-0001.html 

From italorossib at gmail.com  Wed Mar 18 19:39:15 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 18 Mar 2015 13:39:15 -0300
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
	<CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>
	<CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>
Message-ID: <CAJR7f2Hm07Hw=MQ4W5n_ddHsZMP74NTcTsg2RduYXbomr=remw@mail.gmail.com>

You can set variables in your agent contact:

<agent name="1001 at default" type="callback"
contact="[origination_caller_id_name='Queue
Caller',call_timeout=10]user/1001 at default" status="Available"
max-no-answer="3" wrap-up-time="10" reject-delay-time="10"
busy-delay-time="60" />

And probably you can set variables to be expanded, in mod_callcenter.c I see

dialstr = switch_channel_expand_variables(member_channel,
h->originate_string);

h->originate_string is the contact of the agent

So it'll probably work.

NOT TESTED!

On Wed, Mar 18, 2015 at 12:10 PM, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> Didn't work :/
>
> So there's no way I can export data to mod_callcenter?
>
>
> On Wednesday, March 18, 2015, ?talo Rossi <italorossib at gmail.com> wrote:
>
>> using mod_callcenter? try setting cc_export_vars
>>
>> On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44 at gmail.com>
>> wrote:
>>
>>> Ok yeah :) thank you that worked.
>>>
>>> One more thing. I can access the Header now in my dialplan. But I pass
>>> the call into call Center.
>>> Now how do I pass the header into call Center so that it pops up to
>>> which ever agent picked it up.
>>>
>>> Like
>>> Web->fs->call center->agent
>>>
>>> I can get the header to Freeswitch but it's not showing after the call
>>> is bridged to agent.
>>> I tried sip_copy_custom_headers=true but it didn't work
>>>
>>> On Wednesday, March 18, 2015, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>>
>>>> Prefix the header with X-. Anything else is invalid.
>>>>
>>>> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>>>>
>>>>> Hi all.
>>>>> I make a call to FS callcenter using webRTC. Now I would like to add
>>>>> some additional data to the header.
>>>>> That I did by using extraheaders. Now all that is good. I can see the
>>>>> custom header in the sip trace on freeswitch.
>>>>>
>>>>> For e.g in sip.js I add:
>>>>> var options = {
>>>>>   extraHeaders: [ 'a: foo' ]
>>>>> };
>>>>>
>>>>>
>>>>> on FS i get:
>>>>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>>>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>>>>    Max-Forwards: 70
>>>>>    To: <sip:900 at anonymous.invalid>
>>>>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>>>>    Call-ID: l5pieavpjguoset7ugvd
>>>>>    CSeq: 1458 INVITE
>>>>>   * a: foo*
>>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>>    Content-Type: application/sdp
>>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>>    Content-Type: application/sdp
>>>>>    Supported: outbound
>>>>>    User-Agent: SIP.js/0.5.0
>>>>>    Content-Length: 3636
>>>>>
>>>>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it
>>>>> didnt't work. Help please?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/94ffb3f4/attachment.html 

From ali.jibran44 at gmail.com  Wed Mar 18 19:42:50 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 21:42:50 +0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAJR7f2Hm07Hw=MQ4W5n_ddHsZMP74NTcTsg2RduYXbomr=remw@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
	<CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>
	<CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>
	<CAJR7f2Hm07Hw=MQ4W5n_ddHsZMP74NTcTsg2RduYXbomr=remw@mail.gmail.com>
Message-ID: <CAP9PqkQZ2KzjUdocEfH7hc5HvRd8_KEuQQfdUe8uE=UR7Aenbw@mail.gmail.com>

Yeah but I can't pass a variable from a leg to b can I?

On Wednesday, March 18, 2015, ?talo Rossi <italorossib at gmail.com> wrote:

> You can set variables in your agent contact:
>
> <agent name="1001 at default" type="callback"
> contact="[origination_caller_id_name='Queue
> Caller',call_timeout=10]user/1001 at default" status="Available"
> max-no-answer="3" wrap-up-time="10" reject-delay-time="10"
> busy-delay-time="60" />
>
> And probably you can set variables to be expanded, in mod_callcenter.c I
> see
>
> dialstr = switch_channel_expand_variables(member_channel,
> h->originate_string);
>
> h->originate_string is the contact of the agent
>
> So it'll probably work.
>
> NOT TESTED!
>
> On Wed, Mar 18, 2015 at 12:10 PM, Ali Jibran <ali.jibran44 at gmail.com
> <javascript:_e(%7B%7D,'cvml','ali.jibran44 at gmail.com');>> wrote:
>
>> Didn't work :/
>>
>> So there's no way I can export data to mod_callcenter?
>>
>>
>> On Wednesday, March 18, 2015, ?talo Rossi <italorossib at gmail.com
>> <javascript:_e(%7B%7D,'cvml','italorossib at gmail.com');>> wrote:
>>
>>> using mod_callcenter? try setting cc_export_vars
>>>
>>> On Wed, Mar 18, 2015 at 9:47 AM, Ali Jibran <ali.jibran44 at gmail.com>
>>> wrote:
>>>
>>>> Ok yeah :) thank you that worked.
>>>>
>>>> One more thing. I can access the Header now in my dialplan. But I pass
>>>> the call into call Center.
>>>> Now how do I pass the header into call Center so that it pops up to
>>>> which ever agent picked it up.
>>>>
>>>> Like
>>>> Web->fs->call center->agent
>>>>
>>>> I can get the header to Freeswitch but it's not showing after the call
>>>> is bridged to agent.
>>>> I tried sip_copy_custom_headers=true but it didn't work
>>>>
>>>> On Wednesday, March 18, 2015, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>>>
>>>>> Prefix the header with X-. Anything else is invalid.
>>>>>
>>>>> On 18 March 2015 at 07:51, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>>>>>
>>>>>> Hi all.
>>>>>> I make a call to FS callcenter using webRTC. Now I would like to add
>>>>>> some additional data to the header.
>>>>>> That I did by using extraheaders. Now all that is good. I can see the
>>>>>> custom header in the sip trace on freeswitch.
>>>>>>
>>>>>> For e.g in sip.js I add:
>>>>>> var options = {
>>>>>>   extraHeaders: [ 'a: foo' ]
>>>>>> };
>>>>>>
>>>>>>
>>>>>> on FS i get:
>>>>>>  INVITE sip:900 at anonymous.invalid SIP/2.0
>>>>>>    Via: SIP/2.0/WS 9jscv0q24s4m.invalid;branch=z9hG4bK1002472
>>>>>>    Max-Forwards: 70
>>>>>>    To: <sip:900 at anonymous.invalid>
>>>>>>    From: <sip:anonymous.hfsqa4 at anonymous.invalid>;tag=depmif1fo5
>>>>>>    Call-ID: l5pieavpjguoset7ugvd
>>>>>>    CSeq: 1458 INVITE
>>>>>>   * a: foo*
>>>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>>>    Content-Type: application/sdp
>>>>>>    Contact: <sip:8rgi365g at 9jscv0q24s4m.invalid;transport=ws;ob>
>>>>>>    Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
>>>>>>    Content-Type: application/sdp
>>>>>>    Supported: outbound
>>>>>>    User-Agent: SIP.js/0.5.0
>>>>>>    Content-Length: 3636
>>>>>>
>>>>>> How do I access it in the dialplan? I tried ${sip_h_X-a} but it
>>>>>> didnt't work. Help please?
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/cba0d951/attachment-0001.html 

From bote_radio at botecomm.com  Wed Mar 18 20:33:57 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 18 Mar 2015 13:33:57 -0400
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <CAP9PqkQZ2KzjUdocEfH7hc5HvRd8_KEuQQfdUe8uE=UR7Aenbw@mail.gmail.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>	<CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>	<CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>	<CAJR7f2Hm07Hw=MQ4W5n_ddHsZMP74NTcTsg2RduYXbomr=remw@mail.gmail.com>
	<CAP9PqkQZ2KzjUdocEfH7hc5HvRd8_KEuQQfdUe8uE=UR7Aenbw@mail.gmail.com>
Message-ID: <005c01d061a1$b7318850$259498f0$@botecomm.com>

The ?export? app can.

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Jibran
Sent: Wednesday, 18 March, 2015 12:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Custom Headers with Sip.js

 

Yeah but I can't pass a variable from a leg to b can I? 




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/f5bf40bb/attachment.html 

From ali.jibran44 at gmail.com  Wed Mar 18 20:36:22 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 22:36:22 +0500
Subject: [Freeswitch-users] Custom Headers with Sip.js
In-Reply-To: <005c01d061a1$b7318850$259498f0$@botecomm.com>
References: <CAP9PqkQS27RSFpM7trS5kc1F0Qd_FXhZZbSksAGdeMQhC6cH7Q@mail.gmail.com>
	<CAAyX+KEiAPErESyfM8WvGAaM_95-tptTGCu72LsnoKNpAnRkCw@mail.gmail.com>
	<CAP9PqkQVZMOs0adnZ2a5TbqAQRbBF_vQCgbNBQiv1tzHUahJQg@mail.gmail.com>
	<CAJR7f2Ggrbkdz8qYeHS+G_BVHgDNbgHE8dqUex9F3fvoHzzOCQ@mail.gmail.com>
	<CAP9PqkQcRBhG-yJ9An9GPsJ8OMkv99sdMBYFj=+6c=bUQ9KKTg@mail.gmail.com>
	<CAJR7f2Hm07Hw=MQ4W5n_ddHsZMP74NTcTsg2RduYXbomr=remw@mail.gmail.com>
	<CAP9PqkQZ2KzjUdocEfH7hc5HvRd8_KEuQQfdUe8uE=UR7Aenbw@mail.gmail.com>
	<005c01d061a1$b7318850$259498f0$@botecomm.com>
Message-ID: <CAP9PqkSxxipt3fUZ-8RhgLTkeKwnY1TWn0Fx8XdqDuDa6Fx6Eg@mail.gmail.com>

Can you explain in detail
Please?
I need to export a header from A leg to B leg....but in the call Center
app. Like the variable should be exported to whoever picks up the call.

On Wednesday, March 18, 2015, Bote Man <bote_radio at botecomm.com> wrote:

> The ?export? app can.
>
>
>
> Bote
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Ali Jibran
> *Sent:* Wednesday, 18 March, 2015 12:43
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Custom Headers with Sip.js
>
>
>
> Yeah but I can't pass a variable from a leg to b can I?
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/12b5708f/attachment.html 

From orn at arnarson.net  Wed Mar 18 21:26:36 2015
From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=)
Date: Wed, 18 Mar 2015 18:26:36 +0000
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked calls
Message-ID: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>

Hello,

Not sure whether this belong in the users list or the dev list, but when in
doubt; start with users :-)

I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
Asterisk on the other, and am seeing some strange behavior when calls are
being forked on the Asterisk.

Call setup is like this:
1. FreeSWITCH receives INVITE from Kamailio
2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH (each
with a unique call-id)
4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new Call-ID
from step 2.

This is causing problems with one of the MGWs behind Kamailio, which is
seeing multiple INVITEs to different destinations with the same Call-ID.

So, firstly, why is FreeSWITCH reusing call-ids?

Secondly, how is it matching up the calls? I can't find anything common in
the INVITEs, other than the source number and obviously that the IP sent to
and received from is the same.

I'm not sure if this is intended behavior or not, but is there a way to
have FreeSWITCH not do that?

Regards,
?rn

P.S. Here is the sequence of INVITEs. I also have the console log (for a
different call) if needed.

*INVITE sent to FreeSWITCH by Kamailio:*
INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
Record-Route: <sip:172.25.200.101;lr=on>
Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
Via: SIP/2.0/UDP 172.25.200.121:5080;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
Max-Forwards: 16
From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
To: <sip:5344446 at 172.25.200.101>
Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
CSeq: 73014324 INVITE
Contact: <sip:mod_sofia at 172.25.200.121:5080>
User-Agent: FreeSWITCH-mod_sofia/1.2.7
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 229
X-FS-Support: update_display,send_info
Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
s=FreeSWITCH
c=IN IP4 172.25.200.121
t=0 0
m=audio 19026 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20


*INVITE sent to Asterisk by FreeSWITCH:*
INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
Max-Forwards: 15
From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
To: <sip:5344446 at 172.26.0.62:5060>
Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
CSeq: 73014353 INVITE
Contact: <sip:mod_sofia at 10.11.12.13:5060>
User-Agent: FreeSWITCH-mod_sofia/1.2.12
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: 223
X-FS-Support: update_display,send_info
Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
s=FreeSWITCH
c=IN IP4 10.11.12.13
t=0 0
m=audio 23230 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

*First INVITE sent to FreeSWITCH by Asterisk (forked call):*
INVITE sip:7712552 at 10.11.12.13 SIP/2.0
Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
Max-Forwards: 70
From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
To: <sip:7712552 at 10.11.12.13>
Contact: <sip:4151502 at 172.26.0.62:5060>
Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15-cert2
Date: Wed, 18 Mar 2015 17:47:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Diversion: <sip:5344446 at 172.26.0.62>
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 693576967 693576967 IN IP4 172.26.0.62
s=Asterisk PBX 1.8.15-cert2
c=IN IP4 172.26.0.62
t=0 0
m=audio 30440 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Second INVITE sent to FreeSWITCH by Asterisk (forked call):
INVITE sip:6595454 at 10.11.12.13 SIP/2.0
Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
Max-Forwards: 70
From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
To: <sip:6595454 at 10.11.12.13>
Contact: <sip:4151502 at 172.26.0.62:5060>
Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15-cert2
Date: Wed, 18 Mar 2015 17:47:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Diversion: <sip:5344446 at 172.26.0.62>
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 89056081 89056081 IN IP4 172.26.0.62
s=Asterisk PBX 1.8.15-cert2
c=IN IP4 172.26.0.62
t=0 0
m=audio 30708 RTP/AVP 8 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

*First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
INVITE sip:7712552 at 172.25.200.101 SIP/2.0
Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
Max-Forwards: 69
From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
To: <sip:7712552 at 172.25.200.101>
Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
CSeq: 73014353 INVITE
Contact: <sip:mod_sofia at 172.25.200.111:5080>
User-Agent: FreeSWITCH-mod_sofia/1.2.12
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 209
Diversion: <sip:5344446 at 172.25.200.101>
X-FS-Support: update_display,send_info

v=0
o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
s=FreeSWITCH
c=IN IP4 172.25.200.111
t=0 0
m=audio 19804 RTP/AVP 8 0 9 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
INVITE sip:6595454 at 172.25.200.101 SIP/2.0
Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
Max-Forwards: 69
From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
To: <sip:6595454 at 172.25.200.101>
Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
CSeq: 73014353 INVITE
Contact: <sip:mod_sofia at 172.25.200.111:5080>
User-Agent: FreeSWITCH-mod_sofia/1.2.12
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 209
Diversion: <sip:5344446 at 172.25.200.101>
X-FS-Support: update_display,send_info

v=0
o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
s=FreeSWITCH
c=IN IP4 172.25.200.111
t=0 0
m=audio 31376 RTP/AVP 8 0 9 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/29f08ed3/attachment-0001.html 

From peter at hartmanncomputer.com  Wed Mar 18 21:30:27 2015
From: peter at hartmanncomputer.com (Peter Hartmann)
Date: Wed, 18 Mar 2015 14:30:27 -0400
Subject: [Freeswitch-users] g729 installer 404
In-Reply-To: <CALXC2kdLE2dqeQ92F3X7suKcXaA1gz3j=LZPfxO5tMReUQkXnQ@mail.gmail.com>
References: <CALXC2kdLE2dqeQ92F3X7suKcXaA1gz3j=LZPfxO5tMReUQkXnQ@mail.gmail.com>
Message-ID: <CALXC2kceYgp+307Nq5Kc-6qj+p4UhTb_kfp47ZF7+U35QSpkag@mail.gmail.com>

Hey,
There seems to be a hosting problem with g729 installer.  It's located at:

http://files.freeswitch.org/g729/fs-201501231218-installer

However, when one attempts to build FS from source
(1.5.15b+git~20150317T232708Z~42961b7f2e~64bit and also the 1.4
version)  the scripts are attempting to retrieve it from:

http://files.freeswitch.org/g729/fsg729-201501231218-installer


resulting in a 404.  Note the different filename prefix.

-----

cd /usr/src/freeswitch/libs && /bin/bash
/usr/src/freeswitch/build/getg729.sh fsg729-201501231218-installer
--2015-03-18 12:31:47--
http://files.freeswitch.org/g729/fsg729-201501231218-installer
Resolving files.freeswitch.org (files.freeswitch.org)...
209.105.235.7, 2607:f348:1021::7
Connecting to files.freeswitch.org
(files.freeswitch.org)|209.105.235.7|:80... connected.
HTTP request sent, awaiting response... 404 Not Found
2015-03-18 12:31:47 ERROR 404: Not Found.
-------



I'd file a JIRA but I don't think there's anything wrong with the
code.    'fsg729' seems to be a more descriptive filename prefix as
compared to 'fs'.    Sed got me through the build, but it seems
something that should be fixed, no?  And if  'fs' is the preferred
filename prefix I'd be happy to file a report.


Thanks much,



Peter Hartmann
Hartmann Computer Consulting
http://blog.hartmanncomputer.com
(212)203-8870

If I can't explain it to you in plain language, that means I don't
understand it.


From grcamauer at gmail.com  Wed Mar 18 21:44:08 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 18 Mar 2015 15:44:08 -0300
Subject: [Freeswitch-users] ASR and PocketSphinx
In-Reply-To: <CAEJMVkDA9ah3jWE7PfuVDb+CTE9Npf2=F8CHg4sA7CPWM4nQng@mail.gmail.com>
References: <CAPNTnnoSquK6DdZ3ej_9uCrgoPyO4eu3n470W5b3W1KeVRTNsw@mail.gmail.com>
	<CAEJMVkDA9ah3jWE7PfuVDb+CTE9Npf2=F8CHg4sA7CPWM4nQng@mail.gmail.com>
Message-ID: <CAPNTnnoqe8ONnxoNomHkvvOTkLGbJaRvQ+S6wyN38fpztt76tQ@mail.gmail.com>

Brian,

Thanks for your response.  I found this:
http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/Spanish%20Voxforge/
 as a first approximation to a Spanish acoustical model.  I'll have to try
it out once I get it running for English.  Are there any more examples
other than those in the Wiki?  Who is using this in production for a
reasonable number of channels?  I would like to get an idea of how CPU
intensive this is.

Thanks,

Guillermo

On Wed, Mar 18, 2015 at 10:16 AM, Brian West <brian at freeswitch.org> wrote:

> It should work per the examples on the wiki.  But sadly there are no
> acoustical models for any other language but English.
>
> On Tue, Mar 17, 2015 at 4:20 PM, Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> I am looking to experiment with some simple ASR. Small vocabulary (yes,
>> No, one, two.. zero, repeat).  I found mod_pocketsphinx in the old WIKI.  I
>> am wondering if this module is being maintained or if it has fallen into
>> disuse.  If the latter, can anyone share their experiences with other FREE
>> ASRs?  Are there any free still out there or must we go the commercial
>> route?  By the way, I am looking for both English and Spanish ASR.
>>
>> Thanks,
>>
>> --
>> Guillermo Ruiz Camauer
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/f5a649f1/attachment.html 

From brian at freeswitch.org  Wed Mar 18 21:52:21 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 13:52:21 -0500
Subject: [Freeswitch-users] g729 installer 404
In-Reply-To: <CALXC2kceYgp+307Nq5Kc-6qj+p4UhTb_kfp47ZF7+U35QSpkag@mail.gmail.com>
References: <CALXC2kdLE2dqeQ92F3X7suKcXaA1gz3j=LZPfxO5tMReUQkXnQ@mail.gmail.com>
	<CALXC2kceYgp+307Nq5Kc-6qj+p4UhTb_kfp47ZF7+U35QSpkag@mail.gmail.com>
Message-ID: <CAEJMVkCeNEOp2ZmJm6mLCr1NtX0uLY4e-ho0WzbjTQip+1JJew@mail.gmail.com>

Its fixed now, I didn't complete the fix from FS-7297,  You can always
download and run the installer yourself too, In the future please do not
report bugs to the mailing list, these belong on JIRA.

Thanks,

On Wed, Mar 18, 2015 at 1:30 PM, Peter Hartmann <peter at hartmanncomputer.com>
wrote:

> Hey,
> There seems to be a hosting problem with g729 installer.  It's located at:
>
> http://files.freeswitch.org/g729/fs-201501231218-installer
>
> However, when one attempts to build FS from source
> (1.5.15b+git~20150317T232708Z~42961b7f2e~64bit and also the 1.4
> version)  the scripts are attempting to retrieve it from:
>
> http://files.freeswitch.org/g729/fsg729-201501231218-installer
>
>
> resulting in a 404.  Note the different filename prefix.
>
> -----
>
> cd /usr/src/freeswitch/libs && /bin/bash
> /usr/src/freeswitch/build/getg729.sh fsg729-201501231218-installer
> --2015-03-18 12:31:47--
> http://files.freeswitch.org/g729/fsg729-201501231218-installer
> Resolving files.freeswitch.org (files.freeswitch.org)...
> 209.105.235.7, 2607:f348:1021::7
> Connecting to files.freeswitch.org
> (files.freeswitch.org)|209.105.235.7|:80... connected.
> HTTP request sent, awaiting response... 404 Not Found
> 2015-03-18 12:31:47 ERROR 404: Not Found.
> -------
>
>
>
> I'd file a JIRA but I don't think there's anything wrong with the
> code.    'fsg729' seems to be a more descriptive filename prefix as
> compared to 'fs'.    Sed got me through the build, but it seems
> something that should be fixed, no?  And if  'fs' is the preferred
> filename prefix I'd be happy to file a report.
>
>
> Thanks much,
>
>
>
> Peter Hartmann
> Hartmann Computer Consulting
> http://blog.hartmanncomputer.com
> (212)203-8870
>
> If I can't explain it to you in plain language, that means I don't
> understand it.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/df222f20/attachment.html 

From brian at freeswitch.org  Wed Mar 18 21:53:22 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 13:53:22 -0500
Subject: [Freeswitch-users] g729 installer 404
In-Reply-To: <CAEJMVkCeNEOp2ZmJm6mLCr1NtX0uLY4e-ho0WzbjTQip+1JJew@mail.gmail.com>
References: <CALXC2kdLE2dqeQ92F3X7suKcXaA1gz3j=LZPfxO5tMReUQkXnQ@mail.gmail.com>
	<CALXC2kceYgp+307Nq5Kc-6qj+p4UhTb_kfp47ZF7+U35QSpkag@mail.gmail.com>
	<CAEJMVkCeNEOp2ZmJm6mLCr1NtX0uLY4e-ho0WzbjTQip+1JJew@mail.gmail.com>
Message-ID: <CAEJMVkAWfcMfjyqxR2upiMsbvMKFRxN011iSmjwf2tvi=thVdQ@mail.gmail.com>

I've alos installed a Symlink:  fsg729-201501231218-installer ->
fs-201501231218-installer

On Wed, Mar 18, 2015 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:

> Its fixed now, I didn't complete the fix from FS-7297,  You can always
> download and run the installer yourself too, In the future please do not
> report bugs to the mailing list, these belong on JIRA.
>
> Thanks,
>
> On Wed, Mar 18, 2015 at 1:30 PM, Peter Hartmann <
> peter at hartmanncomputer.com> wrote:
>
>> Hey,
>> There seems to be a hosting problem with g729 installer.  It's located at:
>>
>> http://files.freeswitch.org/g729/fs-201501231218-installer
>>
>> However, when one attempts to build FS from source
>> (1.5.15b+git~20150317T232708Z~42961b7f2e~64bit and also the 1.4
>> version)  the scripts are attempting to retrieve it from:
>>
>> http://files.freeswitch.org/g729/fsg729-201501231218-installer
>>
>>
>> resulting in a 404.  Note the different filename prefix.
>>
>> -----
>>
>> cd /usr/src/freeswitch/libs && /bin/bash
>> /usr/src/freeswitch/build/getg729.sh fsg729-201501231218-installer
>> --2015-03-18 12:31:47--
>> http://files.freeswitch.org/g729/fsg729-201501231218-installer
>> Resolving files.freeswitch.org (files.freeswitch.org)...
>> 209.105.235.7, 2607:f348:1021::7
>> Connecting to files.freeswitch.org
>> (files.freeswitch.org)|209.105.235.7|:80... connected.
>> HTTP request sent, awaiting response... 404 Not Found
>> 2015-03-18 12:31:47 ERROR 404: Not Found.
>> -------
>>
>>
>>
>> I'd file a JIRA but I don't think there's anything wrong with the
>> code.    'fsg729' seems to be a more descriptive filename prefix as
>> compared to 'fs'.    Sed got me through the build, but it seems
>> something that should be fixed, no?  And if  'fs' is the preferred
>> filename prefix I'd be happy to file a report.
>>
>>
>> Thanks much,
>>
>>
>>
>> Peter Hartmann
>> Hartmann Computer Consulting
>> http://blog.hartmanncomputer.com
>> (212)203-8870
>>
>> If I can't explain it to you in plain language, that means I don't
>> understand it.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/92293905/attachment-0001.html 

From brian at freeswitch.org  Wed Mar 18 21:54:24 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 13:54:24 -0500
Subject: [Freeswitch-users] ASR and PocketSphinx
In-Reply-To: <CAPNTnnoqe8ONnxoNomHkvvOTkLGbJaRvQ+S6wyN38fpztt76tQ@mail.gmail.com>
References: <CAPNTnnoSquK6DdZ3ej_9uCrgoPyO4eu3n470W5b3W1KeVRTNsw@mail.gmail.com>
	<CAEJMVkDA9ah3jWE7PfuVDb+CTE9Npf2=F8CHg4sA7CPWM4nQng@mail.gmail.com>
	<CAPNTnnoqe8ONnxoNomHkvvOTkLGbJaRvQ+S6wyN38fpztt76tQ@mail.gmail.com>
Message-ID: <CAEJMVkD8_TrHOZCs1x9YNxUqW+rx60AcZDmr=fg0mETUtSOn=w@mail.gmail.com>

I don't think its what I would call production level ASR, its great for
playing around with, but there is a reason ASR is expensive, its hard to
get right.

On Wed, Mar 18, 2015 at 1:44 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> Brian,
>
> Thanks for your response.  I found this:
> http://sourceforge.net/projects/cmusphinx/files/Acoustic%20and%20Language%20Models/Spanish%20Voxforge/
>  as a first approximation to a Spanish acoustical model.  I'll have to try
> it out once I get it running for English.  Are there any more examples
> other than those in the Wiki?  Who is using this in production for a
> reasonable number of channels?  I would like to get an idea of how CPU
> intensive this is.
>
> Thanks,
>
> Guillermo
>
> On Wed, Mar 18, 2015 at 10:16 AM, Brian West <brian at freeswitch.org> wrote:
>
>> It should work per the examples on the wiki.  But sadly there are no
>> acoustical models for any other language but English.
>>
>> On Tue, Mar 17, 2015 at 4:20 PM, Guillermo Ruiz Camauer <
>> grcamauer at gmail.com> wrote:
>>
>>> I am looking to experiment with some simple ASR. Small vocabulary (yes,
>>> No, one, two.. zero, repeat).  I found mod_pocketsphinx in the old WIKI.  I
>>> am wondering if this module is being maintained or if it has fallen into
>>> disuse.  If the latter, can anyone share their experiences with other FREE
>>> ASRs?  Are there any free still out there or must we go the commercial
>>> route?  By the way, I am looking for both English and Spanish ASR.
>>>
>>> Thanks,
>>>
>>> --
>>> Guillermo Ruiz Camauer
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/73462323/attachment.html 

From brian at freeswitch.org  Wed Mar 18 21:57:25 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 13:57:25 -0500
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
Message-ID: <CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>

You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I would
highly recommend you re-test with Master or at the very least 1.4.17 or
1.4.18 which should be out later today.

1.2 is not receiving patches, updates or support moving forward, our
release branch is 1.4.x




On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net> wrote:

> Hello,
>
> Not sure whether this belong in the users list or the dev list, but when
> in doubt; start with users :-)
>
> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
> Asterisk on the other, and am seeing some strange behavior when calls are
> being forked on the Asterisk.
>
> Call setup is like this:
> 1. FreeSWITCH receives INVITE from Kamailio
> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
> (each with a unique call-id)
> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
> Call-ID from step 2.
>
> This is causing problems with one of the MGWs behind Kamailio, which is
> seeing multiple INVITEs to different destinations with the same Call-ID.
>
> So, firstly, why is FreeSWITCH reusing call-ids?
>
> Secondly, how is it matching up the calls? I can't find anything common in
> the INVITEs, other than the source number and obviously that the IP sent to
> and received from is the same.
>
> I'm not sure if this is intended behavior or not, but is there a way to
> have FreeSWITCH not do that?
>
> Regards,
> ?rn
>
> P.S. Here is the sequence of INVITEs. I also have the console log (for a
> different call) if needed.
>
> *INVITE sent to FreeSWITCH by Kamailio:*
> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
> Record-Route: <sip:172.25.200.101;lr=on>
> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
> Via: SIP/2.0/UDP 172.25.200.121:5080
> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
> Max-Forwards: 16
> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
> To: <sip:5344446 at 172.25.200.101>
> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
> CSeq: 73014324 INVITE
> Contact: <sip:mod_sofia at 172.25.200.121:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.7
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 229
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
> s=FreeSWITCH
> c=IN IP4 172.25.200.121
> t=0 0
> m=audio 19026 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
>
> *INVITE sent to Asterisk by FreeSWITCH:*
> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
> Max-Forwards: 15
> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
> To: <sip:5344446 at 172.26.0.62:5060>
> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 10.11.12.13:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> 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: 223
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
> s=FreeSWITCH
> c=IN IP4 10.11.12.13
> t=0 0
> m=audio 23230 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
> Max-Forwards: 70
> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
> To: <sip:7712552 at 10.11.12.13>
> Contact: <sip:4151502 at 172.26.0.62:5060>
> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert2
> Date: Wed, 18 Mar 2015 17:47:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Diversion: <sip:5344446 at 172.26.0.62>
> Content-Type: application/sdp
> Content-Length: 312
>
> v=0
> o=root 693576967 693576967 IN IP4 172.26.0.62
> s=Asterisk PBX 1.8.15-cert2
> c=IN IP4 172.26.0.62
> t=0 0
> m=audio 30440 RTP/AVP 8 0 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
> Max-Forwards: 70
> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
> To: <sip:6595454 at 10.11.12.13>
> Contact: <sip:4151502 at 172.26.0.62:5060>
> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.15-cert2
> Date: Wed, 18 Mar 2015 17:47:11 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Diversion: <sip:5344446 at 172.26.0.62>
> Content-Type: application/sdp
> Content-Length: 310
>
> v=0
> o=root 89056081 89056081 IN IP4 172.26.0.62
> s=Asterisk PBX 1.8.15-cert2
> c=IN IP4 172.26.0.62
> t=0 0
> m=audio 30708 RTP/AVP 8 0 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
> Max-Forwards: 69
> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
> To: <sip:7712552 at 172.25.200.101>
> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 172.25.200.111:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 209
> Diversion: <sip:5344446 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
> s=FreeSWITCH
> c=IN IP4 172.25.200.111
> t=0 0
> m=audio 19804 RTP/AVP 8 0 9 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
> Max-Forwards: 69
> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
> To: <sip:6595454 at 172.25.200.101>
> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
> CSeq: 73014353 INVITE
> Contact: <sip:mod_sofia at 172.25.200.111:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.12
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 209
> Diversion: <sip:5344446 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
> s=FreeSWITCH
> c=IN IP4 172.25.200.111
> t=0 0
> m=audio 31376 RTP/AVP 8 0 9 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/0632f4f9/attachment-0001.html 

From peter at hartmanncomputer.com  Wed Mar 18 21:59:11 2015
From: peter at hartmanncomputer.com (Peter Hartmann)
Date: Wed, 18 Mar 2015 14:59:11 -0400
Subject: [Freeswitch-users] g729 installer 404
In-Reply-To: <CAEJMVkAWfcMfjyqxR2upiMsbvMKFRxN011iSmjwf2tvi=thVdQ@mail.gmail.com>
References: <CALXC2kdLE2dqeQ92F3X7suKcXaA1gz3j=LZPfxO5tMReUQkXnQ@mail.gmail.com>
	<CALXC2kceYgp+307Nq5Kc-6qj+p4UhTb_kfp47ZF7+U35QSpkag@mail.gmail.com>
	<CAEJMVkCeNEOp2ZmJm6mLCr1NtX0uLY4e-ho0WzbjTQip+1JJew@mail.gmail.com>
	<CAEJMVkAWfcMfjyqxR2upiMsbvMKFRxN011iSmjwf2tvi=thVdQ@mail.gmail.com>
Message-ID: <CALXC2keqSPbvhE1ChezM4QrFdRj3bGRfKf+WkN+MV1U8a0Gmbw@mail.gmail.com>

Thanks Brian.




Peter Hartmann
Hartmann Computer Consulting
http://blog.hartmanncomputer.com
(212)203-8870

If I can't explain it to you in plain language, that means I don't
understand it.

On Wed, Mar 18, 2015 at 2:53 PM, Brian West <brian at freeswitch.org> wrote:

> I've alos installed a Symlink:  fsg729-201501231218-installer ->
> fs-201501231218-installer
>
> On Wed, Mar 18, 2015 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Its fixed now, I didn't complete the fix from FS-7297,  You can always
>> download and run the installer yourself too, In the future please do not
>> report bugs to the mailing list, these belong on JIRA.
>>
>> Thanks,
>>
>> On Wed, Mar 18, 2015 at 1:30 PM, Peter Hartmann <
>> peter at hartmanncomputer.com> wrote:
>>
>>> Hey,
>>> There seems to be a hosting problem with g729 installer.  It's located
>>> at:
>>>
>>> http://files.freeswitch.org/g729/fs-201501231218-installer
>>>
>>> However, when one attempts to build FS from source
>>> (1.5.15b+git~20150317T232708Z~42961b7f2e~64bit and also the 1.4
>>> version)  the scripts are attempting to retrieve it from:
>>>
>>> http://files.freeswitch.org/g729/fsg729-201501231218-installer
>>>
>>>
>>> resulting in a 404.  Note the different filename prefix.
>>>
>>> -----
>>>
>>> cd /usr/src/freeswitch/libs && /bin/bash
>>> /usr/src/freeswitch/build/getg729.sh fsg729-201501231218-installer
>>> --2015-03-18 12:31:47--
>>> http://files.freeswitch.org/g729/fsg729-201501231218-installer
>>> Resolving files.freeswitch.org (files.freeswitch.org)...
>>> 209.105.235.7, 2607:f348:1021::7
>>> Connecting to files.freeswitch.org
>>> (files.freeswitch.org)|209.105.235.7|:80... connected.
>>> HTTP request sent, awaiting response... 404 Not Found
>>> 2015-03-18 12:31:47 ERROR 404: Not Found.
>>> -------
>>>
>>>
>>>
>>> I'd file a JIRA but I don't think there's anything wrong with the
>>> code.    'fsg729' seems to be a more descriptive filename prefix as
>>> compared to 'fs'.    Sed got me through the build, but it seems
>>> something that should be fixed, no?  And if  'fs' is the preferred
>>> filename prefix I'd be happy to file a report.
>>>
>>>
>>> Thanks much,
>>>
>>>
>>>
>>> Peter Hartmann
>>> Hartmann Computer Consulting
>>> http://blog.hartmanncomputer.com
>>> (212)203-8870
>>>
>>> If I can't explain it to you in plain language, that means I don't
>>> understand it.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/10ef3948/attachment.html 

From brian at freeswitch.org  Wed Mar 18 22:09:42 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 14:09:42 -0500
Subject: [Freeswitch-users] sofia recovery,
	A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CACOYK=h8h=L6r3Y2UNSmsLNtT=U02eb43Fehe=aGWMJMKfpgqQ@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>
	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>
	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>
	<CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>
	<CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>
	<CACOYK=iDD35byh3xM+LSZeniX-=Tjyd_7E2i41iDZEQSvsQLXw@mail.gmail.com>
	<CACOYK=h8h=L6r3Y2UNSmsLNtT=U02eb43Fehe=aGWMJMKfpgqQ@mail.gmail.com>
Message-ID: <CAEJMVkAcRCmcS2+2fhNW42Coqs6pv=hUm6kUBVFr+mM_4tH4GA@mail.gmail.com>

You'll need to either build it yourself from source or 'debian/util.sh
build-all' from a master checkout.

On Tue, Mar 17, 2015 at 4:44 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> duh, deb-master repo is about a month old.
>
> On Tue, Mar 17, 2015 at 10:35 PM, Stanislav Sinyagin
> <ssinyagin at gmail.com> wrote:
> > http://files.freeswitch.org/repo/deb-master/debian/
> > seems to be quite fresh. How often is it updated?
> >
> > On Tue, Mar 17, 2015 at 5:01 PM, Joseph Dickson <jdickson at evolvetsi.com>
> wrote:
> >> Understood.. I'll spend some time this week getting things compiled from
> >> source instead of using the binary packages, and see if behavior is the
> >> same.
> >>
> >>
> >> On Tue, Mar 17, 2015 at 11:56 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> Start by always testing on master.  Today's master is tomorrows
> release.
> >>> Todays release is yesterdays problem that won't be solved.
> >>> secondly, report issues to JIRA (after reproducing on master)
> >>>
> >>>
> >>> On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson <
> jdickson at evolvetsi.com>
> >>> wrote:
> >>>>
> >>>> Yes, I can definitely see where master-master mode in MySQL can put
> >>>> you in a pickle consistency wise..  I just wish Postgres was easier
> >>>> with regard to fail-back.. failover isn't a problem, but getting
> >>>> things to fail back requires rsync of data files, etc..
> >>>>
> >>>> Before I set that up, I'd like to see sofia recover work in my
> >>>> environment..  Currently it appears that the correct information is
> >>>> being retrieved from the database, but that FS is hanging up the A leg
> >>>> with INCOMPATIBLE_DESTINATION, and I can't quite understand the cause
> >>>> behind that hangup.
> >>>>
> >>>> Is there better logging that I can grab that might be more helpful?
> >>>>
> >>>> Thanks!
> >>>>
> >>>> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov <s.safarov at gmail.com>
> >>>> wrote:
> >>>> > I has used MySQL in master-master cluster and find it is unstable
> >>>> > configuration. Many troubles with cluster consistence.
> >>>> > I recommend use configuration master-slave with tools DB
> >>>> > synchronization.
> >>>> >
> >>>> > Also migration to PostgreSQL cluster will be smart decision.
> >>>> >
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> >>>
> >>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> >>> http://twitter.com/FreeSWITCH
> >>> ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+
> >>>
> >>> ClueCon Weekly Development Call
> >>> ? sip:888 at conference.freeswitch.org  ? +19193869900
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/43d24836/attachment-0001.html 

From sdevoy at bizfocused.com  Wed Mar 18 22:12:42 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 18 Mar 2015 19:12:42 +0000
Subject: [Freeswitch-users] Call intercept.
Message-ID: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>

Hi All,

I have directed call intercept running via dialplan **\d\d  (two digit extensions).

In this pastebin:  https://pastebin.freeswitch.org/24023

You should see a call come from the IVR into "MAINLINE" which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>

Can someone who is more log savvy than I review that log and tell me what happened?

Perhaps my dialplan is missing something:

    <extension name="directed-call-pickup">
      <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?

Thanks,
Sean Devoy

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/d4a92006/attachment.html 

From t.mahe at b-and-c.net  Wed Mar 18 22:24:38 2015
From: t.mahe at b-and-c.net (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Wed, 18 Mar 2015 12:24:38 -0700
Subject: [Freeswitch-users] sofia recovery,
 A leg is being hung up with INCOMPATIBLE_DESTINATION
In-Reply-To: <CAEJMVkAcRCmcS2+2fhNW42Coqs6pv=hUm6kUBVFr+mM_4tH4GA@mail.gmail.com>
References: <CAOHpEouAvKLtGLhhFWiehFj_pXuZpzQ60=C6q7GGMj18zaGp2A@mail.gmail.com>	<CAHtxdDfvJcWuA6R6LVV4Q9=RokofOF9M5a0E0OsPS9CtxQ86Jw@mail.gmail.com>	<CAOHpEosO+ZvjWA_tX6COR2B_vsaV8dJqj0DM4z6AXxEUP32TCg@mail.gmail.com>	<CAKbxfG8tURSvCjadUvv22JtS1snOtWCZX4KaTtuc-vvnj87msQ@mail.gmail.com>	<CAOHpEot6Ao8VycHYknc0X01ByQv-Y0r8LaKWTmhe=8jC9RDdgg@mail.gmail.com>	<CACOYK=iDD35byh3xM+LSZeniX-=Tjyd_7E2i41iDZEQSvsQLXw@mail.gmail.com>	<CACOYK=h8h=L6r3Y2UNSmsLNtT=U02eb43Fehe=aGWMJMKfpgqQ@mail.gmail.com>
	<CAEJMVkAcRCmcS2+2fhNW42Coqs6pv=hUm6kUBVFr+mM_4tH4GA@mail.gmail.com>
Message-ID: <5509D0F6.6050804@b-and-c.net>

Brian,

Till we have an official freeswitch build environnment, I'm maintaining
the unstable repo at http://repo.remote-shell.net

Stable is no more maintained for the moment, but I'll add the CI build
job as soon as I have a chance.

Each commit on the master branch of freeswitch triggers a debian package
build, so it's always up to date with stash.

Everyone is welcome to use it to test ( master is not recommanded
anymore on production ).

Le 18/03/2015 12:09, Brian West a ?crit :
> You'll need to either build it yourself from source or 'debian/util.sh
> build-all' from a master checkout.
>
> On Tue, Mar 17, 2015 at 4:44 PM, Stanislav Sinyagin
> <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
>
>     duh, deb-master repo is about a month old.
>
>     On Tue, Mar 17, 2015 at 10:35 PM, Stanislav Sinyagin
>     <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
>     > http://files.freeswitch.org/repo/deb-master/debian/
>     > seems to be quite fresh. How often is it updated?
>     >
>     > On Tue, Mar 17, 2015 at 5:01 PM, Joseph Dickson
>     <jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>> wrote:
>     >> Understood.. I'll spend some time this week getting things
>     compiled from
>     >> source instead of using the binary packages, and see if
>     behavior is the
>     >> same.
>     >>
>     >>
>     >> On Tue, Mar 17, 2015 at 11:56 AM, Anthony Minessale
>     >> <anthony.minessale at gmail.com
>     <mailto:anthony.minessale at gmail.com>> wrote:
>     >>>
>     >>> Start by always testing on master.  Today's master is
>     tomorrows release.
>     >>> Todays release is yesterdays problem that won't be solved.
>     >>> secondly, report issues to JIRA (after reproducing on master)
>     >>>
>     >>>
>     >>> On Fri, Mar 13, 2015 at 9:48 AM, Joseph Dickson
>     <jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>>
>     >>> wrote:
>     >>>>
>     >>>> Yes, I can definitely see where master-master mode in MySQL
>     can put
>     >>>> you in a pickle consistency wise..  I just wish Postgres was
>     easier
>     >>>> with regard to fail-back.. failover isn't a problem, but getting
>     >>>> things to fail back requires rsync of data files, etc..
>     >>>>
>     >>>> Before I set that up, I'd like to see sofia recover work in my
>     >>>> environment..  Currently it appears that the correct
>     information is
>     >>>> being retrieved from the database, but that FS is hanging up
>     the A leg
>     >>>> with INCOMPATIBLE_DESTINATION, and I can't quite understand
>     the cause
>     >>>> behind that hangup.
>     >>>>
>     >>>> Is there better logging that I can grab that might be more
>     helpful?
>     >>>>
>     >>>> Thanks!
>     >>>>
>     >>>> On Thu, Mar 12, 2015 at 2:51 PM, Sergey Safarov
>     <s.safarov at gmail.com <mailto:s.safarov at gmail.com>>
>     >>>> wrote:
>     >>>> > I has used MySQL in master-master cluster and find it is
>     unstable
>     >>>> > configuration. Many troubles with cluster consistence.
>     >>>> > I recommend use configuration master-slave with tools DB
>     >>>> > synchronization.
>     >>>> >
>     >>>> > Also migration to PostgreSQL cluster will be smart decision.
>     >>>> >
>     >>>>
>     >>>>
>     _________________________________________________________________________
>     >>>> 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://confluence.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       ? @anthmfs  ? @FreeSWITCH  ?
>     >>>
>     >>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>     >>> http://twitter.com/FreeSWITCH
>     >>> ? irc.freenode.net <http://irc.freenode.net> #freeswitch ?
>     http://freeswitch.org/g+
>     >>>
>     >>> ClueCon Weekly Development Call
>     >>> ? sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>  ? +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://confluence.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://confluence.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://confluence.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
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/039d1bcd/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 473 bytes
Desc: OpenPGP digital signature
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/039d1bcd/attachment-0001.bin 

From brian at freeswitch.org  Wed Mar 18 22:39:03 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 14:39:03 -0500
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>

intercept takes a UUID, not an extension number.

On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

>  Hi All,
>
>
>
> I have directed call intercept running via dialplan **\d\d  (two digit
> extensions).
>
>
>
> In this pastebin:  https://pastebin.freeswitch.org/24023
>
>
>
> You should see a call come from the IVR into ?MAINLINE? which dials ext
> 10, then after delay adds other extensions.  In this case, the person from
> extension 10 was at extension 20 and dialed **10.  They report it did not
> work.  I must say I am not clear if it worked or not.  It appears to work,
> but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the
> last dialplan instruction, hanging up.
> <sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>
>
>
>
> Can someone who is more log savvy than I review that log and tell me what
> happened?
>
>
>
> Perhaps my dialplan is missing something:
>
>
>
>     <extension name="directed-call-pickup">
>
>       <!-- picks up the call with db key saved like this:
> <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>
> -->
>
>       <condition field="destination_number" expression="^\*\*(\d\d)$">
>
>         <action application="answer"/>
>
>         <action application="set" data="intercept_unanswered_only=true"/>
>
>         <action application="intercept" data="
> ${db(select/esta_call_pickup_uuid/$1)}"/>
>
>         <action application="sleep" data="2000"/>
>
>       </condition>
>
>     </extension>
>
>
>
> I see in the log the uuid for the appropriate call is in the intercept
> statement.  After intercept, I am missing something to bridge the 2 legs
> together?
>
>
>
> Thanks,
>
> Sean Devoy
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/eec80fe5/attachment.html 

From sdevoy at bizfocused.com  Wed Mar 18 22:45:38 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 18 Mar 2015 19:45:38 +0000
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
Message-ID: <BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>

Yes, that is what I pass to the intercept call vie the dialplan/database.  Check the log:

62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan: sofia/external/20 at fs_esta.bizfocused.com Action intercept(${db(select/esta_call_pickup_uuid/10)})
. . . .
62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE sofia/external/20 at fs_esta.bizfocused.com intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)


Sean

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, March 18, 2015 3:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

intercept takes a UUID, not an extension number.

On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Hi All,

I have directed call intercept running via dialplan **\d\d  (two digit extensions).

In this pastebin:  https://pastebin.freeswitch.org/24023

You should see a call come from the IVR into ?MAINLINE? which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>

Can someone who is more log savvy than I review that log and tell me what happened?

Perhaps my dialplan is missing something:

    <extension name="directed-call-pickup">
      <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?

Thanks,
Sean Devoy


_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/38f6a870/attachment-0001.html 

From grcamauer at gmail.com  Wed Mar 18 22:49:44 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 18 Mar 2015 16:49:44 -0300
Subject: [Freeswitch-users] Problem with make current
Message-ID: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>

I am having problems trying to update my Freeswitch instances.  MAKE
CURRENT ends with:

make[1]: Entering directory `/usr/src/freeswitch'
Pulling updates...
error: Failed connect to freeswitch.org:443; Connection timed out while
accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs
fatal: HTTP request failed
make[1]: *** [update] Error 1
make[1]: Leaving directory `/usr/src/freeswitch'


Are the stash servers down?


-- 
Guillermo Ruiz Camauer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/a5ffd59d/attachment.html 

From krice at freeswitch.org  Wed Mar 18 23:25:39 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 18 Mar 2015 20:25:39 +0000
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
Message-ID: <5509df43323ec_f3fe8bd32881237@resque-worker-ip-10-101-150-189.mail>

New Post on freeswitch.org from krice387
check it out at http://ift.tt/1GpR0Yu
ClueCon Call For Speakers is Open!
Call for speakers at ClueCon 2015 is open!

Visit http://ift.tt/1FBM7gN to submit your speaking proposal Below!

What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves technical presentations. In general, the more technical the presentation, the better.

If you are thinking about a presentation then consider these points:
* ClueCon talks are 30 minutes in length, including Q&amp;A time with the audience (Lightning Talks are also open! 10 minute talks, 5 minutes Q&amp;A)
* ClueCon has a special focus on open source communications, VoIP, and telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and Kamailio
* Attendees enjoy hearing about projects built with open source tools, especially those in a production environment
* Highly technical discussions that show the nuts and bolts are especially well-liked
* The audience appreciates seeing and participating in live demonstrations

We are especially interested in Security-related talks and demonstrations

Please register your?proposals at http://ift.tt/1FBM7gN. To complete registration you need the following items:
* Working title
* Brief description of the talk (abstract)
* Name of the presenter(s)
* Bio and headshot of presenter(s)
* Presenter?s contact information (including mobile phone the presenter will have with them at the conference)

Don?t delay! Speaking proposals must be in by July 4, 2015 and scheduling requests are handled on a first come first serve basis.

ClueCon 2015: See You There!
For more details on ClueCon see https://cluecon.com.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/5897f54d/attachment.html 

From ali.jibran44 at gmail.com  Thu Mar 19 00:48:18 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 18 Mar 2015 21:48:18 +0000
Subject: [Freeswitch-users] mod_callcenter custom headers
Message-ID: <CAP9PqkSQtR4bdpGjFtR5Jj=Bcrj=N=zCuAFRKJn=RYxEvtP7og@mail.gmail.com>

Hi all.
I wanted to know if it was possible to pass/export variables or custom
headers into mod callcenter.
Like if I pass a variable in A-Leg I can retrieve it in the B-Leg whenever
an agent picks up.

Is this possible?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/eb6bc49c/attachment.html 

From mike at jerris.com  Thu Mar 19 01:12:04 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 18 Mar 2015 18:12:04 -0400
Subject: [Freeswitch-users] mod_callcenter custom headers
In-Reply-To: <CAP9PqkSQtR4bdpGjFtR5Jj=Bcrj=N=zCuAFRKJn=RYxEvtP7og@mail.gmail.com>
References: <CAP9PqkSQtR4bdpGjFtR5Jj=Bcrj=N=zCuAFRKJn=RYxEvtP7og@mail.gmail.com>
Message-ID: <D9D7931E-1851-420F-9D86-3AD00BE346FC@jerris.com>

As you have already asked in another thread... yes we know.

> On Mar 18, 2015, at 5:48 PM, Ali Jibran <ali.jibran44 at gmail.com> wrote:
> 
> Hi all.
> I wanted to know if it was possible to pass/export variables or custom headers into mod callcenter.
> Like if I pass a variable in A-Leg I can retrieve it in the B-Leg whenever an agent picks up.
> 
> Is this possible? _________________________________________________________________________



From ssinyagin at gmail.com  Thu Mar 19 02:43:54 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 19 Mar 2015 00:43:54 +0100
Subject: [Freeswitch-users] Simple PBX tutorial
Message-ID: <CACOYK=i0Rn+=urZC-J6FKHw6RwxEtK6Jk2tLSfyjPiiRu3ZmNQ@mail.gmail.com>

hi,

Here's a short article that I wrote to help people start using FreeSWITCH:
https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutorial_01_simple_pbx.md

your feedback will be appreciated.


From brian at freeswitch.org  Thu Mar 19 03:08:06 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 18 Mar 2015 19:08:06 -0500
Subject: [Freeswitch-users] Problem with make current
In-Reply-To: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>
References: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>
Message-ID: <CAEJMVkDrnWw9Vqg=5i=UJ7xjLfCcaJDUa2zL0Kr2SuySEK5sxQ@mail.gmail.com>

What country are you accessing from?  That should only be doing a git pull,
does that fail too?

On Wed, Mar 18, 2015 at 2:49 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> I am having problems trying to update my Freeswitch instances.  MAKE
> CURRENT ends with:
>
> make[1]: Entering directory `/usr/src/freeswitch'
> Pulling updates...
> error: Failed connect to freeswitch.org:443; Connection timed out while
> accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs
> fatal: HTTP request failed
> make[1]: *** [update] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch'
>
>
> Are the stash servers down?
>
>
> --
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150318/722030d9/attachment.html 

From jdayola at spingine.com  Thu Mar 19 08:20:12 2015
From: jdayola at spingine.com (jdayola at spingine.com)
Date: Thu, 19 Mar 2015 13:20:12 +0800
Subject: [Freeswitch-users] Simple PBX tutorial
In-Reply-To: <CACOYK=i0Rn+=urZC-J6FKHw6RwxEtK6Jk2tLSfyjPiiRu3ZmNQ@mail.gmail.com>
References: <CACOYK=i0Rn+=urZC-J6FKHw6RwxEtK6Jk2tLSfyjPiiRu3ZmNQ@mail.gmail.com>
Message-ID: <dd67e6dad4c970b664552a0b49661e91@spingine.com>

Nice job. This is a big help for beginners like me. Hope you can do more 
guides.

On 2015-03-19 7:43 am, Stanislav Sinyagin wrote:
> hi,
> 
> Here's a short article that I wrote to help people start using 
> FreeSWITCH:
> https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutorial_01_simple_pbx.md
> 
> your feedback will be appreciated.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Thu Mar 19 10:59:37 2015
From: gb at cm.nl (Grant Bagdasarian)
Date: Thu, 19 Mar 2015 07:59:37 +0000
Subject: [Freeswitch-users] Keep A Leg alive after bridge fails
Message-ID: <987ed982fa414d5f9fce49e60932aa74@CM-EX-V01.cm.local>

Hello,

I'm using outbound ESL socket to control calls and one of the actions is to bridge the A leg to another destination.
This works great when the remote party answers, but when the remote party does not answer or is busy the bridge fails and disconnects the A leg.
I wish to keep the A leg alive. I've tried setting different kinds of channel variables on the a leg and even the b leg, but without success.

There is a thread from a few years ago discussing this very same issue (although they were talking about inbound socket), so I was wondering if anyone has a solution for this?

Continue_on_fail, hangup_after_bridge, park_after_bridge, ignore_early_media, all don't work.

Any ideas?



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/d6935adf/attachment-0001.html 

From gb at cm.nl  Thu Mar 19 11:09:48 2015
From: gb at cm.nl (Grant Bagdasarian)
Date: Thu, 19 Mar 2015 08:09:48 +0000
Subject: [Freeswitch-users] Keep A Leg alive after bridge fails
In-Reply-To: <987ed982fa414d5f9fce49e60932aa74@CM-EX-V01.cm.local>
References: <987ed982fa414d5f9fce49e60932aa74@CM-EX-V01.cm.local>
Message-ID: <98335d476d8847ff859d6e2d1747e1f7@CM-EX-V01.cm.local>

Nevermind, my uuid_setvar wasn't working as expected.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian
Sent: Thursday, March 19, 2015 9:00 AM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: [Freeswitch-users] Keep A Leg alive after bridge fails

Hello,

I'm using outbound ESL socket to control calls and one of the actions is to bridge the A leg to another destination.
This works great when the remote party answers, but when the remote party does not answer or is busy the bridge fails and disconnects the A leg.
I wish to keep the A leg alive. I've tried setting different kinds of channel variables on the a leg and even the b leg, but without success.

There is a thread from a few years ago discussing this very same issue (although they were talking about inbound socket), so I was wondering if anyone has a solution for this?

Continue_on_fail, hangup_after_bridge, park_after_bridge, ignore_early_media, all don't work.

Any ideas?



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/016e89cb/attachment.html 

From denis at ringme.ru  Thu Mar 19 15:35:38 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Thu, 19 Mar 2015 15:35:38 +0300
Subject: [Freeswitch-users] bug with events?
In-Reply-To: <E501824C-F192-4335-85E0-C240679F433A@jerris.com>
References: <5509824F.6090608@ringme.ru>
	<E501824C-F192-4335-85E0-C240679F433A@jerris.com>
Message-ID: <550AC29A.4070009@ringme.ru>

If we send INVITE, we HAVE Call-ID (it's requred for call).

On 18.03.2015 17:04, Michael Jerris wrote:
> This is probably expected, although with so little information as provided I can't say for sure.  You might have luck finding it from the other leg's cdr, we might not even have that information yet.
>
>> On Mar 18, 2015, at 9:49 AM, ????? <denis at ringme.ru> wrote:
>>
>> Hi.
>>
>> Possible, we found a bug.
>> If we call to freeswitch with redirect to mobile phone and hangup before
>> state "ringing" - ESL events show leg B without Call-ID (i.e.
>> CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...)
>>
>> example (json view)
>> https://pastebin.freeswitch.org/24022
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From denis at ringme.ru  Thu Mar 19 15:38:12 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Thu, 19 Mar 2015 15:38:12 +0300
Subject: [Freeswitch-users] Disable Re-INVITE on REFER ?
In-Reply-To: <D2117D269EE53A49B1CB699FE2E7B3241F2BBC28@SJCPEX01CL01.citrite.net>
References: <D2117D269EE53A49B1CB699FE2E7B3241F2BBC28@SJCPEX01CL01.citrite.net>
Message-ID: <550AC334.4030700@ringme.ru>

<param name="disable-transfer" value="true"/>
?

On 17.03.2015 20:02, Patrick Shea wrote:
> ?I am seeing 2 problems that I can't seem to find the answers to. I've scoured the Freeswitch Wiki (old), Confluence, and the users list.
>
> I have 2 SIP calls bridge through Freeswitch - they are CONTROL channels, so no media is negotiated. A Leg is SIP over Websockets, B Leg is SIP.
>
> SIPjs (WS)  <---- A Leg ---> Freeswitch  <---- B Leg ---> EP
>
> The problem I have is in some cases I get a REFER from the EP to transfer the call somewhere else, and this results in a Re-INVITE from Freeswitch to the SIPjs client. (Ideally, I'd like to "skip" this re-Invite and just bridge the A-leg to the new AP).
>
> Problem 1.
> Freeswitch sends media in the SDP of the Re-INVITE. This doesn't look right to me, but I assume using some local codecs specified somewhere.
>        m=audio 28692 RTP/AVP 0 8
>        a=rtpmap:0 PCMU/8000
>        a=rtpmap:8 PCMA/8000
> Can I disable these and just send the original m= line received for the initial call?
>        m=application 28118 udp as
>
> Problem 2.
> SIPjs doesn't support Re-INVITE, yet, so it responds with 488 Unacceptable Here. At this point, Freeswitch, drops the A leg and the transfer fails. I can't find a solution to work around this - any ideas?
>
> Like I said, Ideally I'd like to skip the re-INVITE altogether. Currently debugging SOFIA to see how this can be done - is there a parameter already somewhere that I am overlooking?
>
> Thanks
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Mar 19 15:36:45 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 19 Mar 2015 07:36:45 -0500
Subject: [Freeswitch-users] Disable Re-INVITE on REFER ?
In-Reply-To: <550AC334.4030700@ringme.ru>
References: <D2117D269EE53A49B1CB699FE2E7B3241F2BBC28@SJCPEX01CL01.citrite.net>
	<550AC334.4030700@ringme.ru>
Message-ID: <CAEJMVkA=UNWdBXRo3xscyJknxk7wsjOkQhTYT99w4bnZ37Hriw@mail.gmail.com>

Disable session timers wil probably fix this

On Thursday, March 19, 2015, ????? <denis at ringme.ru> wrote:

> <param name="disable-transfer" value="true"/>
> ?
>
> On 17.03.2015 20:02, Patrick Shea wrote:
> > ?I am seeing 2 problems that I can't seem to find the answers to. I've
> scoured the Freeswitch Wiki (old), Confluence, and the users list.
> >
> > I have 2 SIP calls bridge through Freeswitch - they are CONTROL
> channels, so no media is negotiated. A Leg is SIP over Websockets, B Leg is
> SIP.
> >
> > SIPjs (WS)  <---- A Leg ---> Freeswitch  <---- B Leg ---> EP
> >
> > The problem I have is in some cases I get a REFER from the EP to
> transfer the call somewhere else, and this results in a Re-INVITE from
> Freeswitch to the SIPjs client. (Ideally, I'd like to "skip" this re-Invite
> and just bridge the A-leg to the new AP).
> >
> > Problem 1.
> > Freeswitch sends media in the SDP of the Re-INVITE. This doesn't look
> right to me, but I assume using some local codecs specified somewhere.
> >        m=audio 28692 RTP/AVP 0 8
> >        a=rtpmap:0 PCMU/8000
> >        a=rtpmap:8 PCMA/8000
> > Can I disable these and just send the original m= line received for the
> initial call?
> >        m=application 28118 udp as
> >
> > Problem 2.
> > SIPjs doesn't support Re-INVITE, yet, so it responds with 488
> Unacceptable Here. At this point, Freeswitch, drops the A leg and the
> transfer fails. I can't find a solution to work around this - any ideas?
> >
> > Like I said, Ideally I'd like to skip the re-INVITE altogether.
> Currently debugging SOFIA to see how this can be done - is there a
> parameter already somewhere that I am overlooking?
> >
> > Thanks
> > Patrick
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.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 <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/f43e7429/attachment.html 

From mike at jerris.com  Thu Mar 19 16:41:10 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 19 Mar 2015 09:41:10 -0400
Subject: [Freeswitch-users] bug with events?
In-Reply-To: <550AC29A.4070009@ringme.ru>
References: <5509824F.6090608@ringme.ru>
	<E501824C-F192-4335-85E0-C240679F433A@jerris.com>
	<550AC29A.4070009@ringme.ru>
Message-ID: <78C2F90D-2A8D-47D5-B14C-6896F509860C@jerris.com>

Again, you have provided basically no information about what is going on so there is very little anyone can do to help you.

> On Mar 19, 2015, at 8:35 AM, ????? <denis at ringme.ru> wrote:
> 
> If we send INVITE, we HAVE Call-ID (it's requred for call).
> 
> On 18.03.2015 17:04, Michael Jerris wrote:
>> This is probably expected, although with so little information as provided I can't say for sure.  You might have luck finding it from the other leg's cdr, we might not even have that information yet.
>> 
>>> On Mar 18, 2015, at 9:49 AM, ????? <denis at ringme.ru> wrote:
>>> 
>>> Hi.
>>> 
>>> Possible, we found a bug.
>>> If we call to freeswitch with redirect to mobile phone and hangup before
>>> state "ringing" - ESL events show leg B without Call-ID (i.e.
>>> CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE...)
>>> 
>>> example (json view)
>>> https://pastebin.freeswitch.org/24022
>>> 



From victor.medina at cibersys.com  Thu Mar 19 16:46:51 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 19 Mar 2015 09:16:51 -0430
Subject: [Freeswitch-users] Multilanguages in dialplan and voicemail
Message-ID: <CACXsapOTMPtEe_9-R-YUVKfkNUO5UX89E999XU7L+MiW7-6+Nw@mail.gmail.com>

Hi guys!

Im having troubles trying to support multilanguages destinations in
freeswitch dialplan and voicemail.

I have modules correctly loaded in modules as:

    <load module="mod_say_en"/>
    <load module="mod_say_es"/>


In vars I only have the following...

<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>



And two internal destinations defined as...

Spanish exts....

    <extension name="Local_Extension1">
      <condition field="destination_number"
expression="^([0-9][0-9][0-9])$">
      .......................
      <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

English.....

    <extension name="Local_Extension2">
      <condition field="destination_number"
expression="^(1[0-9][0-9][0-9])$">
      .................
      <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>


I need that Local_Extension1 have a spanish VoiceMail Voice and
Local_Extension2 have a english VoiceMails. I have tried explicitly setting
languages in each of the dialplan sections like..

  <action application="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/>
  <action application="set" data="default_language=es"/>
  <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>


I'm even seeing correctly processed in console:

Dialplan: sofia/internal/163 at cibersys.com Action sleep(1000)
Dialplan: sofia/internal/163 at cibersys.com Action
set(sound_prefix=/opt/CloudVoice-vPBX/fsw14/sounds/es/mx/maria)
Dialplan: sofia/internal/163 at cibersys.com Action set(default_language=es)
Dialplan: sofia/internal/163 at cibersys.com Action
bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})


Any ideas on what could I do? Are things setted in vars.xml taking
precedence ? For example should I remove sound_prefix from vars?

Also... any way to reload vars.xml?




-- 

V?ctor E. Medina M.
Software
[image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image:
ve]
BB #79A8AFA2 /@VMCibersys
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/af1dd37c/attachment.html 

From victor.medina at cibersys.com  Thu Mar 19 16:47:34 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 19 Mar 2015 09:17:34 -0430
Subject: [Freeswitch-users] Any whay to reload vars.xml?
Message-ID: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>

Hi guys!

Is there any way to reload vars.xml?


-- 

V?ctor E. Medina M.
Software
[image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image:
ve]
BB #79A8AFA2 /@VMCibersys
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/201074d4/attachment.html 

From mike at jerris.com  Thu Mar 19 16:56:20 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 19 Mar 2015 09:56:20 -0400
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
Message-ID: <A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>

reloadxml command will do that.  Now, where those vars are used, that is a different question altogether.

> On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com> wrote:
> 
> Hi guys!
> 
> Is there any way to reload vars.xml?
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/6f42f741/attachment.html 

From gmaruzz at gmail.com  Thu Mar 19 17:07:44 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 19 Mar 2015 15:07:44 +0100
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
	<A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
Message-ID: <CALXCt0rfE8WDJnFfHoDLg3Dv0Rtx_tywL0mdoYfqBB8F8LiJOw@mail.gmail.com>

Ciao Victor,

To expand on what Mike wrote, while is technically true that vars.xml
is reloaded by reloadxml, that vars.xml file contains a lot of
pre-processed variables that are not reloaded when you reloadxml.
All the lines in vars.xml where you see PREPROCESS are *not* reloaded.
You must restart FreeSWITCH to change the PREPROCESSed variables.

-giovanni

On Thu, Mar 19, 2015 at 2:56 PM, Michael Jerris <mike at jerris.com> wrote:
> reloadxml command will do that.  Now, where those vars are used, that is a
> different question altogether.
>
> On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> Hi guys!
>
> Is there any way to reload vars.xml?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From victor.medina at cibersys.com  Thu Mar 19 17:11:28 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 19 Mar 2015 09:41:28 -0430
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
	<A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
Message-ID: <CACXsapNoOa_jGLrRG2DOrirLgx7Dq60S=vYbK7UA_Dyw_p__2w@mail.gmail.com>

so... I better do a sofia channel reload and a roloadxml reload, right?

2015-03-19 9:26 GMT-04:30 Michael Jerris <mike at jerris.com>:

> reloadxml command will do that.  Now, where those vars are used, that is a
> different question altogether.
>
> On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> Hi guys!
>
> Is there any way to reload vars.xml?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

V?ctor E. Medina M.
Software
[image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image:
ve]
BB #79A8AFA2 /@VMCibersys
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/e799c78b/attachment-0001.html 

From gmaruzz at gmail.com  Thu Mar 19 17:10:48 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 19 Mar 2015 15:10:48 +0100
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <CALXCt0rfE8WDJnFfHoDLg3Dv0Rtx_tywL0mdoYfqBB8F8LiJOw@mail.gmail.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
	<A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
	<CALXCt0rfE8WDJnFfHoDLg3Dv0Rtx_tywL0mdoYfqBB8F8LiJOw@mail.gmail.com>
Message-ID: <CALXCt0oGsNsT4n-hwraLmBtu7Zhz43DV1ivZ8dT0jhriHiUm3Q@mail.gmail.com>

(btw, even if you reloaded non preprocessed variables in memory, you
may need to tell the single modules to update their configuration. A
#fscli> reload modulename will do. There are less disruptive ways to
update config in certain modules, notably sofia. Check their
documentation)

On Thu, Mar 19, 2015 at 3:07 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Ciao Victor,
>
> To expand on what Mike wrote, while is technically true that vars.xml
> is reloaded by reloadxml, that vars.xml file contains a lot of
> pre-processed variables that are not reloaded when you reloadxml.
> All the lines in vars.xml where you see PREPROCESS are *not* reloaded.
> You must restart FreeSWITCH to change the PREPROCESSed variables.
>
> -giovanni
>
> On Thu, Mar 19, 2015 at 2:56 PM, Michael Jerris <mike at jerris.com> wrote:
>> reloadxml command will do that.  Now, where those vars are used, that is a
>> different question altogether.
>>
>> On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>> Hi guys!
>>
>> Is there any way to reload vars.xml?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From victor.medina at cibersys.com  Thu Mar 19 17:13:16 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 19 Mar 2015 09:43:16 -0430
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <CALXCt0rfE8WDJnFfHoDLg3Dv0Rtx_tywL0mdoYfqBB8F8LiJOw@mail.gmail.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
	<A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
	<CALXCt0rfE8WDJnFfHoDLg3Dv0Rtx_tywL0mdoYfqBB8F8LiJOw@mail.gmail.com>
Message-ID: <CACXsapPvStY769m+Cz9WviVyiPva-1s-_T6=hMw0PHwpv2MQDw@mail.gmail.com>

OK! I got it... =)

Restart freeswitch

Thanks Giovanni and  Michael.

2015-03-19 9:37 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Ciao Victor,
>
> To expand on what Mike wrote, while is technically true that vars.xml
> is reloaded by reloadxml, that vars.xml file contains a lot of
> pre-processed variables that are not reloaded when you reloadxml.
> All the lines in vars.xml where you see PREPROCESS are *not* reloaded.
> You must restart FreeSWITCH to change the PREPROCESSed variables.
>
> -giovanni
>
> On Thu, Mar 19, 2015 at 2:56 PM, Michael Jerris <mike at jerris.com> wrote:
> > reloadxml command will do that.  Now, where those vars are used, that is
> a
> > different question altogether.
> >
> > On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com>
> > wrote:
> >
> > Hi guys!
> >
> > Is there any way to reload vars.xml?
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

V?ctor E. Medina M.
Software
[image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image:
ve]
BB #79A8AFA2 /@VMCibersys
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/09fd55a3/attachment.html 

From gmaruzz at gmail.com  Thu Mar 19 17:14:11 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 19 Mar 2015 15:14:11 +0100
Subject: [Freeswitch-users] Any whay to reload vars.xml?
In-Reply-To: <CACXsapNoOa_jGLrRG2DOrirLgx7Dq60S=vYbK7UA_Dyw_p__2w@mail.gmail.com>
References: <CACXsapOGWofbM7+G+8fyWcLZ17_V3LTPoobqQfPzgXQtKQPqAQ@mail.gmail.com>
	<A0C39FA4-F071-4B82-A4E1-64B09A39E161@jerris.com>
	<CACXsapNoOa_jGLrRG2DOrirLgx7Dq60S=vYbK7UA_Dyw_p__2w@mail.gmail.com>
Message-ID: <CALXCt0r1A18_Zs8PPzzeOEQaUDo1kaFt91KMUw35oFWxdnqQ_g@mail.gmail.com>

On Thu, Mar 19, 2015 at 3:11 PM, Victor Medina <victor.medina at cibersys.com>
wrote:

> so... I better do a sofia channel reload and a roloadxml reload, right?
>

it depends on what you changed...


>
> 2015-03-19 9:26 GMT-04:30 Michael Jerris <mike at jerris.com>:
>
>> reloadxml command will do that.  Now, where those vars are used, that is
>> a different question altogether.
>>
>> On Mar 19, 2015, at 9:47 AM, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>> Hi guys!
>>
>> Is there any way to reload vars.xml?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> V?ctor E. Medina M.
> Software
> [image: Zoiper Click2Dial]+58424 291 4561[image: ve]
> BB #79A8AFA2 /@VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/ad8562f5/attachment.html 

From sdevoy at bizfocused.com  Thu Mar 19 18:28:28 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 19 Mar 2015 15:28:28 +0000
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
	<BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>

Anyone help with this, PLEASE?  The customer is quite upset.

Thank you,
Sean

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
Sent: Wednesday, March 18, 2015 3:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

Yes, that is what I pass to the intercept call vie the dialplan/database.  Check the log:

62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan: sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> Action intercept(${db(select/esta_call_pickup_uuid/10)})
. . . .
62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)


Sean

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 Brian West
Sent: Wednesday, March 18, 2015 3:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

intercept takes a UUID, not an extension number.

On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Hi All,

I have directed call intercept running via dialplan **\d\d  (two digit extensions).

In this pastebin:  https://pastebin.freeswitch.org/24023

You should see a call come from the IVR into ?MAINLINE? which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>

Can someone who is more log savvy than I review that log and tell me what happened?

Perhaps my dialplan is missing something:

    <extension name="directed-call-pickup">
      <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?

Thanks,
Sean Devoy


_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/8e7d3168/attachment-0001.html 

From ssinyagin at gmail.com  Thu Mar 19 18:43:37 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 19 Mar 2015 16:43:37 +0100
Subject: [Freeswitch-users] Automated testing
Message-ID: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>

I need to build a test suite for a service provider's SIP service
infrastructure. The box will initiate and accept calls and verify that
SIP messages are valid, audio media is two-way, and call control
features are also working.

My plan for the moment is to use FreeSWITCH and to trigger the
outbound calls via ESL, and then use packet capture and Perl modules
for parsing and verifying the SIP messages.

What are other options? I see that SIPp doesn't look too bad, but I
never worked with it.

This page lists mostly outdated and closed-source packages, so not very helpful:
http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing

I'm planning to publish as much of my work as possible as open source.

Any input will be appreciated.

thanks,
stan


From brian at freeswitch.org  Thu Mar 19 18:48:09 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 19 Mar 2015 10:48:09 -0500
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
	<BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <CAEJMVkBu1qaq25eV26j94==9rZ=Gz0SsJ44wTqZZU2tUV+EsXQ@mail.gmail.com>

Sounds like the UUID isn't there that you expected,  so it couldn't pick it
up. I'm guessing the full logs would show that.

On Thu, Mar 19, 2015 at 10:28 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

>  Anyone help with this, PLEASE?  The customer is quite upset.
>
>
>
> Thank you,
>
> Sean
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sean Devoy
> *Sent:* Wednesday, March 18, 2015 3:46 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call intercept.
>
>
>
> Yes, that is what I pass to the intercept call vie the dialplan/database.
> Check the log:
>
>
>
> 62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan:
> sofia/external/20 at fs_esta.bizfocused.com Action
> intercept(${db(select/esta_call_pickup_uuid/10)})
>
> . . . .
>
> 62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE
> sofia/external/20 at fs_esta.bizfocused.com
> intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)
>
>
>
>
>
> Sean
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Brian West
> *Sent:* Wednesday, March 18, 2015 3:39 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call intercept.
>
>
>
> intercept takes a UUID, not an extension number.
>
>
>
> On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>  Hi All,
>
>
>
> I have directed call intercept running via dialplan **\d\d  (two digit
> extensions).
>
>
>
> In this pastebin:  https://pastebin.freeswitch.org/24023
>
>
>
> You should see a call come from the IVR into ?MAINLINE? which dials ext
> 10, then after delay adds other extensions.  In this case, the person from
> extension 10 was at extension 20 and dialed **10.  They report it did not
> work.  I must say I am not clear if it worked or not.  It appears to work,
> but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the
> last dialplan instruction, hanging up.
>
>
>
> Can someone who is more log savvy than I review that log and tell me what
> happened?
>
>
>
> Perhaps my dialplan is missing something:
>
>
>
>     <extension name="directed-call-pickup">
>
>       <!-- picks up the call with db key saved like this:
> <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>
> -->
>
>       <condition field="destination_number" expression="^\*\*(\d\d)$">
>
>         <action application="answer"/>
>
>         <action application="set" data="intercept_unanswered_only=true"/>
>
>         <action application="intercept" data="
> ${db(select/esta_call_pickup_uuid/$1)}"/>
>
>         <action application="sleep" data="2000"/>
>
>       </condition>
>
>     </extension>
>
>
>
> I see in the log the uuid for the appropriate call is in the intercept
> statement.  After intercept, I am missing something to bridge the 2 legs
> together?
>
>
>
> Thanks,
>
> Sean Devoy
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>  *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/5a05d882/attachment-0001.html 

From ssinyagin at gmail.com  Thu Mar 19 18:54:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 19 Mar 2015 16:54:51 +0100
Subject: [Freeswitch-users] Automated testing
In-Reply-To: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
References: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
Message-ID: <CACOYK=hhMSyHKeheEM4PtNFd3zGjoM_fv_pF1uZdpzpw1wCyQg@mail.gmail.com>

just found the Sippy Cup which looks promising:
https://github.com/mojolingo/sippy_cup

On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> I need to build a test suite for a service provider's SIP service
> infrastructure. The box will initiate and accept calls and verify that
> SIP messages are valid, audio media is two-way, and call control
> features are also working.
>
> My plan for the moment is to use FreeSWITCH and to trigger the
> outbound calls via ESL, and then use packet capture and Perl modules
> for parsing and verifying the SIP messages.
>
> What are other options? I see that SIPp doesn't look too bad, but I
> never worked with it.
>
> This page lists mostly outdated and closed-source packages, so not very helpful:
> http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
>
> I'm planning to publish as much of my work as possible as open source.
>
> Any input will be appreciated.
>
> thanks,
> stan


From aronp at guaranteedplus.com  Thu Mar 19 19:10:24 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Thu, 19 Mar 2015 12:10:24 -0400
Subject: [Freeswitch-users] DTMF missing digits
Message-ID: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>

I am experiencing problems with sonus dtmf. I tried setting different
parameters related to the RTP BUGS listed in switch_types.h but it did not
seem to fix the problem.

I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch

*freeswitch sends the following digits to our carrier, and is received by
the other end (also a freeswitch) missing the digits marked in black.*
*pcap shows the missing digits as being part of the previous packet, so two
1's becomes one 1 with the duration sum of two.*

* 1   -    1   -    2   -    1   -    1   -    #*

*What are the relevant parameters I should set?*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/29b182dc/attachment.html 

From orn at arnarson.net  Thu Mar 19 19:26:20 2015
From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=)
Date: Thu, 19 Mar 2015 16:26:20 +0000
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
	<CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
Message-ID: <CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>

Hi,

Thanks for your response.

I installed FS from git on another server, and it still shows exactly the
same behavior. See the two new INVITES for the forked call outbound from FS:
INVITE sip:7712552 at 192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
Max-Forwards: 69
From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
To: <sip:7712552 at 192.168.10.3>
Call-ID: 174ff657-48f7-1233-d586-080027f911ca
CSeq: 73055034 INVITE
Contact: <sip:mod_sofia at 192.168.10.101:5080>
User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
Diversion: <sip:4151500 at 172.25.200.101>
X-FS-Support: update_display,send_info

v=0
o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
s=FreeSWITCH
c=IN IP4 192.168.10.101
t=0 0
m=audio 20950 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

INVITE sip:6595454 at 192.168.10.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
Max-Forwards: 69
From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
To: <sip:6595454 at 192.168.10.3>
Call-ID: 174f7870-48f7-1233-d586-080027f911ca
CSeq: 73055034 INVITE
Contact: <sip:mod_sofia at 192.168.10.101:5080>
User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 251
Diversion: <sip:4151500 at 172.25.200.101>
X-FS-Support: update_display,send_info

v=0
o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
s=FreeSWITCH
c=IN IP4 192.168.10.101
t=0 0
m=audio 31206 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Regards,
?rn

On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:

> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
> would highly recommend you re-test with Master or at the very least 1.4.17
> or 1.4.18 which should be out later today.
>
> 1.2 is not receiving patches, updates or support moving forward, our
> release branch is 1.4.x
>
>
>
>
> On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net> wrote:
>
>> Hello,
>>
>> Not sure whether this belong in the users list or the dev list, but when
>> in doubt; start with users :-)
>>
>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>> Asterisk on the other, and am seeing some strange behavior when calls are
>> being forked on the Asterisk.
>>
>> Call setup is like this:
>> 1. FreeSWITCH receives INVITE from Kamailio
>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>> (each with a unique call-id)
>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>> Call-ID from step 2.
>>
>> This is causing problems with one of the MGWs behind Kamailio, which is
>> seeing multiple INVITEs to different destinations with the same Call-ID.
>>
>> So, firstly, why is FreeSWITCH reusing call-ids?
>>
>> Secondly, how is it matching up the calls? I can't find anything common
>> in the INVITEs, other than the source number and obviously that the IP sent
>> to and received from is the same.
>>
>> I'm not sure if this is intended behavior or not, but is there a way to
>> have FreeSWITCH not do that?
>>
>> Regards,
>> ?rn
>>
>> P.S. Here is the sequence of INVITEs. I also have the console log (for a
>> different call) if needed.
>>
>> *INVITE sent to FreeSWITCH by Kamailio:*
>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>> Record-Route: <sip:172.25.200.101;lr=on>
>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>> Via: SIP/2.0/UDP 172.25.200.121:5080
>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>> Max-Forwards: 16
>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>> To: <sip:5344446 at 172.25.200.101>
>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>> CSeq: 73014324 INVITE
>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 229
>> X-FS-Support: update_display,send_info
>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>> >;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>> s=FreeSWITCH
>> c=IN IP4 172.25.200.121
>> t=0 0
>> m=audio 19026 RTP/AVP 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>>
>>
>> *INVITE sent to Asterisk by FreeSWITCH:*
>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>> Max-Forwards: 15
>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>> To: <sip:5344446 at 172.26.0.62:5060>
>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>> CSeq: 73014353 INVITE
>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>> 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: 223
>> X-FS-Support: update_display,send_info
>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>> >;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>> s=FreeSWITCH
>> c=IN IP4 10.11.12.13
>> t=0 0
>> m=audio 23230 RTP/AVP 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>>
>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>> Max-Forwards: 70
>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>> To: <sip:7712552 at 10.11.12.13>
>> Contact: <sip:4151502 at 172.26.0.62:5060>
>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.8.15-cert2
>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Diversion: <sip:5344446 at 172.26.0.62>
>> Content-Type: application/sdp
>> Content-Length: 312
>>
>> v=0
>> o=root 693576967 693576967 IN IP4 172.26.0.62
>> s=Asterisk PBX 1.8.15-cert2
>> c=IN IP4 172.26.0.62
>> t=0 0
>> m=audio 30440 RTP/AVP 8 0 9 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>
>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>> Max-Forwards: 70
>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>> To: <sip:6595454 at 10.11.12.13>
>> Contact: <sip:4151502 at 172.26.0.62:5060>
>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.8.15-cert2
>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Diversion: <sip:5344446 at 172.26.0.62>
>> Content-Type: application/sdp
>> Content-Length: 310
>>
>> v=0
>> o=root 89056081 89056081 IN IP4 172.26.0.62
>> s=Asterisk PBX 1.8.15-cert2
>> c=IN IP4 172.26.0.62
>> t=0 0
>> m=audio 30708 RTP/AVP 8 0 9 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
>> Max-Forwards: 69
>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>> To: <sip:7712552 at 172.25.200.101>
>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>> CSeq: 73014353 INVITE
>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 209
>> Diversion: <sip:5344446 at 172.25.200.101>
>> X-FS-Support: update_display,send_info
>>
>> v=0
>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>> s=FreeSWITCH
>> c=IN IP4 172.25.200.111
>> t=0 0
>> m=audio 19804 RTP/AVP 8 0 9 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
>> Max-Forwards: 69
>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>> To: <sip:6595454 at 172.25.200.101>
>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>> CSeq: 73014353 INVITE
>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 209
>> Diversion: <sip:5344446 at 172.25.200.101>
>> X-FS-Support: update_display,send_info
>>
>> v=0
>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>> s=FreeSWITCH
>> c=IN IP4 172.25.200.111
>> t=0 0
>> m=audio 31376 RTP/AVP 8 0 9 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/7254c1cc/attachment-0001.html 

From soulofmischief87 at gmail.com  Thu Mar 19 19:42:03 2015
From: soulofmischief87 at gmail.com (Tito Cumpen)
Date: Thu, 19 Mar 2015 12:42:03 -0400
Subject: [Freeswitch-users] Automated testing
In-Reply-To: <CACOYK=hhMSyHKeheEM4PtNFd3zGjoM_fv_pF1uZdpzpw1wCyQg@mail.gmail.com>
References: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
	<CACOYK=hhMSyHKeheEM4PtNFd3zGjoM_fv_pF1uZdpzpw1wCyQg@mail.gmail.com>
Message-ID: <CAPJLwtUeR4zDyM+RJzj4a+mwyUgPOZd5O6WrMzS1uj5+qWgNHA@mail.gmail.com>

Stanislav,

also look into http://sipp.sourceforge.net/doc/reference.html


Thanks,
Tito

On Thu, Mar 19, 2015 at 11:54 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> just found the Sippy Cup which looks promising:
> https://github.com/mojolingo/sippy_cup
>
> On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> > I need to build a test suite for a service provider's SIP service
> > infrastructure. The box will initiate and accept calls and verify that
> > SIP messages are valid, audio media is two-way, and call control
> > features are also working.
> >
> > My plan for the moment is to use FreeSWITCH and to trigger the
> > outbound calls via ESL, and then use packet capture and Perl modules
> > for parsing and verifying the SIP messages.
> >
> > What are other options? I see that SIPp doesn't look too bad, but I
> > never worked with it.
> >
> > This page lists mostly outdated and closed-source packages, so not very
> helpful:
> > http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
> >
> > I'm planning to publish as much of my work as possible as open source.
> >
> > Any input will be appreciated.
> >
> > thanks,
> > stan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/66fc9d7c/attachment.html 

From bote_radio at botecomm.com  Thu Mar 19 19:48:55 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Thu, 19 Mar 2015 12:48:55 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
Message-ID: <019001d06264$973a33d0$c5ae9b70$@botecomm.com>

Wouldn?t it be better to allow the two FreeSWITCH boxes to talk directly to each other for calls between them? Eliminate the middle man and perhaps this and other problems disappear?

 

https://freeswitch.org/confluence/display/FREESWITCH/IMT

 

Bote

 

 

 

 

From: Podrigal, Aron
Sent: Thursday, 19 March, 2015 12:10
Subject: [Freeswitch-users] DTMF missing digits

 

I am experiencing problems with sonus dtmf. I tried setting different parameters related to the RTP BUGS listed in switch_types.h but it did not seem to fix the problem.

 

I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch

 

freeswitch sends the following digits to our carrier, and is received by the other end (also a freeswitch) missing the digits marked in black.

pcap shows the missing digits as being part of the previous packet, so two 1's becomes one 1 with the duration sum of two.

 

 1   -    1   -    2   -    1   -    1   -    #

 

What are the relevant parameters I should set?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/92d86877/attachment.html 

From aronp at guaranteedplus.com  Thu Mar 19 19:56:09 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Thu, 19 Mar 2015 12:56:09 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <019001d06264$973a33d0$c5ae9b70$@botecomm.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<019001d06264$973a33d0$c5ae9b70$@botecomm.com>
Message-ID: <CANJp-yghOEfbjnyWUEpaL4wYMdbVFBzxwa-kqMDjpkrFYORefg@mail.gmail.com>

Of course, but I've made this just to troubleshoot why when sending dtmf
through my carrier it doesnt't work. So I wanted to check how they
interpret my original dtmf.

On Thu, Mar 19, 2015 at 12:48 PM, Bote Man <bote_radio at botecomm.com> wrote:

> Wouldn?t it be better to allow the two FreeSWITCH boxes to talk directly
> to each other for calls between them? Eliminate the middle man and perhaps
> this and other problems disappear?
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/IMT
>
>
>
> Bote
>
>
>
>
>
>
>
>
>
> *From:* Podrigal, Aron
> *Sent:* Thursday, 19 March, 2015 12:10
> *Subject:* [Freeswitch-users] DTMF missing digits
>
>
>
> I am experiencing problems with sonus dtmf. I tried setting different
> parameters related to the RTP BUGS listed in switch_types.h but it did not
> seem to fix the problem.
>
>
>
> I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>
>
>
> *freeswitch sends the following digits to our carrier, and is received by
> the other end (also a freeswitch) missing the digits marked in black.*
>
> *pcap shows the missing digits as being part of the previous packet, so
> two 1's becomes one 1 with the duration sum of two.*
>
>
>
> * 1   -    1   -    2   -    1   -    1   -    #*
>
>
>
> *What are the relevant parameters I should set?*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/78f4e3cd/attachment.html 

From luis.azedo at factorlusitano.com  Thu Mar 19 20:05:27 2015
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Thu, 19 Mar 2015 17:05:27 +0000
Subject: [Freeswitch-users] variable_sip_to_tag not set in Leg A for bridged
	call
Message-ID: <CAEvFzR8A7jQX44Cffs53Qc6cs_8wOdVu4wZ61hH0w_-N5qoq=g@mail.gmail.com>

Hi,

making a bridge call and listening to all events in both legs with esl.
i was expecting that CHANNEL_PROGRESS_MEDIA / CHANNEL_ANSWER (on Leg
A) would include variable_sip_to_tag since there is already this info
(seen in message sent) but we never get this variable to be set in any
event.

we only get the to_tag on leg A when the call is disconnected  or if
we put onhold/unhold to force the creation of event.

during the call (before forcing hold/unhold) i do a uuid_dump of leg A
and i can see the to_tag variable.

the channel variable seems to be set only on nua_i_ack and there seems
to be no event fired after that.

thoughts ?

Best


From cmrienzo at gmail.com  Thu Mar 19 20:12:15 2015
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 19 Mar 2015 13:12:15 -0400
Subject: [Freeswitch-users] Automated testing
In-Reply-To: <CAPJLwtUeR4zDyM+RJzj4a+mwyUgPOZd5O6WrMzS1uj5+qWgNHA@mail.gmail.com>
References: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
	<CACOYK=hhMSyHKeheEM4PtNFd3zGjoM_fv_pF1uZdpzpw1wCyQg@mail.gmail.com>
	<CAPJLwtUeR4zDyM+RJzj4a+mwyUgPOZd5O6WrMzS1uj5+qWgNHA@mail.gmail.com>
Message-ID: <CAD-qRki7uqA2Ef=qm+5pcL8q=pWQBQ4CmdSPMgmwf+dWrRbG7Q@mail.gmail.com>

sippy_cup actually generates test scenarios that run on sipp.

On Thu, Mar 19, 2015 at 12:42 PM, Tito Cumpen <soulofmischief87 at gmail.com>
wrote:

> Stanislav,
>
> also look into http://sipp.sourceforge.net/doc/reference.html
>
>
> Thanks,
> Tito
>
> On Thu, Mar 19, 2015 at 11:54 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> just found the Sippy Cup which looks promising:
>> https://github.com/mojolingo/sippy_cup
>>
>> On Thu, Mar 19, 2015 at 4:43 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>> > I need to build a test suite for a service provider's SIP service
>> > infrastructure. The box will initiate and accept calls and verify that
>> > SIP messages are valid, audio media is two-way, and call control
>> > features are also working.
>> >
>> > My plan for the moment is to use FreeSWITCH and to trigger the
>> > outbound calls via ESL, and then use packet capture and Perl modules
>> > for parsing and verifying the SIP messages.
>> >
>> > What are other options? I see that SIPp doesn't look too bad, but I
>> > never worked with it.
>> >
>> > This page lists mostly outdated and closed-source packages, so not very
>> helpful:
>> > http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
>> >
>> > I'm planning to publish as much of my work as possible as open source.
>> >
>> > Any input will be appreciated.
>> >
>> > thanks,
>> > stan
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/28ebe3e3/attachment-0001.html 

From red.rain.seven at gmail.com  Thu Mar 19 21:36:59 2015
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 19 Mar 2015 11:36:59 -0700
Subject: [Freeswitch-users] Automated testing
In-Reply-To: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
References: <CACOYK=gZ5qqi7KAoSon4aoaqrzeSUJWje3Znp=CRLiHN2ZHJFA@mail.gmail.com>
Message-ID: <CAP_G397j-fe7XzKY-aLE2gLoQPFe_gdJXj0cEAp=iHgZkMrDpw@mail.gmail.com>

Stan,

I started something on Github a while ago utilizing sipp, sox, wireshark,
and pcapsipdump to automate call quality monitoring with Nagios. I was
planning on expanding it to monitor some TTS and ASR service but never got
the cycle to continue. I believe we all needed some automated monitoring in
our infrastructure. I am hoping to get more people involved so we can all
benefit from the fruits of our collaboration.

The documentation is not complete, but you should be able to tell what I am
trying to do in the perl script.

https://github.com/bbhenry/check_voip_call

Thanks,

On Thu, Mar 19, 2015 at 8:43 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> I need to build a test suite for a service provider's SIP service
> infrastructure. The box will initiate and accept calls and verify that
> SIP messages are valid, audio media is two-way, and call control
> features are also working.
>
> My plan for the moment is to use FreeSWITCH and to trigger the
> outbound calls via ESL, and then use packet capture and Perl modules
> for parsing and verifying the SIP messages.
>
> What are other options? I see that SIPp doesn't look too bad, but I
> never worked with it.
>
> This page lists mostly outdated and closed-source packages, so not very
> helpful:
> http://www.voip-info.org/wiki/view/Protocol+Verification+and+Testing
>
> I'm planning to publish as much of my work as possible as open source.
>
> Any input will be appreciated.
>
> thanks,
> stan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/dbe53064/attachment.html 

From mike at jerris.com  Thu Mar 19 21:58:23 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 19 Mar 2015 14:58:23 -0400
Subject: [Freeswitch-users] variable_sip_to_tag not set in Leg A for
	bridged call
In-Reply-To: <CAEvFzR8A7jQX44Cffs53Qc6cs_8wOdVu4wZ61hH0w_-N5qoq=g@mail.gmail.com>
References: <CAEvFzR8A7jQX44Cffs53Qc6cs_8wOdVu4wZ61hH0w_-N5qoq=g@mail.gmail.com>
Message-ID: <2E87E1ED-4111-4309-AACE-0899EBB501B7@jerris.com>

Not sure there is another place we can reliably set it, particularly in transfer and 3pcc scenarios.  If you can come up with a modification that gets it in earlier in the scenarios it can, and confirm it doesn't break anything, toss us the change via a pull request.

Mike

> On Mar 19, 2015, at 1:05 PM, Luis Azedo <luis.azedo at factorlusitano.com> wrote:
> 
> Hi,
> 
> making a bridge call and listening to all events in both legs with esl.
> i was expecting that CHANNEL_PROGRESS_MEDIA / CHANNEL_ANSWER (on Leg
> A) would include variable_sip_to_tag since there is already this info
> (seen in message sent) but we never get this variable to be set in any
> event.
> 
> we only get the to_tag on leg A when the call is disconnected  or if
> we put onhold/unhold to force the creation of event.
> 
> during the call (before forcing hold/unhold) i do a uuid_dump of leg A
> and i can see the to_tag variable.
> 
> the channel variable seems to be set only on nua_i_ack and there seems
> to be no event fired after that.
> 
> thoughts ?
> 
> Best


From brian at freeswitch.org  Thu Mar 19 22:11:46 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 19 Mar 2015 14:11:46 -0500
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
	<CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
	<CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>
Message-ID: <CAEJMVkDWNRaHbXEF=p_90Q590M=VBaZvWXbM3MqcqxT+vzwH5A@mail.gmail.com>

Looks like an invite that didnt' get a response.  You sure dude?

On Thu, Mar 19, 2015 at 11:26 AM, ?rn Arnarson <orn at arnarson.net> wrote:

> Hi,
>
> Thanks for your response.
>
> I installed FS from git on another server, and it still shows exactly the
> same behavior. See the two new INVITES for the forked call outbound from FS:
> INVITE sip:7712552 at 192.168.10.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
> Max-Forwards: 69
> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
> To: <sip:7712552 at 192.168.10.3>
> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
> CSeq: 73055034 INVITE
> Contact: <sip:mod_sofia at 192.168.10.101:5080>
> User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 251
> Diversion: <sip:4151500 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
> s=FreeSWITCH
> c=IN IP4 192.168.10.101
> t=0 0
> m=audio 20950 RTP/AVP 8 0 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> INVITE sip:6595454 at 192.168.10.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
> Max-Forwards: 69
> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
> To: <sip:6595454 at 192.168.10.3>
> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
> CSeq: 73055034 INVITE
> Contact: <sip:mod_sofia at 192.168.10.101:5080>
> User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 251
> Diversion: <sip:4151500 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
> s=FreeSWITCH
> c=IN IP4 192.168.10.101
> t=0 0
> m=audio 31206 RTP/AVP 8 0 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> Regards,
> ?rn
>
> On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
>> would highly recommend you re-test with Master or at the very least 1.4.17
>> or 1.4.18 which should be out later today.
>>
>> 1.2 is not receiving patches, updates or support moving forward, our
>> release branch is 1.4.x
>>
>>
>>
>>
>> On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net> wrote:
>>
>>> Hello,
>>>
>>> Not sure whether this belong in the users list or the dev list, but when
>>> in doubt; start with users :-)
>>>
>>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>>> Asterisk on the other, and am seeing some strange behavior when calls are
>>> being forked on the Asterisk.
>>>
>>> Call setup is like this:
>>> 1. FreeSWITCH receives INVITE from Kamailio
>>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>>> (each with a unique call-id)
>>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>>> Call-ID from step 2.
>>>
>>> This is causing problems with one of the MGWs behind Kamailio, which is
>>> seeing multiple INVITEs to different destinations with the same Call-ID.
>>>
>>> So, firstly, why is FreeSWITCH reusing call-ids?
>>>
>>> Secondly, how is it matching up the calls? I can't find anything common
>>> in the INVITEs, other than the source number and obviously that the IP sent
>>> to and received from is the same.
>>>
>>> I'm not sure if this is intended behavior or not, but is there a way to
>>> have FreeSWITCH not do that?
>>>
>>> Regards,
>>> ?rn
>>>
>>> P.S. Here is the sequence of INVITEs. I also have the console log (for a
>>> different call) if needed.
>>>
>>> *INVITE sent to FreeSWITCH by Kamailio:*
>>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>>> Record-Route: <sip:172.25.200.101;lr=on>
>>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>>> Via: SIP/2.0/UDP 172.25.200.121:5080
>>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>>> Max-Forwards: 16
>>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>>> To: <sip:5344446 at 172.25.200.101>
>>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>>> CSeq: 73014324 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 229
>>> X-FS-Support: update_display,send_info
>>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>>> >;party=calling;screen=yes;privacy=off
>>>
>>> v=0
>>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.121
>>> t=0 0
>>> m=audio 19026 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>>
>>> *INVITE sent to Asterisk by FreeSWITCH:*
>>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>>> Max-Forwards: 15
>>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>>> To: <sip:5344446 at 172.26.0.62:5060>
>>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> 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: 223
>>> X-FS-Support: update_display,send_info
>>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>>> >;party=calling;screen=yes;privacy=off
>>>
>>> v=0
>>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>>> s=FreeSWITCH
>>> c=IN IP4 10.11.12.13
>>> t=0 0
>>> m=audio 23230 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>>> Max-Forwards: 70
>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>>> To: <sip:7712552 at 10.11.12.13>
>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Diversion: <sip:5344446 at 172.26.0.62>
>>> Content-Type: application/sdp
>>> Content-Length: 312
>>>
>>> v=0
>>> o=root 693576967 693576967 IN IP4 172.26.0.62
>>> s=Asterisk PBX 1.8.15-cert2
>>> c=IN IP4 172.26.0.62
>>> t=0 0
>>> m=audio 30440 RTP/AVP 8 0 9 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>>
>>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>>> Max-Forwards: 70
>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>>> To: <sip:6595454 at 10.11.12.13>
>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Diversion: <sip:5344446 at 172.26.0.62>
>>> Content-Type: application/sdp
>>> Content-Length: 310
>>>
>>> v=0
>>> o=root 89056081 89056081 IN IP4 172.26.0.62
>>> s=Asterisk PBX 1.8.15-cert2
>>> c=IN IP4 172.26.0.62
>>> t=0 0
>>> m=audio 30708 RTP/AVP 8 0 9 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
>>> Max-Forwards: 69
>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>>> To: <sip:7712552 at 172.25.200.101>
>>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 209
>>> Diversion: <sip:5344446 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.111
>>> t=0 0
>>> m=audio 19804 RTP/AVP 8 0 9 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
>>> Max-Forwards: 69
>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>>> To: <sip:6595454 at 172.25.200.101>
>>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 209
>>> Diversion: <sip:5344446 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.111
>>> t=0 0
>>> m=audio 31376 RTP/AVP 8 0 9 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/6f0eede2/attachment-0001.html 

From grcamauer at gmail.com  Thu Mar 19 22:49:36 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 19 Mar 2015 16:49:36 -0300
Subject: [Freeswitch-users] Problem with make current
In-Reply-To: <CAEJMVkDrnWw9Vqg=5i=UJ7xjLfCcaJDUa2zL0Kr2SuySEK5sxQ@mail.gmail.com>
References: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>
	<CAEJMVkDrnWw9Vqg=5i=UJ7xjLfCcaJDUa2zL0Kr2SuySEK5sxQ@mail.gmail.com>
Message-ID: <CAPNTnnoF8S2dFrs59YQivOyEG4kKAr6ENxrDOomEDfWrdr951A@mail.gmail.com>

I am in Argentina.  Never had a problem before.
If I put https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs in a
browser it returns a 400 - Bad Request.

Guillermo

On Wed, Mar 18, 2015 at 9:08 PM, Brian West <brian at freeswitch.org> wrote:

> What country are you accessing from?  That should only be doing a git
> pull, does that fail too?
>
> On Wed, Mar 18, 2015 at 2:49 PM, Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> I am having problems trying to update my Freeswitch instances.  MAKE
>> CURRENT ends with:
>>
>> make[1]: Entering directory `/usr/src/freeswitch'
>> Pulling updates...
>> error: Failed connect to freeswitch.org:443; Connection timed out while
>> accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs
>> fatal: HTTP request failed
>> make[1]: *** [update] Error 1
>> make[1]: Leaving directory `/usr/src/freeswitch'
>>
>>
>> Are the stash servers down?
>>
>>
>> --
>> Guillermo Ruiz Camauer
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/5389db32/attachment.html 

From alhakeem at gmail.com  Thu Mar 19 22:54:19 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Thu, 19 Mar 2015 19:54:19 -0000
Subject: [Freeswitch-users] XML and HTTP engines or libraries
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>

Hello,
Can someone point me to the XML/HTTP  engines or libraries used in Freeswitch ?
Thanks,
Abdul Hakeem




From brian at freeswitch.org  Thu Mar 19 22:56:57 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 19 Mar 2015 14:56:57 -0500
Subject: [Freeswitch-users] Problem with make current
In-Reply-To: <CAPNTnnoF8S2dFrs59YQivOyEG4kKAr6ENxrDOomEDfWrdr951A@mail.gmail.com>
References: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>
	<CAEJMVkDrnWw9Vqg=5i=UJ7xjLfCcaJDUa2zL0Kr2SuySEK5sxQ@mail.gmail.com>
	<CAPNTnnoF8S2dFrs59YQivOyEG4kKAr6ENxrDOomEDfWrdr951A@mail.gmail.com>
Message-ID: <CAEJMVkCOVk-UKUPDCJxYKpBamzHZTfmU9pkNt27ca2mU3oiZxA@mail.gmail.com>

You must be behind a nasty transparent proxy.

On Thu, Mar 19, 2015 at 2:49 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> I am in Argentina.  Never had a problem before.
> If I put https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs in
> a browser it returns a 400 - Bad Request.
>
> Guillermo
>
> On Wed, Mar 18, 2015 at 9:08 PM, Brian West <brian at freeswitch.org> wrote:
>
>> What country are you accessing from?  That should only be doing a git
>> pull, does that fail too?
>>
>> On Wed, Mar 18, 2015 at 2:49 PM, Guillermo Ruiz Camauer <
>> grcamauer at gmail.com> wrote:
>>
>>> I am having problems trying to update my Freeswitch instances.  MAKE
>>> CURRENT ends with:
>>>
>>> make[1]: Entering directory `/usr/src/freeswitch'
>>> Pulling updates...
>>> error: Failed connect to freeswitch.org:443; Connection timed out while
>>> accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs
>>> fatal: HTTP request failed
>>> make[1]: *** [update] Error 1
>>> make[1]: Leaving directory `/usr/src/freeswitch'
>>>
>>>
>>> Are the stash servers down?
>>>
>>>
>>> --
>>> Guillermo Ruiz Camauer
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/f72b2c19/attachment.html 

From brian at freeswitch.org  Thu Mar 19 22:57:30 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 19 Mar 2015 14:57:30 -0500
Subject: [Freeswitch-users] XML and HTTP engines or libraries
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>
Message-ID: <CAEJMVkAuxpPS0c3DskAPuY1VbPxo3xof79Dp7LXm3n0Htfav4w@mail.gmail.com>

see xml_curl, confluence.freeswitch.org or wiki.freeswitch.org

On Thu, Mar 19, 2015 at 2:54 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
> Can someone point me to the XML/HTTP  engines or libraries used in
> Freeswitch ?
> Thanks,
> Abdul Hakeem
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/d1c8e3d2/attachment-0001.html 

From grcamauer at gmail.com  Thu Mar 19 23:23:17 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 19 Mar 2015 17:23:17 -0300
Subject: [Freeswitch-users] Problem with make current
In-Reply-To: <CAEJMVkCOVk-UKUPDCJxYKpBamzHZTfmU9pkNt27ca2mU3oiZxA@mail.gmail.com>
References: <CAPNTnnqL0S-x+u_TEHXQOCZfMHDB_8V3DLNqPQ4vuR=EcpH0NQ@mail.gmail.com>
	<CAEJMVkDrnWw9Vqg=5i=UJ7xjLfCcaJDUa2zL0Kr2SuySEK5sxQ@mail.gmail.com>
	<CAPNTnnoF8S2dFrs59YQivOyEG4kKAr6ENxrDOomEDfWrdr951A@mail.gmail.com>
	<CAEJMVkCOVk-UKUPDCJxYKpBamzHZTfmU9pkNt27ca2mU3oiZxA@mail.gmail.com>
Message-ID: <CAPNTnnqZLN2NKgx6CPgjpHjNW1C-o7bw0Wu75-GkUNKSz2GC8A@mail.gmail.com>

Same proxy I have had for years.  Anything I can do?

Guillermo

On Thu, Mar 19, 2015 at 4:56 PM, Brian West <brian at freeswitch.org> wrote:

> You must be behind a nasty transparent proxy.
>
> On Thu, Mar 19, 2015 at 2:49 PM, Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>> I am in Argentina.  Never had a problem before.
>> If I put https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs in
>> a browser it returns a 400 - Bad Request.
>>
>> Guillermo
>>
>> On Wed, Mar 18, 2015 at 9:08 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> What country are you accessing from?  That should only be doing a git
>>> pull, does that fail too?
>>>
>>> On Wed, Mar 18, 2015 at 2:49 PM, Guillermo Ruiz Camauer <
>>> grcamauer at gmail.com> wrote:
>>>
>>>> I am having problems trying to update my Freeswitch instances.  MAKE
>>>> CURRENT ends with:
>>>>
>>>> make[1]: Entering directory `/usr/src/freeswitch'
>>>> Pulling updates...
>>>> error: Failed connect to freeswitch.org:443; Connection timed out
>>>> while accessing
>>>> https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs
>>>> fatal: HTTP request failed
>>>> make[1]: *** [update] Error 1
>>>> make[1]: Leaving directory `/usr/src/freeswitch'
>>>>
>>>>
>>>> Are the stash servers down?
>>>>
>>>>
>>>> --
>>>> Guillermo Ruiz Camauer
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/17d46fb9/attachment.html 

From pjintheusa at gmail.com  Fri Mar 20 01:08:11 2015
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 19 Mar 2015 18:08:11 -0400
Subject: [Freeswitch-users] start_dtmf when no RFC 2833
In-Reply-To: <CAGK4cYyNxr9WVwmBP0=Wv4QkCJ2-7WGBafgSvWahbhH3MATb4g@mail.gmail.com>
References: <CAGK4cYwTOv4RwWePOFCm6ASuamg4MmGPCnYfWbLCVH5WOAUyTA@mail.gmail.com>
	<CAD-qRkjmWAkhgo4pL1NoOW2jAVfnnFqAz32b0TFf2zfHBmTZMw@mail.gmail.com>
	<CAGK4cYw0xP=+=fQO7JQUtiJTASUehrV3ADQbQx5nWUD0cdz7=A@mail.gmail.com>
	<CAFiqYumQo2V=b0eDwNXfzm0EU9EZ=qN+m2_6HtfTWzB4=Hk-+w@mail.gmail.com>
	<CAGK4cYyNxr9WVwmBP0=Wv4QkCJ2-7WGBafgSvWahbhH3MATb4g@mail.gmail.com>
Message-ID: <CAGK4cYxNo-9zVSgNQKUshcMwbDzQr-oqsegOYpNJLa1PK8gYLA@mail.gmail.com>

Just for others looking later, here is the final code I came up with, with
rtp_payload extracted.

if session:getVariable("switch_r_sdp") ~= nil then
 local s = '[[' .. session:getVariable("switch_r_sdp") ..']]'
 if string.find(s, "a=rtpmap:(%d+)%stelephone%-event%/8000") ~= nil then

freeswitch.consoleLog("debug", "NO NEED TO START starting
spandsp_start_dtmf\n")
 local payload = string.match(s,"a=rtpmap:(%d+)%stelephone%-event%/8000")
 if payload  ~= nil then
freeswitch.consoleLog("debug", "rtp_payload_number = " .. payload  .."\n")
end

else

freeswitch.consoleLog("info", "starting spandsp_start_dtmf\n")
session:execute("spandsp_start_dtmf")
session:sleep(2000)
 end

end


Cheers

Phil




>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150319/89dc8f44/attachment.html 

From krice at freeswitch.org  Fri Mar 20 04:54:05 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Mar 2015 01:54:05 +0000
Subject: [Freeswitch-users] ClueCon is Coming. Start Getting Ready!
Message-ID: <550b7dbd8544d_3a0d79d33082641@resque-worker-ip-10-153-136-71.mail>

New Post on freeswitch.org from anthm
check it out at http://ift.tt/1x6SG9o
ClueCon is Coming. Start Getting Ready!

ClueCon was a big hit last year but don?t take it from us. Here are some testimonials from 2014.
Don?t forget the call for speakers is now live. Visit http://ift.tt/1FBM7gN and reserve a slot!

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/1b2897f2/attachment.html 

From orn at arnarson.net  Fri Mar 20 12:46:53 2015
From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=)
Date: Fri, 20 Mar 2015 09:46:53 +0000
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAEJMVkDWNRaHbXEF=p_90Q590M=VBaZvWXbM3MqcqxT+vzwH5A@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
	<CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
	<CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>
	<CAEJMVkDWNRaHbXEF=p_90Q590M=VBaZvWXbM3MqcqxT+vzwH5A@mail.gmail.com>
Message-ID: <CAG_7Y1QjzjqEayv+PGrA7R7DdHrr_LgP06NKg1Co25zj9GS8Ag@mail.gmail.com>

Look at the URI -- it's different in each case. These were received within
1 ms from each other as well. This is not a retransmission.
Do you think this is a bug? Should I mail this to the dev list?

On Thu, Mar 19, 2015 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:

> Looks like an invite that didnt' get a response.  You sure dude?
>
> On Thu, Mar 19, 2015 at 11:26 AM, ?rn Arnarson <orn at arnarson.net> wrote:
>
>> Hi,
>>
>> Thanks for your response.
>>
>> I installed FS from git on another server, and it still shows exactly the
>> same behavior. See the two new INVITES for the forked call outbound from FS:
>> INVITE sip:7712552 at 192.168.10.3 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
>> Max-Forwards: 69
>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
>> To: <sip:7712552 at 192.168.10.3>
>> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
>> CSeq: 73055034 INVITE
>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 251
>> Diversion: <sip:4151500 at 172.25.200.101>
>> X-FS-Support: update_display,send_info
>>
>> v=0
>> o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
>> s=FreeSWITCH
>> c=IN IP4 192.168.10.101
>> t=0 0
>> m=audio 20950 RTP/AVP 8 0 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> INVITE sip:6595454 at 192.168.10.3 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
>> Max-Forwards: 69
>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
>> To: <sip:6595454 at 192.168.10.3>
>> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
>> CSeq: 73055034 INVITE
>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 251
>> Diversion: <sip:4151500 at 172.25.200.101>
>> X-FS-Support: update_display,send_info
>>
>> v=0
>> o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
>> s=FreeSWITCH
>> c=IN IP4 192.168.10.101
>> t=0 0
>> m=audio 31206 RTP/AVP 8 0 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> Regards,
>> ?rn
>>
>> On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
>>> would highly recommend you re-test with Master or at the very least 1.4.17
>>> or 1.4.18 which should be out later today.
>>>
>>> 1.2 is not receiving patches, updates or support moving forward, our
>>> release branch is 1.4.x
>>>
>>>
>>>
>>>
>>> On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net> wrote:
>>>
>>>> Hello,
>>>>
>>>> Not sure whether this belong in the users list or the dev list, but
>>>> when in doubt; start with users :-)
>>>>
>>>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>>>> Asterisk on the other, and am seeing some strange behavior when calls are
>>>> being forked on the Asterisk.
>>>>
>>>> Call setup is like this:
>>>> 1. FreeSWITCH receives INVITE from Kamailio
>>>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>>>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>>>> (each with a unique call-id)
>>>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>>>> Call-ID from step 2.
>>>>
>>>> This is causing problems with one of the MGWs behind Kamailio, which is
>>>> seeing multiple INVITEs to different destinations with the same Call-ID.
>>>>
>>>> So, firstly, why is FreeSWITCH reusing call-ids?
>>>>
>>>> Secondly, how is it matching up the calls? I can't find anything common
>>>> in the INVITEs, other than the source number and obviously that the IP sent
>>>> to and received from is the same.
>>>>
>>>> I'm not sure if this is intended behavior or not, but is there a way to
>>>> have FreeSWITCH not do that?
>>>>
>>>> Regards,
>>>> ?rn
>>>>
>>>> P.S. Here is the sequence of INVITEs. I also have the console log (for
>>>> a different call) if needed.
>>>>
>>>> *INVITE sent to FreeSWITCH by Kamailio:*
>>>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>>>> Record-Route: <sip:172.25.200.101;lr=on>
>>>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>>>> Via: SIP/2.0/UDP 172.25.200.121:5080
>>>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>>>> Max-Forwards: 16
>>>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>>>> To: <sip:5344446 at 172.25.200.101>
>>>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>>>> CSeq: 73014324 INVITE
>>>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 229
>>>> X-FS-Support: update_display,send_info
>>>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>>>> >;party=calling;screen=yes;privacy=off
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>>>> s=FreeSWITCH
>>>> c=IN IP4 172.25.200.121
>>>> t=0 0
>>>> m=audio 19026 RTP/AVP 8 101
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>>
>>>> *INVITE sent to Asterisk by FreeSWITCH:*
>>>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>>>> Max-Forwards: 15
>>>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>>>> To: <sip:5344446 at 172.26.0.62:5060>
>>>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>>>> CSeq: 73014353 INVITE
>>>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>> 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: 223
>>>> X-FS-Support: update_display,send_info
>>>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>>>> >;party=calling;screen=yes;privacy=off
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.11.12.13
>>>> t=0 0
>>>> m=audio 23230 RTP/AVP 8 101
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>>>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>>>> Max-Forwards: 70
>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>>>> To: <sip:7712552 at 10.11.12.13>
>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>>>> CSeq: 102 INVITE
>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>> INFO, PUBLISH
>>>> Supported: replaces, timer
>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>> Content-Type: application/sdp
>>>> Content-Length: 312
>>>>
>>>> v=0
>>>> o=root 693576967 693576967 IN IP4 172.26.0.62
>>>> s=Asterisk PBX 1.8.15-cert2
>>>> c=IN IP4 172.26.0.62
>>>> t=0 0
>>>> m=audio 30440 RTP/AVP 8 0 9 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>>
>>>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>>>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>>>> Max-Forwards: 70
>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>>>> To: <sip:6595454 at 10.11.12.13>
>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>>>> CSeq: 102 INVITE
>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>> INFO, PUBLISH
>>>> Supported: replaces, timer
>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>> Content-Type: application/sdp
>>>> Content-Length: 310
>>>>
>>>> v=0
>>>> o=root 89056081 89056081 IN IP4 172.26.0.62
>>>> s=Asterisk PBX 1.8.15-cert2
>>>> c=IN IP4 172.26.0.62
>>>> t=0 0
>>>> m=audio 30708 RTP/AVP 8 0 9 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>>>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
>>>> Max-Forwards: 69
>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>>>> To: <sip:7712552 at 172.25.200.101>
>>>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>>>> CSeq: 73014353 INVITE
>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 209
>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>> X-FS-Support: update_display,send_info
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>>>> s=FreeSWITCH
>>>> c=IN IP4 172.25.200.111
>>>> t=0 0
>>>> m=audio 19804 RTP/AVP 8 0 9 101 13
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by
>>>> Asterisk):*
>>>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
>>>> Max-Forwards: 69
>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>>>> To: <sip:6595454 at 172.25.200.101>
>>>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>>>> CSeq: 73014353 INVITE
>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 209
>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>> X-FS-Support: update_display,send_info
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>>>> s=FreeSWITCH
>>>> c=IN IP4 172.25.200.111
>>>> t=0 0
>>>> m=audio 31376 RTP/AVP 8 0 9 101 13
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/a83080e8/attachment-0001.html 

From ssinyagin at gmail.com  Fri Mar 20 13:33:07 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 20 Mar 2015 11:33:07 +0100
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
Message-ID: <CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>

how do you send DTMF to sonus? info or rfc2833?


On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
<aronp at guaranteedplus.com> wrote:
> I am experiencing problems with sonus dtmf. I tried setting different
> parameters related to the RTP BUGS listed in switch_types.h but it did not
> seem to fix the problem.
>
> I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>
> freeswitch sends the following digits to our carrier, and is received by the
> other end (also a freeswitch) missing the digits marked in black.
> pcap shows the missing digits as being part of the previous packet, so two
> 1's becomes one 1 with the duration sum of two.
>
>  1   -    1   -    2   -    1   -    1   -    #
>
> What are the relevant parameters I should set?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Mar 20 14:45:31 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 20 Mar 2015 11:45:31 +0000
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAG_7Y1QjzjqEayv+PGrA7R7DdHrr_LgP06NKg1Co25zj9GS8Ag@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
	<CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
	<CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>
	<CAEJMVkDWNRaHbXEF=p_90Q590M=VBaZvWXbM3MqcqxT+vzwH5A@mail.gmail.com>
	<CAG_7Y1QjzjqEayv+PGrA7R7DdHrr_LgP06NKg1Co25zj9GS8Ag@mail.gmail.com>
Message-ID: <CAFiqYundS=a7x6z1RCTP40fjtQGGhMXtQndF+mCbt5zYTepDqQ@mail.gmail.com>

The Call-IDs are different on each INVITE. Not the same as you claim.

Call-ID: 174ff657-48f7-1233-d586-080027f911ca
Call-ID: 174f7870-48f7-1233-d586-080027f911ca

The first part varies even though the others are the same.

On 20 March 2015 at 09:46, ?rn Arnarson <orn at arnarson.net> wrote:

> Look at the URI -- it's different in each case. These were received within
> 1 ms from each other as well. This is not a retransmission.
> Do you think this is a bug? Should I mail this to the dev list?
>
> On Thu, Mar 19, 2015 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Looks like an invite that didnt' get a response.  You sure dude?
>>
>> On Thu, Mar 19, 2015 at 11:26 AM, ?rn Arnarson <orn at arnarson.net> wrote:
>>
>>> Hi,
>>>
>>> Thanks for your response.
>>>
>>> I installed FS from git on another server, and it still shows exactly
>>> the same behavior. See the two new INVITES for the forked call outbound
>>> from FS:
>>> INVITE sip:7712552 at 192.168.10.3 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
>>> Max-Forwards: 69
>>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
>>> To: <sip:7712552 at 192.168.10.3>
>>> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
>>> CSeq: 73055034 INVITE
>>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 251
>>> Diversion: <sip:4151500 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.10.101
>>> t=0 0
>>> m=audio 20950 RTP/AVP 8 0 101 13
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> INVITE sip:6595454 at 192.168.10.3 SIP/2.0
>>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
>>> Max-Forwards: 69
>>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
>>> To: <sip:6595454 at 192.168.10.3>
>>> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
>>> CSeq: 73055034 INVITE
>>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 251
>>> Diversion: <sip:4151500 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.10.101
>>> t=0 0
>>> m=audio 31206 RTP/AVP 8 0 101 13
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> Regards,
>>> ?rn
>>>
>>> On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
>>>> would highly recommend you re-test with Master or at the very least 1.4.17
>>>> or 1.4.18 which should be out later today.
>>>>
>>>> 1.2 is not receiving patches, updates or support moving forward, our
>>>> release branch is 1.4.x
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Not sure whether this belong in the users list or the dev list, but
>>>>> when in doubt; start with users :-)
>>>>>
>>>>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>>>>> Asterisk on the other, and am seeing some strange behavior when calls are
>>>>> being forked on the Asterisk.
>>>>>
>>>>> Call setup is like this:
>>>>> 1. FreeSWITCH receives INVITE from Kamailio
>>>>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>>>>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>>>>> (each with a unique call-id)
>>>>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>>>>> Call-ID from step 2.
>>>>>
>>>>> This is causing problems with one of the MGWs behind Kamailio, which
>>>>> is seeing multiple INVITEs to different destinations with the same Call-ID.
>>>>>
>>>>> So, firstly, why is FreeSWITCH reusing call-ids?
>>>>>
>>>>> Secondly, how is it matching up the calls? I can't find anything
>>>>> common in the INVITEs, other than the source number and obviously that the
>>>>> IP sent to and received from is the same.
>>>>>
>>>>> I'm not sure if this is intended behavior or not, but is there a way
>>>>> to have FreeSWITCH not do that?
>>>>>
>>>>> Regards,
>>>>> ?rn
>>>>>
>>>>> P.S. Here is the sequence of INVITEs. I also have the console log (for
>>>>> a different call) if needed.
>>>>>
>>>>> *INVITE sent to FreeSWITCH by Kamailio:*
>>>>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>>>>> Record-Route: <sip:172.25.200.101;lr=on>
>>>>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>>>>> Via: SIP/2.0/UDP 172.25.200.121:5080
>>>>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>>>>> Max-Forwards: 16
>>>>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>>>>> To: <sip:5344446 at 172.25.200.101>
>>>>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>>>>> CSeq: 73014324 INVITE
>>>>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY
>>>>> Supported: timer, precondition, path, replaces
>>>>> Allow-Events: talk, hold, conference, refer
>>>>> Content-Type: application/sdp
>>>>> Content-Disposition: session
>>>>> Content-Length: 229
>>>>> X-FS-Support: update_display,send_info
>>>>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>>>>> >;party=calling;screen=yes;privacy=off
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 172.25.200.121
>>>>> t=0 0
>>>>> m=audio 19026 RTP/AVP 8 101
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>>
>>>>> *INVITE sent to Asterisk by FreeSWITCH:*
>>>>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>>>>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>>>>> Max-Forwards: 15
>>>>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>>>>> To: <sip:5344446 at 172.26.0.62:5060>
>>>>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>>>>> CSeq: 73014353 INVITE
>>>>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>> 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: 223
>>>>> X-FS-Support: update_display,send_info
>>>>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>>>>> >;party=calling;screen=yes;privacy=off
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 10.11.12.13
>>>>> t=0 0
>>>>> m=audio 23230 RTP/AVP 8 101
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>>>>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>>>>> Max-Forwards: 70
>>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>>>>> To: <sip:7712552 at 10.11.12.13>
>>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>>>>> CSeq: 102 INVITE
>>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>> INFO, PUBLISH
>>>>> Supported: replaces, timer
>>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 312
>>>>>
>>>>> v=0
>>>>> o=root 693576967 693576967 IN IP4 172.26.0.62
>>>>> s=Asterisk PBX 1.8.15-cert2
>>>>> c=IN IP4 172.26.0.62
>>>>> t=0 0
>>>>> m=audio 30440 RTP/AVP 8 0 9 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:9 G722/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>>
>>>>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>>>>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>>>>> Max-Forwards: 70
>>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>>>>> To: <sip:6595454 at 10.11.12.13>
>>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>>>>> CSeq: 102 INVITE
>>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>> INFO, PUBLISH
>>>>> Supported: replaces, timer
>>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>>> Content-Type: application/sdp
>>>>> Content-Length: 310
>>>>>
>>>>> v=0
>>>>> o=root 89056081 89056081 IN IP4 172.26.0.62
>>>>> s=Asterisk PBX 1.8.15-cert2
>>>>> c=IN IP4 172.26.0.62
>>>>> t=0 0
>>>>> m=audio 30708 RTP/AVP 8 0 9 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:9 G722/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>>
>>>>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by
>>>>> Asterisk):*
>>>>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
>>>>> Max-Forwards: 69
>>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>>>>> To: <sip:7712552 at 172.25.200.101>
>>>>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>>>>> CSeq: 73014353 INVITE
>>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY
>>>>> Supported: timer, precondition, path, replaces
>>>>> Allow-Events: talk, hold, conference, refer
>>>>> Content-Type: application/sdp
>>>>> Content-Disposition: session
>>>>> Content-Length: 209
>>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>>> X-FS-Support: update_display,send_info
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 172.25.200.111
>>>>> t=0 0
>>>>> m=audio 19804 RTP/AVP 8 0 9 101 13
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>>
>>>>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by
>>>>> Asterisk):*
>>>>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>>>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
>>>>> Max-Forwards: 69
>>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>>>>> To: <sip:6595454 at 172.25.200.101>
>>>>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>>>>> CSeq: 73014353 INVITE
>>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY
>>>>> Supported: timer, precondition, path, replaces
>>>>> Allow-Events: talk, hold, conference, refer
>>>>> Content-Type: application/sdp
>>>>> Content-Disposition: session
>>>>> Content-Length: 209
>>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>>> X-FS-Support: update_display,send_info
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 172.25.200.111
>>>>> t=0 0
>>>>> m=audio 31376 RTP/AVP 8 0 9 101 13
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/23b8d1bf/attachment-0001.html 

From hardyanto.donny at gmail.com  Fri Mar 20 15:11:53 2015
From: hardyanto.donny at gmail.com (Donny Hardyanto)
Date: Fri, 20 Mar 2015 19:11:53 +0700
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
Message-ID: <CAEhEJ-VeWrOSqJ7XwjEn=vGY5ztoYtyQBtpHqHxaZ4tU2y9LDA@mail.gmail.com>

I have done integration with sonus. I am sending rfc2833. Usually the dtmf
suppression and fax detection setting on the sonus is the problem.

donny

On Friday, March 20, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> how do you send DTMF to sonus? info or rfc2833?
>
>
> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
> <aronp at guaranteedplus.com <javascript:;>> wrote:
> > I am experiencing problems with sonus dtmf. I tried setting different
> > parameters related to the RTP BUGS listed in switch_types.h but it did
> not
> > seem to fix the problem.
> >
> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
> >
> > freeswitch sends the following digits to our carrier, and is received by
> the
> > other end (also a freeswitch) missing the digits marked in black.
> > pcap shows the missing digits as being part of the previous packet, so
> two
> > 1's becomes one 1 with the duration sum of two.
> >
> >  1   -    1   -    2   -    1   -    1   -    #
> >
> > What are the relevant parameters I should set?
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.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 <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/51c28f51/attachment.html 

From fs at voice2net.ca  Fri Mar 20 15:30:41 2015
From: fs at voice2net.ca (Darcy Primrose)
Date: Fri, 20 Mar 2015 08:30:41 -0400
Subject: [Freeswitch-users] Multilanguages in dialplan and voicemail
References: <CACXsapOTMPtEe_9-R-YUVKfkNUO5UX89E999XU7L+MiW7-6+Nw@mail.gmail.com>
Message-ID: <E463823477AD4A138BDDD5B2C455C8BF@DARCY>

had a similar problem with french, here is what I found that works great for us.

This did not work until I copied the sounds.xml file and the voicemail_ivr.xml file from /lang/en directory to fr directory.  Probably you could just modify it but this was easy.
The sounds.xml file in the fr directory was using speak_text instead of playing wav files and was looking for tts or cepstral.  I also had to modify the tree to the French wav files like so/usr/local/freeswitch/sounds/fr/us/callieWe deploy freeswitch into a bilingual market, so this need to work out of the boxNot sure if this is 100% the correct way to do it, but it does work.Darcy
  ----- Original Message ----- 
  From: Victor Medina 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 19, 2015 9:46 AM
  Subject: [Freeswitch-users] Multilanguages in dialplan and voicemail


  Hi guys!


  Im having troubles trying to support multilanguages destinations in freeswitch dialplan and voicemail.


  I have modules correctly loaded in modules as:

      <load module="mod_say_en"/>
      <load module="mod_say_es"/>



  In vars I only have the following...

  <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>





  And two internal destinations defined as...


  Spanish exts....


      <extension name="Local_Extension1">
        <condition field="destination_number" expression="^([0-9][0-9][0-9])$">
        .......................
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>


  English.....


      <extension name="Local_Extension2">
        <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$">
        .................
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>



  I need that Local_Extension1 have a spanish VoiceMail Voice and Local_Extension2 have a english VoiceMails. I have tried explicitly setting languages in each of the dialplan sections like..

    <action application="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/>
    <action application="set" data="default_language=es"/>
    <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>



  I'm even seeing correctly processed in console:

  Dialplan: sofia/internal/163 at cibersys.com Action sleep(1000)
  Dialplan: sofia/internal/163 at cibersys.com Action set(sound_prefix=/opt/CloudVoice-vPBX/fsw14/sounds/es/mx/maria)
  Dialplan: sofia/internal/163 at cibersys.com Action set(default_language=es)
  Dialplan: sofia/internal/163 at cibersys.com Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})



  Any ideas on what could I do? Are things setted in vars.xml taking precedence ? For example should I remove sound_prefix from vars?


  Also... any way to reload vars.xml?







  -- 


  V?ctor E. Medina M.

  Software

  +58424 291 4561
  BB #79A8AFA2 /@VMCibersys




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


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

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://confluence.freeswitch.org
  http://www.cluecon.com

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/f3f76809/attachment.html 

From lexxua at gmail.com  Fri Mar 20 10:13:46 2015
From: lexxua at gmail.com (=?UTF-8?B?0JLQu9Cw0LTQuNC80LjRgCDQpNC10LTQvtGA0L7Qsg==?=)
Date: Fri, 20 Mar 2015 09:13:46 +0200
Subject: [Freeswitch-users] Packages in Debian Repo is outdated
Message-ID: <CAO_GGAr61VPE9f+DxQfXiLb=53hz4NRq1GBXNOG-OZjjTd6wsA@mail.gmail.com>

Hello.
Is it ok that in repo is only 1.4.15 aviable ?
Thank you.

-- 
Cheers !
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/c7fc37cb/attachment.html 

From alhakeem at gmail.com  Fri Mar 20 16:37:03 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Fri, 20 Mar 2015 13:37:03 -0000
Subject: [Freeswitch-users] XML and HTTP engines or libraries
In-Reply-To: <CAEJMVkAuxpPS0c3DskAPuY1VbPxo3xof79Dp7LXm3n0Htfav4w@mail.gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>
	<CAEJMVkAuxpPS0c3DskAPuY1VbPxo3xof79Dp7LXm3n0Htfav4w@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAPPlOoX5+sFMi07ftKEL4gcBAAAAAA==@gmail.com>

 
Is there any way to replace Abyss webserver used in mod_http with NGINX or a more capable webserver, which can also be used as a reverse proxy for MySQL, Redis or other services such MySQL/InnoDB access via HTTP ?
Also, is there any way to update Libcurl sources in FS sources to support the Websocket/HTTP2 extensions without breaking mod_xml_curl ?
 
Cheers,
Abdul Hakeem
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Thursday, March 19, 2015 7:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML and HTTP engines or libraries
 
see xml_curl, confluence.freeswitch.org or wiki.freeswitch.org
 
On Thu, Mar 19, 2015 at 2:54 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
Hello,
Can someone point me to the XML/HTTP  engines or libraries used in Freeswitch ?
Thanks,
Abdul Hakeem



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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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



 
-- 
Brian West
brian at freeswitch.org
  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 
Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/>  | Register <https://freeswitch.com/cart.php?gid=1>  TODAY!
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/94a08b10/attachment-0001.html 

From ssinyagin at gmail.com  Fri Mar 20 16:52:34 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 20 Mar 2015 14:52:34 +0100
Subject: [Freeswitch-users] XML and HTTP engines or libraries
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAPPlOoX5+sFMi07ftKEL4gcBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>
	<CAEJMVkAuxpPS0c3DskAPuY1VbPxo3xof79Dp7LXm3n0Htfav4w@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAPPlOoX5+sFMi07ftKEL4gcBAAAAAA==@gmail.com>
Message-ID: <CACOYK=jRuS34T82ajp_SVgKe_6PJTv3-eCh9WDq61PGaro6sBQ@mail.gmail.com>

Abdul,

if your expected call rate is not too big, you can utilize mod_perl or
mod_lua or mod_python to take the control of a call and do all necessary
lookups in external databases.

I'm doing that for a small application server, and it works great: a
1300-lines Perl script loads and makes several SELECT's from local
MySQLserver, and then performs the dialplan applications, like playing
audio and waiting for DTMF input. The whole initialization, including
database access, takes approximately 8-10ms. Of course it would not survive
thousands calls per minute, but works perfectly for the actual task.

As a more scalable solution, you may utilize ESL and control the calls from
an external daemon. Then you are completely flexible in designing it in a
scalable and resilient way.






On Fri, Mar 20, 2015 at 2:37 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:

>
>
> Is there any way to replace Abyss webserver used in mod_http with NGINX
> or a more capable webserver, which can also be used as a reverse proxy for
> MySQL, Redis or other services such MySQL/InnoDB access via HTTP ?
>
> Also, is there any way to update Libcurl sources in FS sources to support
> the Websocket/HTTP2 extensions without breaking mod_xml_curl ?
>
>
>
> Cheers,
>
> Abdul Hakeem
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Thursday, March 19, 2015 7:57 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML and HTTP engines or libraries
>
>
>
> see xml_curl, confluence.freeswitch.org or wiki.freeswitch.org
>
>
>
> On Thu, Mar 19, 2015 at 2:54 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>
> Hello,
> Can someone point me to the XML/HTTP  engines or libraries used in
> Freeswitch ?
> Thanks,
> Abdul Hakeem
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM**:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/1b5ab259/attachment.html 

From mike at jerris.com  Fri Mar 20 16:53:12 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 20 Mar 2015 09:53:12 -0400
Subject: [Freeswitch-users] Multilanguages in dialplan and voicemail
In-Reply-To: <E463823477AD4A138BDDD5B2C455C8BF@DARCY>
References: <CACXsapOTMPtEe_9-R-YUVKfkNUO5UX89E999XU7L+MiW7-6+Nw@mail.gmail.com>
	<E463823477AD4A138BDDD5B2C455C8BF@DARCY>
Message-ID: <4CD950D5-4D4A-4296-96BC-8973971E81B1@jerris.com>

That change should be put into tree.  We dist FR sound files, we should have those phrase macros in tree that match those files.  Could you create a pull request with this update?

> On Mar 20, 2015, at 8:30 AM, Darcy Primrose <fs at voice2net.ca> wrote:
> 
> had a similar problem with french, here is what I found that works great for us.
>  
> This did not work until I copied the sounds.xml file and the voicemail_ivr.xml file from /lang/en directory to fr directory.  Probably you could just modify it but this was easy.
> The sounds.xml file in the fr directory was using speak_text instead of playing wav files and was looking for tts or cepstral.  
> I also had to modify the tree to the French wav files like so
> /usr/local/freeswitch/sounds/fr/us/callie
> We deploy freeswitch into a bilingual market, so this need to work out of the box
> Not sure if this is 100% the correct way to do it, but it does work.
> Darcy
>  
>> ----- Original Message ----- 
>> From: Victor Medina <mailto:victor.medina at cibersys.com>
>> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, March 19, 2015 9:46 AM
>> Subject: [Freeswitch-users] Multilanguages in dialplan and voicemail
>> 
>> Hi guys!
>> 
>> Im having troubles trying to support multilanguages destinations in freeswitch dialplan and voicemail.
>> 
>> I have modules correctly loaded in modules as:
>> 
>>     <load module="mod_say_en"/>
>>     <load module="mod_say_es"/>
>> 
>> 
>> In vars I only have the following...
>> 
>> <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
>> 
>> 
>> 
>> And two internal destinations defined as...
>> 
>> Spanish exts....
>> 
>>     <extension name="Local_Extension1">
>>       <condition field="destination_number" expression="^([0-9][0-9][0-9])$">
>>       .......................
>>       <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>> 
>> English.....
>> 
>>     <extension name="Local_Extension2">
>>       <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$">
>>       .................
>>       <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>> 
>> 
>> I need that Local_Extension1 have a spanish VoiceMail Voice and Local_Extension2 have a english VoiceMails. I have tried explicitly setting languages in each of the dialplan sections like..
>> 
>>   <action application="set" data="sound_prefix=$${sounds_dir}/es/mx/maria"/>
>>   <action application="set" data="default_language=es"/>
>>   <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
>> 
>> 
>> I'm even seeing correctly processed in console:
>> 
>> Dialplan: sofia/internal/163 at cibersys.com <mailto:163 at cibersys.com> Action sleep(1000)
>> Dialplan: sofia/internal/163 at cibersys.com <mailto:163 at cibersys.com> Action set(sound_prefix=/opt/CloudVoice-vPBX/fsw14/sounds/es/mx/maria)
>> Dialplan: sofia/internal/163 at cibersys.com <mailto:163 at cibersys.com> Action set(default_language=es)
>> Dialplan: sofia/internal/163 at cibersys.com <mailto:163 at cibersys.com> Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
>> 
>> 
>> Any ideas on what could I do? Are things setted in vars.xml taking precedence ? For example should I remove sound_prefix from vars?
>> 
>> Also... any way to reload vars.xml?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/80fccf1e/attachment-0001.html 

From mike at jerris.com  Fri Mar 20 16:56:26 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 20 Mar 2015 09:56:26 -0400
Subject: [Freeswitch-users] XML and HTTP engines or libraries
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAPPlOoX5+sFMi07ftKEL4gcBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMatLUF/hIRKhC2X4aCrr3EBAAAAAA==@gmail.com>
	<CAEJMVkAuxpPS0c3DskAPuY1VbPxo3xof79Dp7LXm3n0Htfav4w@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAPPlOoX5+sFMi07ftKEL4gcBAAAAAA==@gmail.com>
Message-ID: <92615D52-A82C-4244-B5E0-38089B532E5B@jerris.com>

nginx is not an embedded web server, you can certainly put it in front of freeswitch.
we build against system libcurl.  We don't have any libcurl sources in freeswitch.  We have our own websocket code already that we use in a number of places including verto, mod_xml_curl, and mod_sofia.  What exactly are you trying to accomplish?

> On Mar 20, 2015, at 9:37 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
>  
> Is there any way to replace Abyss webserver used in mod_http with NGINX or a more capable webserver, which can also be used as a reverse proxy for MySQL, Redis or other services such MySQL/InnoDB access via HTTP ?
> Also, is there any way to update Libcurl sources in FS sources to support the Websocket/HTTP2 extensions without breaking mod_xml_curl ?
>  
> Cheers,
> Abdul Hakeem
> ? <>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Thursday, March 19, 2015 7:57 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] XML and HTTP engines or libraries
>  
> see xml_curl, confluence.freeswitch.org <http://confluence.freeswitch.org/> or wiki.freeswitch.org <http://wiki.freeswitch.org/>
>  
> On Thu, Mar 19, 2015 at 2:54 PM, Abdul Hakeem <alhakeem at gmail.com <mailto:alhakeem at gmail.com>> wrote:
> Hello,
> Can someone point me to the XML/HTTP  engines or libraries used in Freeswitch ?
> Thanks,
> Abdul Hakeem
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/1dd1daa3/attachment.html 

From krice at freeswitch.org  Fri Mar 20 17:00:45 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Mar 2015 14:00:45 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <550c280d63cda_f045ff332c42ae@resque-worker-ip-10-95-163-165.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/1cda2206/attachment.html 

From orn at arnarson.net  Fri Mar 20 18:17:06 2015
From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=)
Date: Fri, 20 Mar 2015 15:17:06 +0000
Subject: [Freeswitch-users] FreeSWITCH using same Call-ID for forked
	calls
In-Reply-To: <CAFiqYundS=a7x6z1RCTP40fjtQGGhMXtQndF+mCbt5zYTepDqQ@mail.gmail.com>
References: <CAG_7Y1R+L4iy0EABT3Q+XTouopZjJ+SUeEEuo3qi_M0peSkuiw@mail.gmail.com>
	<CAEJMVkDacPLdWv17vDTrnYQiK_0w_R8rCH-+hhDXEb=SkTMDwA@mail.gmail.com>
	<CAG_7Y1SXE2NxG284-dJE+48wrm7r-=+y0jtqYg6DYGf5t6OMQg@mail.gmail.com>
	<CAEJMVkDWNRaHbXEF=p_90Q590M=VBaZvWXbM3MqcqxT+vzwH5A@mail.gmail.com>
	<CAG_7Y1QjzjqEayv+PGrA7R7DdHrr_LgP06NKg1Co25zj9GS8Ag@mail.gmail.com>
	<CAFiqYundS=a7x6z1RCTP40fjtQGGhMXtQndF+mCbt5zYTepDqQ@mail.gmail.com>
Message-ID: <CAG_7Y1R7-D2=Qu9dDXKfQEEX3O3pWi9OrJvE-R0Ly=ch8j-LjA@mail.gmail.com>

Ah! So it is. Thanks for pointing that out. I feel very herpy-derp at the
moment. I only looked at the first 4 and last 4 letters. I must look
elsewhere for an explanation. Will update this thread if FreeSWITCH has
anything to do with it.

On Fri, Mar 20, 2015 at 11:45 AM, Steven Ayre <steveayre at gmail.com> wrote:

> The Call-IDs are different on each INVITE. Not the same as you claim.
>
> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
>
> The first part varies even though the others are the same.
>
> On 20 March 2015 at 09:46, ?rn Arnarson <orn at arnarson.net> wrote:
>
>> Look at the URI -- it's different in each case. These were received
>> within 1 ms from each other as well. This is not a retransmission.
>> Do you think this is a bug? Should I mail this to the dev list?
>>
>> On Thu, Mar 19, 2015 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Looks like an invite that didnt' get a response.  You sure dude?
>>>
>>> On Thu, Mar 19, 2015 at 11:26 AM, ?rn Arnarson <orn at arnarson.net> wrote:
>>>
>>>> Hi,
>>>>
>>>> Thanks for your response.
>>>>
>>>> I installed FS from git on another server, and it still shows exactly
>>>> the same behavior. See the two new INVITES for the forked call outbound
>>>> from FS:
>>>> INVITE sip:7712552 at 192.168.10.3 SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
>>>> Max-Forwards: 69
>>>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
>>>> To: <sip:7712552 at 192.168.10.3>
>>>> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
>>>> CSeq: 73055034 INVITE
>>>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>>>> User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 251
>>>> Diversion: <sip:4151500 at 172.25.200.101>
>>>> X-FS-Support: update_display,send_info
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
>>>> s=FreeSWITCH
>>>> c=IN IP4 192.168.10.101
>>>> t=0 0
>>>> m=audio 20950 RTP/AVP 8 0 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> INVITE sip:6595454 at 192.168.10.3 SIP/2.0
>>>> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
>>>> Max-Forwards: 69
>>>> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
>>>> To: <sip:6595454 at 192.168.10.3>
>>>> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
>>>> CSeq: 73055034 INVITE
>>>> Contact: <sip:mod_sofia at 192.168.10.101:5080>
>>>> User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 251
>>>> Diversion: <sip:4151500 at 172.25.200.101>
>>>> X-FS-Support: update_display,send_info
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
>>>> s=FreeSWITCH
>>>> c=IN IP4 192.168.10.101
>>>> t=0 0
>>>> m=audio 31206 RTP/AVP 8 0 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> Regards,
>>>> ?rn
>>>>
>>>> On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>>>> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
>>>>> would highly recommend you re-test with Master or at the very least 1.4.17
>>>>> or 1.4.18 which should be out later today.
>>>>>
>>>>> 1.2 is not receiving patches, updates or support moving forward, our
>>>>> release branch is 1.4.x
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Mar 18, 2015 at 1:26 PM, ?rn Arnarson <orn at arnarson.net>
>>>>> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> Not sure whether this belong in the users list or the dev list, but
>>>>>> when in doubt; start with users :-)
>>>>>>
>>>>>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>>>>>> Asterisk on the other, and am seeing some strange behavior when calls are
>>>>>> being forked on the Asterisk.
>>>>>>
>>>>>> Call setup is like this:
>>>>>> 1. FreeSWITCH receives INVITE from Kamailio
>>>>>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>>>>>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>>>>>> (each with a unique call-id)
>>>>>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>>>>>> Call-ID from step 2.
>>>>>>
>>>>>> This is causing problems with one of the MGWs behind Kamailio, which
>>>>>> is seeing multiple INVITEs to different destinations with the same Call-ID.
>>>>>>
>>>>>> So, firstly, why is FreeSWITCH reusing call-ids?
>>>>>>
>>>>>> Secondly, how is it matching up the calls? I can't find anything
>>>>>> common in the INVITEs, other than the source number and obviously that the
>>>>>> IP sent to and received from is the same.
>>>>>>
>>>>>> I'm not sure if this is intended behavior or not, but is there a way
>>>>>> to have FreeSWITCH not do that?
>>>>>>
>>>>>> Regards,
>>>>>> ?rn
>>>>>>
>>>>>> P.S. Here is the sequence of INVITEs. I also have the console log
>>>>>> (for a different call) if needed.
>>>>>>
>>>>>> *INVITE sent to FreeSWITCH by Kamailio:*
>>>>>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>>>>>> Record-Route: <sip:172.25.200.101;lr=on>
>>>>>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>>>>>> Via: SIP/2.0/UDP 172.25.200.121:5080
>>>>>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>>>>>> Max-Forwards: 16
>>>>>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>>>>>> To: <sip:5344446 at 172.25.200.101>
>>>>>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>>>>>> CSeq: 73014324 INVITE
>>>>>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>> Supported: timer, precondition, path, replaces
>>>>>> Allow-Events: talk, hold, conference, refer
>>>>>> Content-Type: application/sdp
>>>>>> Content-Disposition: session
>>>>>> Content-Length: 229
>>>>>> X-FS-Support: update_display,send_info
>>>>>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 172.25.200.121
>>>>>> t=0 0
>>>>>> m=audio 19026 RTP/AVP 8 101
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=silenceSupp:off - - - -
>>>>>> a=ptime:20
>>>>>>
>>>>>>
>>>>>> *INVITE sent to Asterisk by FreeSWITCH:*
>>>>>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>>>>>> Max-Forwards: 15
>>>>>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>>>>>> To: <sip:5344446 at 172.26.0.62:5060>
>>>>>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>>>>>> CSeq: 73014353 INVITE
>>>>>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>>> 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: 223
>>>>>> X-FS-Support: update_display,send_info
>>>>>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 10.11.12.13
>>>>>> t=0 0
>>>>>> m=audio 23230 RTP/AVP 8 101
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=silenceSupp:off - - - -
>>>>>> a=ptime:20
>>>>>>
>>>>>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>>>>>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>>>>>> Max-Forwards: 70
>>>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>>>>>> To: <sip:7712552 at 10.11.12.13>
>>>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>>>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>>>>>> CSeq: 102 INVITE
>>>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>> INFO, PUBLISH
>>>>>> Supported: replaces, timer
>>>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>>>> Content-Type: application/sdp
>>>>>> Content-Length: 312
>>>>>>
>>>>>> v=0
>>>>>> o=root 693576967 693576967 IN IP4 172.26.0.62
>>>>>> s=Asterisk PBX 1.8.15-cert2
>>>>>> c=IN IP4 172.26.0.62
>>>>>> t=0 0
>>>>>> m=audio 30440 RTP/AVP 8 0 9 101
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:9 G722/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>>
>>>>>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>>>>>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>>>>>> Max-Forwards: 70
>>>>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>>>>>> To: <sip:6595454 at 10.11.12.13>
>>>>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>>>>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>>>>>> CSeq: 102 INVITE
>>>>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>>>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>> INFO, PUBLISH
>>>>>> Supported: replaces, timer
>>>>>> Diversion: <sip:5344446 at 172.26.0.62>
>>>>>> Content-Type: application/sdp
>>>>>> Content-Length: 310
>>>>>>
>>>>>> v=0
>>>>>> o=root 89056081 89056081 IN IP4 172.26.0.62
>>>>>> s=Asterisk PBX 1.8.15-cert2
>>>>>> c=IN IP4 172.26.0.62
>>>>>> t=0 0
>>>>>> m=audio 30708 RTP/AVP 8 0 9 101
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:9 G722/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=silenceSupp:off - - - -
>>>>>> a=ptime:20
>>>>>> a=sendrecv
>>>>>>
>>>>>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by
>>>>>> Asterisk):*
>>>>>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 172.25.200.111:5080
>>>>>> ;rport;branch=z9hG4bK3jvS9UyXU216H
>>>>>> Max-Forwards: 69
>>>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>>>>>> To: <sip:7712552 at 172.25.200.101>
>>>>>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>>>>>> CSeq: 73014353 INVITE
>>>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>> Supported: timer, precondition, path, replaces
>>>>>> Allow-Events: talk, hold, conference, refer
>>>>>> Content-Type: application/sdp
>>>>>> Content-Disposition: session
>>>>>> Content-Length: 209
>>>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>>>> X-FS-Support: update_display,send_info
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 172.25.200.111
>>>>>> t=0 0
>>>>>> m=audio 19804 RTP/AVP 8 0 9 101 13
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=ptime:20
>>>>>>
>>>>>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by
>>>>>> Asterisk):*
>>>>>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>>>>>> Via: SIP/2.0/UDP 172.25.200.111:5080
>>>>>> ;rport;branch=z9hG4bK4UNjBQF1rBrSD
>>>>>> Max-Forwards: 69
>>>>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>>>>>> To: <sip:6595454 at 172.25.200.101>
>>>>>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>>>>>> CSeq: 73014353 INVITE
>>>>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>>> REGISTER, REFER, NOTIFY
>>>>>> Supported: timer, precondition, path, replaces
>>>>>> Allow-Events: talk, hold, conference, refer
>>>>>> Content-Type: application/sdp
>>>>>> Content-Disposition: session
>>>>>> Content-Length: 209
>>>>>> Diversion: <sip:5344446 at 172.25.200.101>
>>>>>> X-FS-Support: update_display,send_info
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 172.25.200.111
>>>>>> t=0 0
>>>>>> m=audio 31376 RTP/AVP 8 0 9 101 13
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=ptime:20
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/3774d06a/attachment-0001.html 

From bote_radio at botecomm.com  Fri Mar 20 18:55:19 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 20 Mar 2015 11:55:19 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CAEhEJ-VeWrOSqJ7XwjEn=vGY5ztoYtyQBtpHqHxaZ4tU2y9LDA@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CAEhEJ-VeWrOSqJ7XwjEn=vGY5ztoYtyQBtpHqHxaZ4tU2y9LDA@mail.gmail.com>
Message-ID: <026d01d06326$44663380$cd329a80$@botecomm.com>

Apparently Aron is not the first to experience such problems. See here:

 

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues#RTPIssues-DTMFProblems

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Donny Hardyanto
Sent: Friday, 20 March, 2015 08:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF missing digits

 

I have done integration with sonus. I am sending rfc2833. Usually the dtmf suppression and fax detection setting on the sonus is the problem.

 

donny

On Friday, March 20, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

how do you send DTMF to sonus? info or rfc2833?


On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
<aronp at guaranteedplus.com <javascript:;> > wrote:
> I am experiencing problems with sonus dtmf. I tried setting different
> parameters related to the RTP BUGS listed in switch_types.h but it did not
> seem to fix the problem.
>
> I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>
> freeswitch sends the following digits to our carrier, and is received by the
> other end (also a freeswitch) missing the digits marked in black.
> pcap shows the missing digits as being part of the previous packet, so two
> 1's becomes one 1 with the duration sum of two.
>
>  1   -    1   -    2   -    1   -    1   -    #
>
> What are the relevant parameters I should set?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;> 
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 <javascript:;> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/cf0ea398/attachment.html 

From bote_radio at botecomm.com  Fri Mar 20 18:56:34 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 20 Mar 2015 11:56:34 -0400
Subject: [Freeswitch-users] Packages in Debian Repo is outdated
In-Reply-To: <CAO_GGAr61VPE9f+DxQfXiLb=53hz4NRq1GBXNOG-OZjjTd6wsA@mail.gmail.com>
References: <CAO_GGAr61VPE9f+DxQfXiLb=53hz4NRq1GBXNOG-OZjjTd6wsA@mail.gmail.com>
Message-ID: <027801d06326$716f48d0$544dda70$@botecomm.com>

That is correct.

 

To build the latest code use the ?Master? branch and compile from source code.

 

https://freeswitch.org/confluence/display/FREESWITCH/Debian#Debian-BuildingFromSource

 

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ???????? ???????
Sent: Friday, 20 March, 2015 03:14
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Packages in Debian Repo is outdated

 

Hello.

Is it ok that in repo is only 1.4.15 aviable ?

Thank you.


 

-- 

Cheers !

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/dd2e5bf9/attachment.html 

From bote_radio at botecomm.com  Fri Mar 20 19:05:50 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 20 Mar 2015 12:05:50 -0400
Subject: [Freeswitch-users] Documentation review: mod_perl expert needed
Message-ID: <027d01d06327$bc6dc810$35495830$@botecomm.com>

The FreeSWITCH community needs someone who is familiar with mod_perl to review this page and update it to reflect the current revision. The old page copied from the wiki is outdated.

 

Contact me directly or one of the core developers to add your JIRA account to the confluence-editors group and you can make the mod_perl page useful again to the community.

 

Thanks.

 

---

Bote

 

FreeSWITCH Docs Janitor

 <http://freeswitch.org/confluence> http://freeswitch.org/confluence

 

 

 

 

From: Michael Jerris
Sent: Friday, 20 March, 2015 11:53
Subject: Re: [Freeswitch-docs] mod_perl documentation

 

Quick review of this doc shows lots of out of date info particularly about deps and building.  This still needs review 

On Friday, March 20, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

hi,

I updated mod_perl in Confluence, and moved most of subpages into it:
https://freeswitch.org/confluence/display/FREESWITCH/mod_perl

I'm not sure about the Rosetta page. It looks to me that it can be easily dropped, as it's too little informative:
https://wiki.freeswitch.org/wiki/Mod_perl_Rosetta


cheers,
stan

_______________________________________________
Freeswitch-docs mailing list
Freeswitch-docs at lists.freeswitch.org <javascript:;> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/b195ef19/attachment.html 

From mike at jerris.com  Fri Mar 20 19:33:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 20 Mar 2015 12:33:14 -0400
Subject: [Freeswitch-users] Packages in Debian Repo is outdated
In-Reply-To: <027801d06326$716f48d0$544dda70$@botecomm.com>
References: <CAO_GGAr61VPE9f+DxQfXiLb=53hz4NRq1GBXNOG-OZjjTd6wsA@mail.gmail.com>
	<027801d06326$716f48d0$544dda70$@botecomm.com>
Message-ID: <DBA0C21B-34FA-40F7-B05C-82DB8A47855A@jerris.com>

Also, a new release is coming very soon, we are just working on finalizing release notes.

> On Mar 20, 2015, at 11:56 AM, Bote Man <bote_radio at botecomm.com> wrote:
> 
> That is correct.
>  
> To build the latest code use the ?Master? branch and compile from source code.
>  
> https://freeswitch.org/confluence/display/FREESWITCH/Debian#Debian-BuildingFromSource <https://freeswitch.org/confluence/display/FREESWITCH/Debian#Debian-BuildingFromSource>
>  
>  
> Bote
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ???????? ???????
> Sent: Friday, 20 March, 2015 03:14
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Packages in Debian Repo is outdated
>  
> Hello.
> Is it ok that in repo is only 1.4.15 aviable ?
> Thank you.
>  
> -- 
> Cheers !

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/34f71f61/attachment-0001.html 

From ssinyagin at gmail.com  Fri Mar 20 19:40:25 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 20 Mar 2015 17:40:25 +0100
Subject: [Freeswitch-users] Documentation review: mod_perl expert needed
In-Reply-To: <027d01d06327$bc6dc810$35495830$@botecomm.com>
References: <027d01d06327$bc6dc810$35495830$@botecomm.com>
Message-ID: <CACOYK=gKJ2E62fJUS6tNR1Ux+0G2J90Q1OBJnBs_5m43GknfGA@mail.gmail.com>

I actually added a few bits, like $session, $stream and $env
variables. But yes, needs a fresh look.

On Fri, Mar 20, 2015 at 5:05 PM, Bote Man <bote_radio at botecomm.com> wrote:
> The FreeSWITCH community needs someone who is familiar with mod_perl to
> review this page and update it to reflect the current revision. The old page
> copied from the wiki is outdated.
>
>
>
> Contact me directly or one of the core developers to add your JIRA account
> to the confluence-editors group and you can make the mod_perl page useful
> again to the community.
>
>
>
> Thanks.
>
>
>
> ---
>
> Bote
>
>
>
> FreeSWITCH Docs Janitor
>
> http://freeswitch.org/confluence
>
>
>
>
>
>
>
>
>
> From: Michael Jerris
> Sent: Friday, 20 March, 2015 11:53
> Subject: Re: [Freeswitch-docs] mod_perl documentation
>
>
>
> Quick review of this doc shows lots of out of date info particularly about
> deps and building.  This still needs review
>
> On Friday, March 20, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
> hi,
>
> I updated mod_perl in Confluence, and moved most of subpages into it:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_perl
>
> I'm not sure about the Rosetta page. It looks to me that it can be easily
> dropped, as it's too little informative:
> https://wiki.freeswitch.org/wiki/Mod_perl_Rosetta
>
>
> cheers,
> stan
>
> _______________________________________________
> Freeswitch-docs mailing list
> Freeswitch-docs at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lexxua at gmail.com  Fri Mar 20 19:24:30 2015
From: lexxua at gmail.com (=?UTF-8?B?0JLQu9Cw0LTQuNC80LjRgCDQpNC10LTQvtGA0L7Qsg==?=)
Date: Fri, 20 Mar 2015 16:24:30 +0000
Subject: [Freeswitch-users] Packages in Debian Repo is outdated
References: <CAO_GGAr61VPE9f+DxQfXiLb=53hz4NRq1GBXNOG-OZjjTd6wsA@mail.gmail.com>
	<027801d06326$716f48d0$544dda70$@botecomm.com>
Message-ID: <CAO_GGAqWyFC=hwXJcJBD5=51JAcHMmEgkoeHy2oZ5xbds2x5DQ@mail.gmail.com>

Hello, I mentioned stable 1.4 branch. Ifaik it's 1.4.17 or repo update
method is not supported anymore for Debian distro ?

On Fri, Mar 20, 2015, 5:59 PM Bote Man <bote_radio at botecomm.com> wrote:

> That is correct.
>
>
>
> To build the latest code use the ?Master? branch and compile from source
> code.
>
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debian#Debian-BuildingFromSource
>
>
>
>
>
> Bote
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *????????
> ???????
> *Sent:* Friday, 20 March, 2015 03:14
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Packages in Debian Repo is outdated
>
>
>
> Hello.
>
> Is it ok that in repo is only 1.4.15 aviable ?
>
> Thank you.
>
>
>
> --
>
> Cheers !
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/d9e7eddc/attachment.html 

From krice at freeswitch.org  Fri Mar 20 21:36:27 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Mar 2015 13:36:27 -0500
Subject: [Freeswitch-users] Packages in Debian Repo is outdated
In-Reply-To: <CAO_GGAqWyFC=hwXJcJBD5=51JAcHMmEgkoeHy2oZ5xbds2x5DQ@mail.gmail.com>
Message-ID: <D131D2DB.98FC2%krice@freeswitch.org>

The release cycle and documentation is being updated and changed to make the
Dev Teams Life easier.

Packages will be updated shorlty


On 3/20/15, 11:24 AM, "???????? ???????" <lexxua at gmail.com> wrote:

> Hello, I mentioned stable 1.4 branch. Ifaik it's 1.4.17 or repo update method
> is not supported anymore for Debian distro ?
> 
> On Fri, Mar 20, 2015, 5:59 PM?Bote Man <bote_radio at botecomm.com> wrote:
>> That is correct.
>> ?
>> To build the latest code use the ?Master? branch and compile from source
>> code.
>> ?
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian#Debian-BuildingFr
>> omSource
>> ?
>> ?
>> Bote
>> ?
>> ?
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ????????
>> ???????
>> Sent: Friday, 20 March, 2015 03:14
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] Packages in Debian Repo is outdated
>> 
>> ?
>> 
>> Hello.
>> 
>> Is it ok that in repo is only 1.4.15 aviable ?
>> 
>> Thank you.
>> ?

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


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/e6ae8895/attachment.html 

From aronp at guaranteedplus.com  Fri Mar 20 22:43:56 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Fri, 20 Mar 2015 15:43:56 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
Message-ID: <CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>

I'm sending rfc2833, I don't do any inband suppression on the channel .

On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> how do you send DTMF to sonus? info or rfc2833?
>
>
> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
> <aronp at guaranteedplus.com> wrote:
> > I am experiencing problems with sonus dtmf. I tried setting different
> > parameters related to the RTP BUGS listed in switch_types.h but it did
> not
> > seem to fix the problem.
> >
> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
> >
> > freeswitch sends the following digits to our carrier, and is received by
> the
> > other end (also a freeswitch) missing the digits marked in black.
> > pcap shows the missing digits as being part of the previous packet, so
> two
> > 1's becomes one 1 with the duration sum of two.
> >
> >  1   -    1   -    2   -    1   -    1   -    #
> >
> > What are the relevant parameters I should set?
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/b12c55fd/attachment.html 

From blasterjr at gmail.com  Fri Mar 20 23:21:42 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Fri, 20 Mar 2015 14:21:42 -0600
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
Message-ID: <CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>

Aron its not that you're doing inband suppression, Sonus is buggy and if
its in your media path it can cause lots of headaches.  Many carriers use
Sonus, some as just SBC's and some that handle media.

On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com>
wrote:

> I'm sending rfc2833, I don't do any inband suppression on the channel .
>
> On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> how do you send DTMF to sonus? info or rfc2833?
>>
>>
>> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
>> <aronp at guaranteedplus.com> wrote:
>> > I am experiencing problems with sonus dtmf. I tried setting different
>> > parameters related to the RTP BUGS listed in switch_types.h but it did
>> not
>> > seem to fix the problem.
>> >
>> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>> >
>> > freeswitch sends the following digits to our carrier, and is received
>> by the
>> > other end (also a freeswitch) missing the digits marked in black.
>> > pcap shows the missing digits as being part of the previous packet, so
>> two
>> > 1's becomes one 1 with the duration sum of two.
>> >
>> >  1   -    1   -    2   -    1   -    1   -    #
>> >
>> > What are the relevant parameters I should set?
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150320/56e693d7/attachment-0001.html 

From carpenter.john at gmail.com  Sun Mar 22 05:06:28 2015
From: carpenter.john at gmail.com (John Carpenter)
Date: Sun, 22 Mar 2015 02:06:28 +0000
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
Message-ID: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>

I am trying to use Lua to retrieve some currency conversion rates from a
postgres database. It works but the numeric field value is only returning
the first three decimal places although if I list the data in postgres it
has four decimal places. I may be doing something daft as I am quite new to
Lua, sample code is below. Any ideas appreciated

regards, John


 local dbh = freeswitch.Dbh("odbc://fs_odbc:fs_user:password")
if dbh:connected() == false then
 freeswitch.consoleLog("notice", "dp.lua cannot connect to database" .. dsn
.. "\n")
 return
end
assert(dbh:connected())

e_rate = 1.123456789
-- just a test to see decimal places
freeswitch.consoleLog("info",string.format(" e_rate = %f", e_rate))

local function eur2usd()
   local sql = "select eur2usd from currency order by id desc limit 1"
   dbh:query(sql, function(row)
       e_rate = row.eur2usd
       end)
end

local function eur2gbp()
   local sql = "select eur2gbp from currency order by id desc limit 1"
   dbh:query(sql, function(row)
       e_rate = row.eur2gbp
       end)
end

eur2usd()
freeswitch.consoleLog("info",string.format(" euro2usd = %f", e_rate))
eur2gbp()
freeswitch.consoleLog("info",string.format(" euro2gbp = %f", e_rate))
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/d34921a6/attachment.html 

From mayank19882001 at gmail.com  Sun Mar 22 18:03:57 2015
From: mayank19882001 at gmail.com (Mayank Nakrani)
Date: Sun, 22 Mar 2015 20:33:57 +0530
Subject: [Freeswitch-users] Concurrent calls with Transcoding Involved
Message-ID: <CADqjmUnADHjhHEKFiXH8xE1gWL3MwSOmpoTQjTUUGqgpBAm10A@mail.gmail.com>

Hey Guys,

A small help needed :)

We have the following configuration for our Voip calling card business.

Opensips: Handling Authentications of users / load balancing more
freeswitch servers in the future
Freeswitch : Connected with a billing software through xml_curl Handling
call and media

Codec used:
 Between user and Our server : SILK 12k
 Between Server and outbound gateway : PCMU/PCMA

All this is hosted on a XEON e3 1230 at 3.2Ghz 4 cores And 8GB RAM server.

We are using Silk as we found it to be of good call quality and better on
bandwith compared to PCMA/U .

My query was to know is this configuration enough to handle around 150-200
conc calls ?

I know freeswitch is capable of handling a lot more than this but we have
Transcoding involved here Silk-->PCMA.

Do we have ppl here having similar configuration ? Can they please share
their results about concurrent calls ?

Thanks And regards
Mayank Nakrani
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/cae7ac72/attachment.html 

From sdevoy at bizfocused.com  Sun Mar 22 19:37:19 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 22 Mar 2015 16:37:19 +0000
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <CAEJMVkBu1qaq25eV26j94==9rZ=Gz0SsJ44wTqZZU2tUV+EsXQ@mail.gmail.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
	<BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkBu1qaq25eV26j94==9rZ=Gz0SsJ44wTqZZU2tUV+EsXQ@mail.gmail.com>
Message-ID: <BY2PR0101MB093391E5D813EF1A27722883BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>

I would think so too.  Since I did provide the full logs, and I can?t figure out what happened, could someone PLEASe take a look:

In this pastebin:  https://pastebin.freeswitch.org/24023

?         Line 1 through about 170 show the call coming in from the IVR to an extension ?MAINLINE?.
it is bridged to one extension immediately (ext 10) and a couple of others with [leg delay]
It has UUID: 574583db-22d2-4b7f-a9a4-3cadd419b93d

?         In Line 284,  you can see extension 20 dialed **10 to intercept extension 10.

?         In line 298, It matches the dial plan **dd.

?         In line 299-to 302 you can see the dialplan for the intercept.

?         In line 334, you can see: ?intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)?

?         In line 340, you see sofia/external/20 at fs_esta.bizfocused.com entering state [ready][200]<mailto:sofia/external/20 at fs_esta.bizfocused.com%20entering%20state%20%5bready%5d%5b200%5d>
Does that mean the intercept was successful?

?         Then after EXCTLY the ?sleep(2000)? the channel for ext 20 says ?has executed the last dialplan instruction, hanging up.?

Does ?intercept()? not bridge the two legs together?
Am I missing a step?
I find no further entries in the log for the intercepted UUID.
Thanks for looking.
Sean


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Thursday, March 19, 2015 11:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

Sounds like the UUID isn't there that you expected,  so it couldn't pick it up. I'm guessing the full logs would show that.

On Thu, Mar 19, 2015 at 10:28 AM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Anyone help with this, PLEASE?  The customer is quite upset.

Thank you,
Sean

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 Sean Devoy
Sent: Wednesday, March 18, 2015 3:46 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

Yes, that is what I pass to the intercept call vie the dialplan/database.  Check the log:

62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan: sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> Action intercept(${db(select/esta_call_pickup_uuid/10)})
. . . .
62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)


Sean

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 Brian West
Sent: Wednesday, March 18, 2015 3:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

intercept takes a UUID, not an extension number.

On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Hi All,

I have directed call intercept running via dialplan **\d\d  (two digit extensions).

In this pastebin:  https://pastebin.freeswitch.org/24023

You should see a call come from the IVR into ?MAINLINE? which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>

Can someone who is more log savvy than I review that log and tell me what happened?

Perhaps my dialplan is missing something:

    <extension name="directed-call-pickup">
      <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?

Thanks,
Sean Devoy


_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/cf40c617/attachment-0001.html 

From rossbcan at gmail.com  Sun Mar 22 19:56:36 2015
From: rossbcan at gmail.com (Bill Ross)
Date: Sun, 22 Mar 2015 12:56:36 -0400
Subject: [Freeswitch-users] T.38 Secure Fax
Message-ID: <0caf01d064c1$2938abf0$7baa03d0$@gmail.com>

Folks;

 

I am attempting to deploy T.38 Fax (ZRTP Secure) with FreeSwitch / FusionPBX
and am seeking system architectural advice regarding how to approach this.

 

Question #1: Recommended T.38 capable ATA (and settings) for reliable FAX.

 

Approach #1 ZRTP mitm with T.38 SIP Extension (either T.38 capable ATA or
soft endpoint) - preferred approach

 

T.38 SIP endpoint <-> FS#1 / mitm <-> cloud <-> FS#2 mitm <-> T.38 SIP
endpoint

 

I am using a soft T.38 SIP endpoint, plugin for Windows Fax and Scan from
faxback, available here:

 

http://www.faxback.com/Products/Trialware.aspx?NET_SAT_EDITION=Small%20Busin
ess

 

SIP Gateway, just prior to bridge:

<action application="export" data="fax_enable_t38_request=true"/>
<action application="set" data="execute_on_answer=t38_gateway peer"/>

 

Extension, just prior to bridge:

 

<action application="set" data="fax_enable_t38_request=true"/>
<action application="set" data="execute_on_answer=t38_gateway self"/>

 

What I am seeing (two FS systems), each with T.38 soft extensions,
attempting to fax each other.

 

                ZRTP negotiation fails (Max retransmissions fail) -> NOZRTP

                ONE SDP is T38Fax (extension or gateway), Other is PCMU/8000

                The extension of T.38 enters state [ready] and immediately
hangs up, I assume from SDP mismatch

 

Question #2: Anyone successfully done this, if so, how?

Question #3: Is what I am attempting architecturally possible? Can T.38 be
mitm'd?

Question #4: If so, advice regarding how to debug the ZRTP negotiation
failure and SDP negotiation failure.

 

Approach #2: FS terminates incoming T.38 fax, stores and forwards to fax
extension, FS terminates outgoing T.38 fax, stores and forwards to SIP
gateway to final destination.

 

T.38 SIP endpoint <-> FS#1 / terminate fax <-> FS#1 forward fax <-> cloud
<-> FS#2 / terminate fax <-> FS#2 forward fax <-> T.38 SIP endpoint

 

Question #5: Will this work? Anyone done it, how? ZRTP gotcha's?

 

Thanks;

Bill

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/d4fdbc8b/attachment.html 

From ssinyagin at gmail.com  Sun Mar 22 21:35:09 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 22 Mar 2015 19:35:09 +0100
Subject: [Freeswitch-users] Concurrent calls with Transcoding Involved
In-Reply-To: <CADqjmUnADHjhHEKFiXH8xE1gWL3MwSOmpoTQjTUUGqgpBAm10A@mail.gmail.com>
References: <CADqjmUnADHjhHEKFiXH8xE1gWL3MwSOmpoTQjTUUGqgpBAm10A@mail.gmail.com>
Message-ID: <CACOYK=jrGMNH_hamPMUspt-k7vO9P=nFP1+Xx0xGXLv8Er9U+w@mail.gmail.com>

Here's a simple test scenario where the server was calling to itself, with
transcoding.
https://txlab.wordpress.com/2014/04/18/freeswitch-performance-test-on-pc-engines-apu/

You can build a similar test with your box. I can also help with
consultancy if needed.
On Mar 22, 2015 4:05 PM, "Mayank Nakrani" <mayank19882001 at gmail.com> wrote:

> Hey Guys,
>
> A small help needed :)
>
> We have the following configuration for our Voip calling card business.
>
> Opensips: Handling Authentications of users / load balancing more
> freeswitch servers in the future
> Freeswitch : Connected with a billing software through xml_curl Handling
> call and media
>
> Codec used:
>  Between user and Our server : SILK 12k
>  Between Server and outbound gateway : PCMU/PCMA
>
> All this is hosted on a XEON e3 1230 at 3.2Ghz 4 cores And 8GB RAM server.
>
> We are using Silk as we found it to be of good call quality and better on
> bandwith compared to PCMA/U .
>
> My query was to know is this configuration enough to handle around 150-200
> conc calls ?
>
> I know freeswitch is capable of handling a lot more than this but we have
> Transcoding involved here Silk-->PCMA.
>
> Do we have ppl here having similar configuration ? Can they please share
> their results about concurrent calls ?
>
> Thanks And regards
> Mayank Nakrani
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/b7bbc080/attachment.html 

From sdevoy at bizfocused.com  Sun Mar 22 22:36:10 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 22 Mar 2015 19:36:10 +0000
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <BY2PR0101MB093391E5D813EF1A27722883BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
	<BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkBu1qaq25eV26j94==9rZ=Gz0SsJ44wTqZZU2tUV+EsXQ@mail.gmail.com>
	<BY2PR0101MB093391E5D813EF1A27722883BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <BY2PR0101MB0933C03291C9AF7CB7BCE982BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>

*** RESOLVED ***

I have some testing to do, but I think I found the problem.  All of my tests have been dialing a did that routes directly to a test extension, then intercepting that call from that extension.  The user?s calls however take a slightly different route.  They come in and got directly to an IVR, then to an extension which seems to not allow intercept!

The problem lies in my intercept call dialplan:
    <extension name="directed-call-pickup">
<!-- picks up the call with db key saved like this:             <action application="db"
     data="insert/ESTA_call_pickup_uuid/22/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

A DID bridged directly to an extension that has not been ?answered? can be picked up just as expected.  However, a DID that comes in and goes to an IVR (which requires the call  to be ?answer()?ed), and then is bridged to one or more extensions CANNOT be intercepted if you include: intercept_unanswered_only=true.

It appears the correct variable is: intercept_unbridged_only=true

Hope this helps someone else in the future.

Maybe it is just me, but perhaps if an ?intercept()? fails it could give a hint in the log what happened and why!

Thanks to those that tried to help.

Sean


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
Sent: Sunday, March 22, 2015 12:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

I would think so too.  Since I did provide the full logs, and I can?t figure out what happened, could someone PLEASe take a look:

In this pastebin:  https://pastebin.freeswitch.org/24023

?         Line 1 through about 170 show the call coming in from the IVR to an extension ?MAINLINE?.
it is bridged to one extension immediately (ext 10) and a couple of others with [leg delay]
It has UUID: 574583db-22d2-4b7f-a9a4-3cadd419b93d

?         In Line 284,  you can see extension 20 dialed **10 to intercept extension 10.

?         In line 298, It matches the dial plan **dd.

?         In line 299-to 302 you can see the dialplan for the intercept.

?         In line 334, you can see: ?intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)?

?         In line 340, you see sofia/external/20 at fs_esta.bizfocused.com entering state [ready][200]<mailto:sofia/external/20 at fs_esta.bizfocused.com%20entering%20state%20%5bready%5d%5b200%5d>
Does that mean the intercept was successful?

?         Then after EXCTLY the ?sleep(2000)? the channel for ext 20 says ?has executed the last dialplan instruction, hanging up.?

Does ?intercept()? not bridge the two legs together?
Am I missing a step?
I find no further entries in the log for the intercepted UUID.
Thanks for looking.
Sean


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 Brian West
Sent: Thursday, March 19, 2015 11:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

Sounds like the UUID isn't there that you expected,  so it couldn't pick it up. I'm guessing the full logs would show that.

On Thu, Mar 19, 2015 at 10:28 AM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Anyone help with this, PLEASE?  The customer is quite upset.

Thank you,
Sean

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 Sean Devoy
Sent: Wednesday, March 18, 2015 3:46 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

Yes, that is what I pass to the intercept call vie the dialplan/database.  Check the log:

62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan: sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> Action intercept(${db(select/esta_call_pickup_uuid/10)})
. . . .
62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE sofia/external/20 at fs_esta.bizfocused.com<mailto:sofia/external/20 at fs_esta.bizfocused.com> intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)


Sean

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 Brian West
Sent: Wednesday, March 18, 2015 3:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call intercept.

intercept takes a UUID, not an extension number.

On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
Hi All,

I have directed call intercept running via dialplan **\d\d  (two digit extensions).

In this pastebin:  https://pastebin.freeswitch.org/24023

You should see a call come from the IVR into ?MAINLINE? which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up.<mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>

Can someone who is more log savvy than I review that log and tell me what happened?

Perhaps my dialplan is missing something:

    <extension name="directed-call-pickup">
      <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
      <condition field="destination_number" expression="^\*\*(\d\d)$">
        <action application="answer"/>
        <action application="set" data="intercept_unanswered_only=true"/>
        <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?

Thanks,
Sean Devoy


_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/ba588f9f/attachment-0001.html 

From brian at freeswitch.org  Sun Mar 22 22:41:08 2015
From: brian at freeswitch.org (Brian West)
Date: Sun, 22 Mar 2015 14:41:08 -0500
Subject: [Freeswitch-users] T.38 Secure Fax
In-Reply-To: <0caf01d064c1$2938abf0$7baa03d0$@gmail.com>
References: <0caf01d064c1$2938abf0$7baa03d0$@gmail.com>
Message-ID: <CAEJMVkCNnVxrRa6BTp7BPmdcE5XNg_RS1dx01LukGGiyvnLM-Q@mail.gmail.com>

Not possible since t.38 doent take place over RTP!

On Sunday, March 22, 2015, Bill Ross <rossbcan at gmail.com> wrote:

> Folks;
>
>
>
> I am attempting to deploy T.38 Fax (ZRTP Secure) with FreeSwitch /
> FusionPBX and am seeking system architectural advice regarding how to
> approach this.
>
>
>
> Question #1: Recommended T.38 capable ATA (and settings) for reliable FAX.
>
>
>
> Approach #1 ZRTP mitm with T.38 SIP Extension (either T.38 capable ATA or
> soft endpoint) ? preferred approach
>
>
>
> T.38 SIP endpoint <-> FS#1 / mitm <-> cloud <-> FS#2 mitm <-> T.38 SIP
> endpoint
>
>
>
> I am using a soft T.38 SIP endpoint, plugin for Windows Fax and Scan from
> faxback, available here:
>
>
>
>
> http://www.faxback.com/Products/Trialware.aspx?NET_SAT_EDITION=Small%20Business
>
>
>
> SIP Gateway, just prior to bridge:
>
> *<action application="export" data="fax_enable_t38_request=true"/>*
> *<action application="set" data="execute_on_answer=t38_gateway peer"/>*
>
>
>
> *Extension, just prior to bridge:*
>
>
>
> *<action application="set" data="fax_enable_t38_request=true"/>*
> *<action application="set" data="execute_on_answer=t38_gateway self"/>*
>
>
>
> *What I am seeing (two FS systems), each with T.38 soft extensions,
> attempting to fax each other.*
>
>
>
> *                ZRTP negotiation fails (Max retransmissions fail) ->
> NOZRTP*
>
> *                ONE SDP is T38Fax (extension or gateway), Other is
> PCMU/8000*
>
> *                The extension of T.38 enters state [ready] and
> immediately hangs up, I assume from SDP mismatch*
>
>
>
> Question #2: Anyone successfully done this, if so, how?
>
> Question #3: Is what I am attempting architecturally possible? Can T.38
> be mitm?d?
>
> Question #4: If so, advice regarding how to debug the ZRTP negotiation
> failure and SDP negotiation failure.
>
>
>
> Approach #2: FS terminates incoming T.38 fax, stores and forwards to fax
> extension, FS terminates outgoing T.38 fax, stores and forwards to SIP
> gateway to final destination.
>
>
>
> T.38 SIP endpoint <-> FS#1 / terminate fax <-> FS#1 forward fax <-> cloud
> <-> FS#2 / terminate fax <-> FS#2 forward fax <-> T.38 SIP endpoint
>
>
>
> Question #5: Will this work? Anyone done it, how? ZRTP gotcha?s?
>
>
>
> Thanks;
>
> Bill
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150322/5e09f067/attachment.html 

From mike at jerris.com  Mon Mar 23 07:28:02 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 23 Mar 2015 00:28:02 -0400
Subject: [Freeswitch-users] Call intercept.
In-Reply-To: <BY2PR0101MB0933C03291C9AF7CB7BCE982BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>
References: <BY2PR0101MB09331E4C331CD683CA866636BB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkB=P01ey5ze+E2MB-fREkmAZGc_bd5vtPoKfLSwJ5Wwag@mail.gmail.com>
	<BY2PR0101MB09330CAD2B765D1FD1B6977EBB000@BY2PR0101MB0933.prod.exchangelabs.com>
	<BY2PR0101MB09338A7D185211CBC41552BBBB010@BY2PR0101MB0933.prod.exchangelabs.com>
	<CAEJMVkBu1qaq25eV26j94==9rZ=Gz0SsJ44wTqZZU2tUV+EsXQ@mail.gmail.com>
	<BY2PR0101MB093391E5D813EF1A27722883BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>
	<BY2PR0101MB0933C03291C9AF7CB7BCE982BB0C0@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <C86A0A20-2AC4-4DB2-942F-39A39368845A@jerris.com>

We would be happy to accept a pull request adding some logging there.

> On Mar 22, 2015, at 3:36 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> *** RESOLVED ***
>  
> I have some testing to do, but I think I found the problem.  All of my tests have been dialing a did that routes directly to a test extension, then intercepting that call from that extension.  The user?s calls however take a slightly different route.  They come in and got directly to an IVR, then to an extension which seems to not allow intercept!
>  
> The problem lies in my intercept call dialplan:
>     <extension name="directed-call-pickup">
> <!-- picks up the call with db key saved like this:             <action application="db"
>      data="insert/ESTA_call_pickup_uuid/22/${uuid}"/>  -->
>       <condition field="destination_number" expression="^\*\*(\d\d)$">
>         <action application="answer"/>
>         <action application="set" data="intercept_unanswered_only=true"/>
>         <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
>         <action application="sleep" data="2000"/>
>       </condition>
>     </extension>
>  
> A DID bridged directly to an extension that has not been ?answered? can be picked up just as expected.  However, a DID that comes in and goes to an IVR (which requires the call  to be ?answer()?ed), and then is bridged to one or more extensions CANNOT be intercepted if you include: intercept_unanswered_only=true.
>  
> It appears the correct variable is: intercept_unbridged_only=true
>  
> Hope this helps someone else in the future.
>  
> Maybe it is just me, but perhaps if an ?intercept()? fails it could give a hint in the log what happened and why!
>  
> Thanks to those that tried to help.
>  
> Sean
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
> Sent: Sunday, March 22, 2015 12:37 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Call intercept.
>  
> I would think so too.  Since I did provide the full logs, and I can?t figure out what happened, could someone PLEASe take a look:
>  
> In this pastebin:  https://pastebin.freeswitch.org/24023 <https://pastebin.freeswitch.org/24023>
> ?         Line 1 through about 170 show the call coming in from the IVR to an extension ?MAINLINE?.
> it is bridged to one extension immediately (ext 10) and a couple of others with [leg delay]
> It has UUID: 574583db-22d2-4b7f-a9a4-3cadd419b93d
> ?         In Line 284,  you can see extension 20 dialed **10 to intercept extension 10.
> ?         In line 298, It matches the dial plan **dd.
> ?         In line 299-to 302 you can see the dialplan for the intercept.
> ?         In line 334, you can see: ?intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)?
> ?         In line 340, you see sofia/external/20 at fs_esta.bizfocused.com entering state [ready][200] <mailto:sofia/external/20 at fs_esta.bizfocused.com%20entering%20state%20%5bready%5d%5b200%5d>
> Does that mean the intercept was successful?
> ?         Then after EXCTLY the ?sleep(2000)? the channel for ext 20 says ?has executed the last dialplan instruction, hanging up.?
>  
> Does ?intercept()? not bridge the two legs together? 
> Am I missing a step? 
> I find no further entries in the log for the intercepted UUID.
> Thanks for looking.
> Sean
>  
>  
> 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 West
> Sent: Thursday, March 19, 2015 11:48 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Call intercept.
>  
> Sounds like the UUID isn't there that you expected,  so it couldn't pick it up. I'm guessing the full logs would show that. 
>  
> On Thu, Mar 19, 2015 at 10:28 AM, Sean Devoy <sdevoy at bizfocused.com <mailto:sdevoy at bizfocused.com>> wrote:
> Anyone help with this, PLEASE?  The customer is quite upset.
>  
> Thank you,
> Sean
>  
> 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 Sean Devoy
> Sent: Wednesday, March 18, 2015 3:46 PM
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Call intercept.
>  
> Yes, that is what I pass to the intercept call vie the dialplan/database.  Check the log:
>  
> 62cc7ef5-f121-4fa2-97be-e45ff28d9c95 Dialplan: sofia/external/20 at fs_esta.bizfocused.com <mailto:sofia/external/20 at fs_esta.bizfocused.com> Actionintercept(${db(select/esta_call_pickup_uuid/10)})
> . . . . 
> 62cc7ef5-f121-4fa2-97be-e45ff28d9c95 EXECUTE sofia/external/20 at fs_esta.bizfocused.com <mailto:sofia/external/20 at fs_esta.bizfocused.com>intercept(574583db-22d2-4b7f-a9a4-3cadd419b93d)
>  
>  
> Sean
>  
> 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 West
> Sent: Wednesday, March 18, 2015 3:39 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Call intercept.
>  
> intercept takes a UUID, not an extension number.
>  
> On Wed, Mar 18, 2015 at 2:12 PM, Sean Devoy <sdevoy at bizfocused.com <mailto:sdevoy at bizfocused.com>> wrote:
> Hi All,
>  
> I have directed call intercept running via dialplan **\d\d  (two digit extensions).
>  
> In this pastebin:  https://pastebin.freeswitch.org/24023 <https://pastebin.freeswitch.org/24023>
>  
> You should see a call come from the IVR into ?MAINLINE? which dials ext 10, then after delay adds other extensions.  In this case, the person from extension 10 was at extension 20 and dialed **10.  They report it did not work.  I must say I am not clear if it worked or not.  It appears to work, but ended with sofia/external/20 at fs_esta.bizfocused.com has executed the last dialplan instruction, hanging up. <mailto:sofia/external/20 at fs_esta.bizfocused.com%20has%20executed%20the%20last%20dialplan%20instruction,%20hanging%20up.>
>  
> Can someone who is more log savvy than I review that log and tell me what happened?
>  
> Perhaps my dialplan is missing something:
>  
>     <extension name="directed-call-pickup">
>       <!-- picks up the call with db key saved like this:             <action application="db" data="insert/ESTA_call_pickup_uuid/223/${uuid}"/>  -->
>       <condition field="destination_number" expression="^\*\*(\d\d)$">
>         <action application="answer"/>
>         <action application="set" data="intercept_unanswered_only=true"/>
>         <action application="intercept" data="${db(select/esta_call_pickup_uuid/$1)}"/>
>         <action application="sleep" data="2000"/>
>       </condition>
>     </extension>
>  
> I see in the log the uuid for the appropriate call is in the intercept statement.  After intercept, I am missing something to bridge the 2 legs together?
>  
> Thanks,
> Sean Devoy
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/8de22681/attachment-0001.html 

From steveayre at gmail.com  Mon Mar 23 13:23:59 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 23 Mar 2015 10:23:59 +0000
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
Message-ID: <CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>

Have you tried a pgsql:// dsn? In case the ODBC driver is having some
truncating effect.

On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com> wrote:

> I am trying to use Lua to retrieve some currency conversion rates from a
> postgres database. It works but the numeric field value is only returning
> the first three decimal places although if I list the data in postgres it
> has four decimal places. I may be doing something daft as I am quite new to
> Lua, sample code is below. Any ideas appreciated
>
> regards, John
>
>
>  local dbh = freeswitch.Dbh("odbc://fs_odbc:fs_user:password")
> if dbh:connected() == false then
>  freeswitch.consoleLog("notice", "dp.lua cannot connect to database" ..
> dsn .. "\n")
>  return
> end
> assert(dbh:connected())
>
> e_rate = 1.123456789
> -- just a test to see decimal places
> freeswitch.consoleLog("info",string.format(" e_rate = %f", e_rate))
>
> local function eur2usd()
>    local sql = "select eur2usd from currency order by id desc limit 1"
>    dbh:query(sql, function(row)
>        e_rate = row.eur2usd
>        end)
> end
>
> local function eur2gbp()
>    local sql = "select eur2gbp from currency order by id desc limit 1"
>    dbh:query(sql, function(row)
>        e_rate = row.eur2gbp
>        end)
> end
>
> eur2usd()
> freeswitch.consoleLog("info",string.format(" euro2usd = %f", e_rate))
> eur2gbp()
> freeswitch.consoleLog("info",string.format(" euro2gbp = %f", e_rate))
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/ad9de2a8/attachment.html 

From carpenter.john at gmail.com  Mon Mar 23 13:59:23 2015
From: carpenter.john at gmail.com (John Carpenter)
Date: Mon, 23 Mar 2015 10:59:23 +0000
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
	<CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
Message-ID: <CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>

Thats it, changed connection to a pgsql:// dsn and I get four decimal
places, changed it back to ODBC and back to 3 decimal places.
Thanks very much, been driving me mad

On 23 March 2015 at 10:23, Steven Ayre <steveayre at gmail.com> wrote:

> Have you tried a pgsql:// dsn? In case the ODBC driver is having some
> truncating effect.
>
> On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com>
> wrote:
>
>> I am trying to use Lua to retrieve some currency conversion rates from a
>> postgres database. It works but the numeric field value is only returning
>> the first three decimal places although if I list the data in postgres it
>> has four decimal places. I may be doing something daft as I am quite new to
>> Lua, sample code is below. Any ideas appreciated
>>
>> regards, John
>>
>>
>>  local dbh = freeswitch.Dbh("odbc://fs_odbc:fs_user:password")
>> if dbh:connected() == false then
>>  freeswitch.consoleLog("notice", "dp.lua cannot connect to database" ..
>> dsn .. "\n")
>>  return
>> end
>> assert(dbh:connected())
>>
>> e_rate = 1.123456789
>> -- just a test to see decimal places
>> freeswitch.consoleLog("info",string.format(" e_rate = %f", e_rate))
>>
>> local function eur2usd()
>>    local sql = "select eur2usd from currency order by id desc limit 1"
>>    dbh:query(sql, function(row)
>>        e_rate = row.eur2usd
>>        end)
>> end
>>
>> local function eur2gbp()
>>    local sql = "select eur2gbp from currency order by id desc limit 1"
>>    dbh:query(sql, function(row)
>>        e_rate = row.eur2gbp
>>        end)
>> end
>>
>> eur2usd()
>> freeswitch.consoleLog("info",string.format(" euro2usd = %f", e_rate))
>> eur2gbp()
>> freeswitch.consoleLog("info",string.format(" euro2gbp = %f", e_rate))
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/3f4c0387/attachment.html 

From bote_radio at botecomm.com  Mon Mar 23 15:05:53 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 23 Mar 2015 08:05:53 -0400
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>	<CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
	<CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>
Message-ID: <009001d06561$b6c5bdf0$245139d0$@botecomm.com>

I?m not much of a programmer, but can?t you specify a format to %f to force it to return 4 decimal places?

 

Or is that not available in Lua?

 

Bote

 

 

From: John Carpenter
Sent: Monday, 23 March, 2015 06:59
Subject: Re: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch

 

Thats it, changed connection to a pgsql:// dsn and I get four decimal places, changed it back to ODBC and back to 3 decimal places. 

Thanks very much, been driving me mad

 

On 23 March 2015 at 10:23, Steven Ayre <steveayre at gmail.com> wrote:

Have you tried a pgsql:// dsn? In case the ODBC driver is having some truncating effect.

 

On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com> wrote:

I am trying to use Lua to retrieve some currency conversion rates from a postgres database. It works but the numeric field value is only returning the first three decimal places although if I list the data in postgres it has four decimal places. I may be doing something daft as I am quite new to Lua, sample code is below. Any ideas appreciated

regards, John

 

 

local dbh = freeswitch.Dbh("odbc://fs_odbc:fs_user:password") 
if dbh:connected() == false then 
 freeswitch.consoleLog("notice", "dp.lua cannot connect to database" .. dsn .. "\n") 
 return 
end 
assert(dbh:connected()) 

e_rate = 1.123456789 
-- just a test to see decimal places
freeswitch.consoleLog("info",string.format(" e_rate = %f", e_rate)) 

local function eur2usd() 
   local sql = "select eur2usd from currency order by id desc limit 1" 
   dbh:query(sql, function(row) 
       e_rate = row.eur2usd 
       end) 
end 

local function eur2gbp() 
   local sql = "select eur2gbp from currency order by id desc limit 1" 
   dbh:query(sql, function(row) 
       e_rate = row.eur2gbp 
       end) 
end 

eur2usd() 
freeswitch.consoleLog("info",string.format(" euro2usd = %f", e_rate)) 
eur2gbp() 
freeswitch.consoleLog("info",string.format(" euro2gbp = %f", e_rate))

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/b9121929/attachment-0001.html 

From lists at telefaks.de  Mon Mar 23 15:43:33 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 23 Mar 2015 13:43:33 +0100
Subject: [Freeswitch-users] Play ringback sound file while transfering a
 call (before ringing state)
Message-ID: <55100A75.7050502@telefaks.de>

Hello,

I have the following scenario

  * a call goes into an IVR
  * there it is answered
  * then I want to forward the call to another number via uuid_transfer
  * a new transfer dialplan is then generated
  * I want to issue a ringback sound file to the caller, in order to
    show that the call is not hung up
  * I just don't get the ringback to work properly

That's what I did
       <action application="set"
data="transfer_ringback=local_stream://moh"/>
This does silence for some seconds, and then when the transfer extension
is ringing, plays MOH. So the caller may assume for some seconds that
the call is hungup before MOH plays.

       <action application="displace_session" data="local_stream://moh"/>
This also does silence for some seconds, and then when the transfer
extension is ringing, plays MOH. So the caller may assume for some
seconds that the call is hungup before MOH plays.

I also tried the ringback application with no better result.

So how can I play a background sound for the time

  * after transfer
  * and before the ringing/progress message is received?


-- 
With kind regards
Peter Steinbach 

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/152d26c2/attachment.html 

From italorossib at gmail.com  Mon Mar 23 15:45:04 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 23 Mar 2015 09:45:04 -0300
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <009001d06561$b6c5bdf0$245139d0$@botecomm.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
	<CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
	<CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>
	<009001d06561$b6c5bdf0$245139d0$@botecomm.com>
Message-ID: <CAJR7f2FdGRYrQKJjZPatFXE-Ohf0xUN+E48ZQcR21PscO0sxKA@mail.gmail.com>

Yep, it's available and defaults to 6 decimal places, but looks like it's
not a Lua issue...

On Mon, Mar 23, 2015 at 9:05 AM, Bote Man <bote_radio at botecomm.com> wrote:

> I?m not much of a programmer, but can?t you specify a format to %f to
> force it to return 4 decimal places?
>
>
>
> Or is that not available in Lua?
>
>
>
> Bote
>
>
>
>
>
> *From:* John Carpenter
> *Sent:* Monday, 23 March, 2015 06:59
> *Subject:* Re: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
>
>
>
> Thats it, changed connection to a pgsql:// dsn and I get four decimal
> places, changed it back to ODBC and back to 3 decimal places.
>
> Thanks very much, been driving me mad
>
>
>
> On 23 March 2015 at 10:23, Steven Ayre <steveayre at gmail.com> wrote:
>
> Have you tried a pgsql:// dsn? In case the ODBC driver is having some
> truncating effect.
>
>
>
> On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com>
> wrote:
>
> I am trying to use Lua to retrieve some currency conversion rates from a
> postgres database. It works but the numeric field value is only returning
> the first three decimal places although if I list the data in postgres it
> has four decimal places. I may be doing something daft as I am quite new to
> Lua, sample code is below. Any ideas appreciated
>
> regards, John
>
>
>
>
>
> *local* dbh = freeswitch.Dbh(*"odbc://fs_odbc:fs_user:password"*)
> *if* dbh:connected() == *false* *then*
>  freeswitch.consoleLog(*"notice"*, *"dp.lua cannot connect to database"*
> .. dsn .. *"\n"*)
>  *return*
> *end*
> *assert*(dbh:connected())
>
> e_rate = *1.123456789*
> *-- just a test* to see decimal places
> freeswitch.consoleLog(*"info"*,*string.format*(*" e_rate = %f"*, e_rate))
>
> *local* *function* eur2usd()
>    *local* sql = *"select eur2usd from currency order by id desc limit 1"*
>    dbh:query(sql, *function*(row)
>        e_rate = row.eur2usd
>        *end*)
> *end*
>
> *local* *function* eur2gbp()
>    *local* sql = *"select eur2gbp from currency order by id desc limit 1"*
>    dbh:query(sql, *function*(row)
>        e_rate = row.eur2gbp
>        *end*)
> *end*
>
> eur2usd()
> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2usd = %f"*,
> e_rate))
> eur2gbp()
> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2gbp = %f"*,
> e_rate))
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/824a37b8/attachment.html 

From krice at freeswitch.org  Mon Mar 23 19:53:26 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 23 Mar 2015 16:53:26 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March
	14th-20th
Message-ID: <551045065e19e_cea275131876df@resque-worker-ip-10-186-156-166.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1HtsI3F
FreeSWITCH Week in Review (Master Branch) March 14th-20th
Hello, again. This passed week in the FreeSWITCH master branch we had 6 commits. The features for this week are: the ability to pass -T[path to custom sources path] and -K[custom keyfile path] options to build the FS packages with a custom sources and keyring path when using the debian/utils.sh script to build FS packages and play_and_detect_speech can now detect DTMF if you set the playback_terminators channel variable to any or specific DTMF.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7373 When using the debian/utils.sh script to build FreeSWITCH packages, add the ability to pass -T[path to custom sources path] and -K[custom keyfile path] options to build the FS packages with a custom sources and keyring path.

FS-7378 Play_and_detect_speech can now detect DTMF if you set the playback_terminators channel variable to any or specific DTMF. The result will be stored in speech_detect_result and in playback_terminator_used. Also added channel variable play_and_detect_speech_close_asr which will release the speech recognition port when the detection is completed. This will prevent speech licenses from being held the entire call.

The following bugs were squashed:

FS-7297 Fix incomplete patch for previous crash in mod_com_g729.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/f5736cf7/attachment.html 

From olegstolyar at gmail.com  Mon Mar 23 21:20:40 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 23 Mar 2015 11:20:40 -0700
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42 and
	WebRTC
Message-ID: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>

Hi guys,

Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
then they complain that they cannot hear audio from FreeSWITCH or send
audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
several occurrences.  It happens on both Windows and Macs.  Here are some
details"

1. Usually the problem is solved if we close Chrome and make sure all the
Chrome processes are killed.

2. When the problem happens, FS logs this on connection:
[ERR] switch_rtp.c:2976 audio DTLS packet not written

3. I tried using this WebRTC test on an affected computer:
https://janus.conf.meetecho.com/echotest.html
and that worked, so it is not a general WebRTC issue.

4. I tried FS WebRTC demo page
https://webrtc.freeswitch.org/verto/index.html#page-main
and on the affected computer the problem was still there - no audio.

5. This happened both on an old FS version from last May and on a very
recent master FreeSWITCH Version
1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
16:44:11Z 64bit)

Any ideas?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/a8641818/attachment.html 

From bote_radio at botecomm.com  Mon Mar 23 21:30:46 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 23 Mar 2015 14:30:46 -0400
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
	March	14th-20th
In-Reply-To: <551045065e19e_cea275131876df@resque-worker-ip-10-186-156-166.mail>
References: <551045065e19e_cea275131876df@resque-worker-ip-10-186-156-166.mail>
Message-ID: <00b601d06597$7b3bd450$71b37cf0$@botecomm.com>

In FS-7378 

Make that ?detect_speech_result? instead.

 

New functionality added to

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech

 

Thanks, Chris!

 

Bote

 

 

From: Ken Rice
Sent: Monday, 23 March, 2015 12:53
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March 14th-20th

 

New features that were added:

*	FS-7378 <http://ift.tt/1HtsE3M>  Play_and_detect_speech can now detect DTMF if you set the playback_terminators channel variable to any or specific DTMF. The result will be stored in speech_detect_result and in playback_terminator_used. Also added channel variable play_and_detect_speech_close_asr which will release the speech recognition port when the detection is completed. This will prevent speech licenses from being held the entire call.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/67a234f9/attachment-0001.html 

From abaci64 at gmail.com  Mon Mar 23 21:40:29 2015
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 23 Mar 2015 14:40:29 -0400
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <CAJR7f2FdGRYrQKJjZPatFXE-Ohf0xUN+E48ZQcR21PscO0sxKA@mail.gmail.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
	<CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
	<CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>
	<009001d06561$b6c5bdf0$245139d0$@botecomm.com>
	<CAJR7f2FdGRYrQKJjZPatFXE-Ohf0xUN+E48ZQcR21PscO0sxKA@mail.gmail.com>
Message-ID: <CAFmZFiPeeAgYjdq4g4MxvOQPbdCDu67_aKEfhb0Y14jH0p0pFw@mail.gmail.com>

I'm doing something very similar, this is what I use:

local rate = string.format("%.4f", currency]["rate"])


On Mon, Mar 23, 2015 at 8:45 AM, ?talo Rossi <italorossib at gmail.com> wrote:

> Yep, it's available and defaults to 6 decimal places, but looks like it's
> not a Lua issue...
>
> On Mon, Mar 23, 2015 at 9:05 AM, Bote Man <bote_radio at botecomm.com> wrote:
>
>> I?m not much of a programmer, but can?t you specify a format to %f to
>> force it to return 4 decimal places?
>>
>>
>>
>> Or is that not available in Lua?
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>> *From:* John Carpenter
>> *Sent:* Monday, 23 March, 2015 06:59
>> *Subject:* Re: [Freeswitch-users] Problem with LUA dbh:query in
>> Freeswitch
>>
>>
>>
>> Thats it, changed connection to a pgsql:// dsn and I get four decimal
>> places, changed it back to ODBC and back to 3 decimal places.
>>
>> Thanks very much, been driving me mad
>>
>>
>>
>> On 23 March 2015 at 10:23, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Have you tried a pgsql:// dsn? In case the ODBC driver is having some
>> truncating effect.
>>
>>
>>
>> On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com>
>> wrote:
>>
>> I am trying to use Lua to retrieve some currency conversion rates from a
>> postgres database. It works but the numeric field value is only returning
>> the first three decimal places although if I list the data in postgres it
>> has four decimal places. I may be doing something daft as I am quite new to
>> Lua, sample code is below. Any ideas appreciated
>>
>> regards, John
>>
>>
>>
>>
>>
>> *local* dbh = freeswitch.Dbh(*"odbc://fs_odbc:fs_user:password"*)
>> *if* dbh:connected() == *false* *then*
>>  freeswitch.consoleLog(*"notice"*, *"dp.lua cannot connect to database"*
>> .. dsn .. *"\n"*)
>>  *return*
>> *end*
>> *assert*(dbh:connected())
>>
>> e_rate = *1.123456789*
>> *-- just a test* to see decimal places
>> freeswitch.consoleLog(*"info"*,*string.format*(*" e_rate = %f"*,
>> e_rate))
>>
>> *local* *function* eur2usd()
>>    *local* sql = *"select eur2usd from currency order by id desc limit
>> 1"*
>>    dbh:query(sql, *function*(row)
>>        e_rate = row.eur2usd
>>        *end*)
>> *end*
>>
>> *local* *function* eur2gbp()
>>    *local* sql = *"select eur2gbp from currency order by id desc limit
>> 1"*
>>    dbh:query(sql, *function*(row)
>>        e_rate = row.eur2gbp
>>        *end*)
>> *end*
>>
>> eur2usd()
>> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2usd = %f"*,
>> e_rate))
>> eur2gbp()
>> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2gbp = %f"*,
>> e_rate))
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/f4f6fc23/attachment.html 

From anthony.minessale at gmail.com  Mon Mar 23 21:41:15 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 23 Mar 2015 13:41:15 -0500
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and WebRTC
In-Reply-To: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
Message-ID: <CAKbxfG81zkW+i_6Z3tA5ktXPwgfTtOMa57ChsZuhyd9cSXonJQ@mail.gmail.com>

We are working to release 1.6.  WebRTC is very new and we are working to
keep up with chrome.
I've experienced this problem before where chrome stops working right until
you kill all instances of chrome.  There is not much we can do its usually
related to hardware enumeration of the devices etc.
You may want to consider having your employer look at our FSA platform at
https://freeswitch.com/cart.php?gid=5  Then when 1.6 comes out you will
have early access to the latest WebRTC work we are doing.




On Mon, Mar 23, 2015 at 1:20 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Hi guys,
>
> Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
> then they complain that they cannot hear audio from FreeSWITCH or send
> audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
> several occurrences.  It happens on both Windows and Macs.  Here are some
> details"
>
> 1. Usually the problem is solved if we close Chrome and make sure all the
> Chrome processes are killed.
>
> 2. When the problem happens, FS logs this on connection:
> [ERR] switch_rtp.c:2976 audio DTLS packet not written
>
> 3. I tried using this WebRTC test on an affected computer:
> https://janus.conf.meetecho.com/echotest.html
> and that worked, so it is not a general WebRTC issue.
>
> 4. I tried FS WebRTC demo page
> https://webrtc.freeswitch.org/verto/index.html#page-main
> and on the affected computer the problem was still there - no audio.
>
> 5. This happened both on an old FS version from last May and on a very
> recent master FreeSWITCH Version
> 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
> 16:44:11Z 64bit)
>
> Any ideas?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/fd080bc2/attachment.html 

From bote_radio at botecomm.com  Mon Mar 23 21:42:54 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 23 Mar 2015 14:42:54 -0400
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and	WebRTC
In-Reply-To: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
Message-ID: <00bb01d06599$2d2d75a0$878860e0$@botecomm.com>

I am running 

Chrome Version 41.0.2272.101 m

on Windows 7 and checked the 888 FreeSWITCH conference yesterday, it had audio both directions. I tried it 1 or 2 weeks ago and it had no audio so it must have updated itself behind my back because I did not explicitly update Chrome.

 

Firefox 36.0.1 did not have any audio last week and now 36.0.4 has no audio either.

 

I am testing this on my home network using a Cisco RV042 router and the default checkboxes on the FS WebRTC page which apparently default to using STUN.

 

Anthm has stated many times that WebRTC is a moving target so enjoy the exhilaration of these rapid developments! I believe this is why he insists that WebRTC users keep FreeSWITCH updated so that it tracks the changes in the major web browsers.

 

Bote

 

 

 

From: Oleg Stolyar
Sent: Monday, 23 March, 2015 14:21
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42 and WebRTC

 

Hi guys,

 

Ever since my users switched to Chrome 40, 41 or 42 beta, every now and then they complain that they cannot hear audio from FreeSWITCH or send audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch several occurrences.  It happens on both Windows and Macs.  Here are some details"

 

1. Usually the problem is solved if we close Chrome and make sure all the Chrome processes are killed.

 

2. When the problem happens, FS logs this on connection:

[ERR] switch_rtp.c:2976 audio DTLS packet not written

 

3. I tried using this WebRTC test on an affected computer:

https://janus.conf.meetecho.com/echotest.html

and that worked, so it is not a general WebRTC issue.

 

4. I tried FS WebRTC demo page

https://webrtc.freeswitch.org/verto/index.html#page-main

and on the affected computer the problem was still there - no audio.

 

5. This happened both on an old FS version from last May and on a very recent master FreeSWITCH Version 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16 16:44:11Z 64bit)

 

Any ideas?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/d035eab6/attachment-0001.html 

From olegstolyar at gmail.com  Mon Mar 23 21:52:37 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 23 Mar 2015 11:52:37 -0700
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and WebRTC
In-Reply-To: <CAKbxfG81zkW+i_6Z3tA5ktXPwgfTtOMa57ChsZuhyd9cSXonJQ@mail.gmail.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
	<CAKbxfG81zkW+i_6Z3tA5ktXPwgfTtOMa57ChsZuhyd9cSXonJQ@mail.gmail.com>
Message-ID: <CAC8JMO+PZ3kG4h08U8mDSBbc21aWzoD0XMRFLfPXog4HaZeAWQ@mail.gmail.com>

Thanks Anthony!

Completely understandable about WebRTC being a moving target.  Are you
saying that 1.6 will have a fix for this?  Do you have an estimate when it
will be available to the community?

On Mon, Mar 23, 2015 at 11:41 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We are working to release 1.6.  WebRTC is very new and we are working to
> keep up with chrome.
> I've experienced this problem before where chrome stops working right
> until you kill all instances of chrome.  There is not much we can do its
> usually related to hardware enumeration of the devices etc.
> You may want to consider having your employer look at our FSA platform at
> https://freeswitch.com/cart.php?gid=5  Then when 1.6 comes out you will
> have early access to the latest WebRTC work we are doing.
>
>
>
>
> On Mon, Mar 23, 2015 at 1:20 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Hi guys,
>>
>> Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
>> then they complain that they cannot hear audio from FreeSWITCH or send
>> audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
>> several occurrences.  It happens on both Windows and Macs.  Here are some
>> details"
>>
>> 1. Usually the problem is solved if we close Chrome and make sure all the
>> Chrome processes are killed.
>>
>> 2. When the problem happens, FS logs this on connection:
>> [ERR] switch_rtp.c:2976 audio DTLS packet not written
>>
>> 3. I tried using this WebRTC test on an affected computer:
>> https://janus.conf.meetecho.com/echotest.html
>> and that worked, so it is not a general WebRTC issue.
>>
>> 4. I tried FS WebRTC demo page
>> https://webrtc.freeswitch.org/verto/index.html#page-main
>> and on the affected computer the problem was still there - no audio.
>>
>> 5. This happened both on an old FS version from last May and on a very
>> recent master FreeSWITCH Version
>> 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
>> 16:44:11Z 64bit)
>>
>> Any ideas?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/f9cb1d8f/attachment.html 

From carpenter.john at gmail.com  Mon Mar 23 21:53:34 2015
From: carpenter.john at gmail.com (John Carpenter)
Date: Mon, 23 Mar 2015 18:53:34 +0000
Subject: [Freeswitch-users] Problem with LUA dbh:query in Freeswitch
In-Reply-To: <CAFmZFiPeeAgYjdq4g4MxvOQPbdCDu67_aKEfhb0Y14jH0p0pFw@mail.gmail.com>
References: <CAD5xgxwmUEdmR08aT7b1aNpfybO=eOgW+kPk+xNqGO_gLKPOXg@mail.gmail.com>
	<CAFiqYukYj2+u4uJR+cw0cAXX6iEDdBOKoS2Ns5PAnrWksCfZSQ@mail.gmail.com>
	<CAD5xgxxzoDu6TCHebNE-932FtGsekYemFgY3=vPnVNoQqOY3Rw@mail.gmail.com>
	<009001d06561$b6c5bdf0$245139d0$@botecomm.com>
	<CAJR7f2FdGRYrQKJjZPatFXE-Ohf0xUN+E48ZQcR21PscO0sxKA@mail.gmail.com>
	<CAFmZFiPeeAgYjdq4g4MxvOQPbdCDu67_aKEfhb0Y14jH0p0pFw@mail.gmail.com>
Message-ID: <CAD5xgxy1-QgpSuN7dCq6ckA4Fp+6Kq5QALXqt=mLWyOztvUs=g@mail.gmail.com>

Thanks for all the other input but the resolution was to ditch ODBC and use
pgsql// dsn. With ODBC I get 3 decimal places, with pgsql dsb I get the
required 4. I think its a bit faster too.

On 23 March 2015 at 18:40, Abaci B <abaci64 at gmail.com> wrote:

> I'm doing something very similar, this is what I use:
>
> local rate = string.format("%.4f", currency]["rate"])
>
>
> On Mon, Mar 23, 2015 at 8:45 AM, ?talo Rossi <italorossib at gmail.com>
> wrote:
>
>> Yep, it's available and defaults to 6 decimal places, but looks like it's
>> not a Lua issue...
>>
>> On Mon, Mar 23, 2015 at 9:05 AM, Bote Man <bote_radio at botecomm.com>
>> wrote:
>>
>>> I?m not much of a programmer, but can?t you specify a format to %f to
>>> force it to return 4 decimal places?
>>>
>>>
>>>
>>> Or is that not available in Lua?
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>> *From:* John Carpenter
>>> *Sent:* Monday, 23 March, 2015 06:59
>>> *Subject:* Re: [Freeswitch-users] Problem with LUA dbh:query in
>>> Freeswitch
>>>
>>>
>>>
>>> Thats it, changed connection to a pgsql:// dsn and I get four decimal
>>> places, changed it back to ODBC and back to 3 decimal places.
>>>
>>> Thanks very much, been driving me mad
>>>
>>>
>>>
>>> On 23 March 2015 at 10:23, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Have you tried a pgsql:// dsn? In case the ODBC driver is having some
>>> truncating effect.
>>>
>>>
>>>
>>> On 22 March 2015 at 02:06, John Carpenter <carpenter.john at gmail.com>
>>> wrote:
>>>
>>> I am trying to use Lua to retrieve some currency conversion rates from a
>>> postgres database. It works but the numeric field value is only returning
>>> the first three decimal places although if I list the data in postgres it
>>> has four decimal places. I may be doing something daft as I am quite new to
>>> Lua, sample code is below. Any ideas appreciated
>>>
>>> regards, John
>>>
>>>
>>>
>>>
>>>
>>> *local* dbh = freeswitch.Dbh(*"odbc://fs_odbc:fs_user:password"*)
>>> *if* dbh:connected() == *false* *then*
>>>  freeswitch.consoleLog(*"notice"*, *"dp.lua cannot connect to database"*
>>> .. dsn .. *"\n"*)
>>>  *return*
>>> *end*
>>> *assert*(dbh:connected())
>>>
>>> e_rate = *1.123456789*
>>> *-- just a test* to see decimal places
>>> freeswitch.consoleLog(*"info"*,*string.format*(*" e_rate = %f"*,
>>> e_rate))
>>>
>>> *local* *function* eur2usd()
>>>    *local* sql = *"select eur2usd from currency order by id desc limit
>>> 1"*
>>>    dbh:query(sql, *function*(row)
>>>        e_rate = row.eur2usd
>>>        *end*)
>>> *end*
>>>
>>> *local* *function* eur2gbp()
>>>    *local* sql = *"select eur2gbp from currency order by id desc limit
>>> 1"*
>>>    dbh:query(sql, *function*(row)
>>>        e_rate = row.eur2gbp
>>>        *end*)
>>> *end*
>>>
>>> eur2usd()
>>> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2usd = %f"*,
>>> e_rate))
>>> eur2gbp()
>>> freeswitch.consoleLog(*"info"*,*string.format*(*" euro2gbp = %f"*,
>>> e_rate))
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/a4e22a56/attachment-0001.html 

From victor.chukalovskiy at gmail.com  Mon Mar 23 22:06:59 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Mon, 23 Mar 2015 15:06:59 -0400
Subject: [Freeswitch-users] PDD SIP timer in freeswitch?
Message-ID: <55106453.4060809@gmail.com>

Good day,

Is there a timer in FS that can be set before SIP bridge so that it 
limits max duration before call gets 180/183/200 response?
Something equivalent to timer T310 in PRI terms.

If such timer is not implemented, how people are managing to re-route in 
fail-over / LCR scenarios where a given carrier is taking too long after 
"100 Trying"?
call_timeout does not seem to be the right thing.

Thanks in advance!
-Victor


From italorossib at gmail.com  Mon Mar 23 22:11:44 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 23 Mar 2015 16:11:44 -0300
Subject: [Freeswitch-users] PDD SIP timer in freeswitch?
In-Reply-To: <55106453.4060809@gmail.com>
References: <55106453.4060809@gmail.com>
Message-ID: <CAJR7f2HGsYv-KCRRe7uL1zCen4rU4XQWHsgWwDYXpED0Ynq-gw@mail.gmail.com>

Try
https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-leg_timeout

On Mon, Mar 23, 2015 at 4:06 PM, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

> Good day,
>
> Is there a timer in FS that can be set before SIP bridge so that it
> limits max duration before call gets 180/183/200 response?
> Something equivalent to timer T310 in PRI terms.
>
> If such timer is not implemented, how people are managing to re-route in
> fail-over / LCR scenarios where a given carrier is taking too long after
> "100 Trying"?
> call_timeout does not seem to be the right thing.
>
> Thanks in advance!
> -Victor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/73e38f20/attachment.html 

From victor.chukalovskiy at gmail.com  Mon Mar 23 22:15:28 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Mon, 23 Mar 2015 15:15:28 -0400
Subject: [Freeswitch-users] PDD SIP timer in freeswitch?
In-Reply-To: <CAJR7f2HGsYv-KCRRe7uL1zCen4rU4XQWHsgWwDYXpED0Ynq-gw@mail.gmail.com>
References: <55106453.4060809@gmail.com>
	<CAJR7f2HGsYv-KCRRe7uL1zCen4rU4XQWHsgWwDYXpED0Ynq-gw@mail.gmail.com>
Message-ID: <55106650.1000304@gmail.com>

Thx!

You mean leg_progress_timeout?

On 15-03-23 03:11 PM, ?talo Rossi wrote:
> Try 
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-leg_timeout
>
> On Mon, Mar 23, 2015 at 4:06 PM, Victor Chukalovskiy 
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>> 
> wrote:
>
>     Good day,
>
>     Is there a timer in FS that can be set before SIP bridge so that it
>     limits max duration before call gets 180/183/200 response?
>     Something equivalent to timer T310 in PRI terms.
>
>     If such timer is not implemented, how people are managing to
>     re-route in
>     fail-over / LCR scenarios where a given carrier is taking too long
>     after
>     "100 Trying"?
>     call_timeout does not seem to be the right thing.
>
>     Thanks in advance!
>     -Victor
>
>     _________________________________________________________________________
>     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://confluence.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
>
>
>
>
> -- 
> ?talo Rossi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/70ade9e2/attachment.html 

From italorossib at gmail.com  Mon Mar 23 22:18:18 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 23 Mar 2015 16:18:18 -0300
Subject: [Freeswitch-users] PDD SIP timer in freeswitch?
In-Reply-To: <55106650.1000304@gmail.com>
References: <55106453.4060809@gmail.com>
	<CAJR7f2HGsYv-KCRRe7uL1zCen4rU4XQWHsgWwDYXpED0Ynq-gw@mail.gmail.com>
	<55106650.1000304@gmail.com>
Message-ID: <CAJR7f2GG9GLjmQy7OCsEpT_XJTps-ccFSYEn0C7F24jPoGg_1w@mail.gmail.com>

If you're not ignoring early media, yes.

On Mon, Mar 23, 2015 at 4:15 PM, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

>  Thx!
>
> You mean leg_progress_timeout?
>
>  On 15-03-23 03:11 PM, ?talo Rossi wrote:
>
> Try
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-leg_timeout
>
> On Mon, Mar 23, 2015 at 4:06 PM, Victor Chukalovskiy <
> victor.chukalovskiy at gmail.com> wrote:
>
>> Good day,
>>
>> Is there a timer in FS that can be set before SIP bridge so that it
>> limits max duration before call gets 180/183/200 response?
>> Something equivalent to timer T310 in PRI terms.
>>
>> If such timer is not implemented, how people are managing to re-route in
>> fail-over / LCR scenarios where a given carrier is taking too long after
>> "100 Trying"?
>> call_timeout does not seem to be the right thing.
>>
>> Thanks in advance!
>> -Victor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>  --
>  ?talo Rossi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/a1008009/attachment.html 

From victor.chukalovskiy at gmail.com  Mon Mar 23 22:30:35 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Mon, 23 Mar 2015 15:30:35 -0400
Subject: [Freeswitch-users] PDD SIP timer in freeswitch?
In-Reply-To: <CAJR7f2GG9GLjmQy7OCsEpT_XJTps-ccFSYEn0C7F24jPoGg_1w@mail.gmail.com>
References: <55106453.4060809@gmail.com>	<CAJR7f2HGsYv-KCRRe7uL1zCen4rU4XQWHsgWwDYXpED0Ynq-gw@mail.gmail.com>	<55106650.1000304@gmail.com>
	<CAJR7f2GG9GLjmQy7OCsEpT_XJTps-ccFSYEn0C7F24jPoGg_1w@mail.gmail.com>
Message-ID: <551069DB.7060503@gmail.com>

Sounds promising, thank you. As long as this timer also stops upon plain 
"180 Ringing", should be good
Will test and update the thread


On 15-03-23 03:18 PM, ?talo Rossi wrote:
> If you're not ignoring early media, yes.
>
> On Mon, Mar 23, 2015 at 4:15 PM, Victor Chukalovskiy 
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>> 
> wrote:
>
>     Thx!
>
>     You mean leg_progress_timeout?
>
>     On 15-03-23 03:11 PM, ?talo Rossi wrote:
>>     Try
>>     https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-leg_timeout
>>
>>     On Mon, Mar 23, 2015 at 4:06 PM, Victor Chukalovskiy
>>     <victor.chukalovskiy at gmail.com
>>     <mailto:victor.chukalovskiy at gmail.com>> wrote:
>>
>>         Good day,
>>
>>         Is there a timer in FS that can be set before SIP bridge so
>>         that it
>>         limits max duration before call gets 180/183/200 response?
>>         Something equivalent to timer T310 in PRI terms.
>>
>>         If such timer is not implemented, how people are managing to
>>         re-route in
>>         fail-over / LCR scenarios where a given carrier is taking too
>>         long after
>>         "100 Trying"?
>>         call_timeout does not seem to be the right thing.
>>
>>         Thanks in advance!
>>         -Victor
>>
>>         _________________________________________________________________________
>>         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://confluence.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
>>
>>
>>
>>
>>     -- 
>>     ?talo Rossi
>>
>>
>>     _________________________________________________________________________
>>     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://confluence.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://confluence.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
>
>
>
>
> -- 
> ?talo Rossi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/47260b6e/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Mon Mar 23 22:37:09 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 23 Mar 2015 19:37:09 +0000
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
	<CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
Message-ID: <CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>

I am having similar issues with Sonus and occasional packet loss etc. I am using not the latest and greatest FS but was updated last Nov 2014.

The wiki states

?Note that the version of FreeSWITCH which has this patch auto-detects Sonus, so you don't need to configure anything?

Does it really apply *all* of the fixes for SONUS or only the DTMF fix.

I was considering using the below param tailored for Sonus bugs.Do these have any effect ? or are they deprecated ?


<param name="manual-rtp-bugs" value=" SONUS_SEND_INVALID_TIMESTAMP_2833
|SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER|IGNORE_DTMF_DURATION| |GEN_ONE_GEN_ALL"/>

And clear the auto-detection param.

Thanks
Sharath


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge
Sent: Friday, March 20, 2015 4:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF missing digits

Aron its not that you're doing inband suppression, Sonus is buggy and if its in your media path it can cause lots of headaches.  Many carriers use Sonus, some as just SBC's and some that handle media.

On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com<mailto:aronp at guaranteedplus.com>> wrote:
I'm sending rfc2833, I don't do any inband suppression on the channel .

On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
how do you send DTMF to sonus? info or rfc2833?


On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
<aronp at guaranteedplus.com<mailto:aronp at guaranteedplus.com>> wrote:
> I am experiencing problems with sonus dtmf. I tried setting different
> parameters related to the RTP BUGS listed in switch_types.h but it did not
> seem to fix the problem.
>
> I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>
> freeswitch sends the following digits to our carrier, and is received by the
> other end (also a freeswitch) missing the digits marked in black.
> pcap shows the missing digits as being part of the previous packet, so two
> 1's becomes one 1 with the duration sum of two.
>
>  1   -    1   -    2   -    1   -    1   -    #
>
> What are the relevant parameters I should set?
>
> _________________________________________________________________________
> 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://confluence.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://confluence.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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/e7cf75ac/attachment.html 

From mike at jerris.com  Mon Mar 23 23:52:00 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 23 Mar 2015 16:52:00 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
	<CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
	<CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <BD303FF5-3B4D-4773-A410-6B163F09557E@jerris.com>

Some sonus settings are VERY specific to specific sonus firmware and will completely break talking to others.  In fact, for the same reasons, some versions of sonus firmware will break when talking to different versions of sonus firmware.


> On Mar 23, 2015, at 3:37 PM, Sharath Kumar <Sharath.Kumar at meZocliq.com> wrote:
> 
> I am having similar issues with Sonus and occasional packet loss etc. I am using not the latest and greatest FS but was updated last Nov 2014.
>  
> The wiki states
> ?Note that the version of FreeSWITCH which has this patch auto-detects Sonus, so you don't need to configure anything?
>  
> Does it really apply *all* of the fixes for SONUS or only the DTMF fix.
>  
> I was considering using the below param tailored for Sonus bugs.Do these have any effect ? or are they deprecated ?
>  
> <param name="manual-rtp-bugs" value=" SONUS_SEND_INVALID_TIMESTAMP_2833
> |SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER|IGNORE_DTMF_DURATION| |GEN_ONE_GEN_ALL"/>
>  
> And clear the auto-detection param.
>  
> Thanks
> Sharath
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Tunbridge
> Sent: Friday, March 20, 2015 4:22 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] DTMF missing digits
>  
> Aron its not that you're doing inband suppression, Sonus is buggy and if its in your media path it can cause lots of headaches.  Many carriers use Sonus, some as just SBC's and some that handle media.
>  
> On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com <mailto:aronp at guaranteedplus.com>> wrote:
> I'm sending rfc2833, I don't do any inband suppression on the channel .
>  
> On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> how do you send DTMF to sonus? info or rfc2833?
> 
> 
> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
> <aronp at guaranteedplus.com <mailto:aronp at guaranteedplus.com>> wrote:
> > I am experiencing problems with sonus dtmf. I tried setting different
> > parameters related to the RTP BUGS listed in switch_types.h but it did not
> > seem to fix the problem.
> >
> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
> >
> > freeswitch sends the following digits to our carrier, and is received by the
> > other end (also a freeswitch) missing the digits marked in black.
> > pcap shows the missing digits as being part of the previous packet, so two
> > 1's becomes one 1 with the duration sum of two.
> >
> >  1   -    1   -    2   -    1   -    1   -    #
> >
> > What are the relevant parameters I should set?
> >
> > _________________________________________________________________________
> > 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/ea165b18/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Tue Mar 24 00:16:30 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 23 Mar 2015 21:16:30 +0000
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <BD303FF5-3B4D-4773-A410-6B163F09557E@jerris.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
	<CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
	<CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<BD303FF5-3B4D-4773-A410-6B163F09557E@jerris.com>
Message-ID: <CY1PR0501MB1098E4686FBDD3920CA8CD49840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>

That is not a good thing. So, if we are using freeswitch and have intermittent audio issues with PSTN calls, I guess there is no real solution except to change the carrier ?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Monday, March 23, 2015 4:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF missing digits

Some sonus settings are VERY specific to specific sonus firmware and will completely break talking to others.  In fact, for the same reasons, some versions of sonus firmware will break when talking to different versions of sonus firmware.


On Mar 23, 2015, at 3:37 PM, Sharath Kumar <Sharath.Kumar at meZocliq.com<mailto:Sharath.Kumar at meZocliq.com>> wrote:

I am having similar issues with Sonus and occasional packet loss etc. I am using not the latest and greatest FS but was updated last Nov 2014.

The wiki states
?Note that the version of FreeSWITCH which has this patch auto-detects Sonus, so you don't need to configure anything?

Does it really apply *all* of the fixes for SONUS or only the DTMF fix.

I was considering using the below param tailored for Sonus bugs.Do these have any effect ? or are they deprecated ?


<param name="manual-rtp-bugs" value=" SONUS_SEND_INVALID_TIMESTAMP_2833
|SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER|IGNORE_DTMF_DURATION| |GEN_ONE_GEN_ALL"/>

And clear the auto-detection param.

Thanks
Sharath


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 Chris Tunbridge
Sent: Friday, March 20, 2015 4:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF missing digits

Aron its not that you're doing inband suppression, Sonus is buggy and if its in your media path it can cause lots of headaches.  Many carriers use Sonus, some as just SBC's and some that handle media.

On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com<mailto:aronp at guaranteedplus.com>> wrote:
I'm sending rfc2833, I don't do any inband suppression on the channel .

On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com<mailto:ssinyagin at gmail.com>> wrote:
how do you send DTMF to sonus? info or rfc2833?


On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
<aronp at guaranteedplus.com<mailto:aronp at guaranteedplus.com>> wrote:
> I am experiencing problems with sonus dtmf. I tried setting different
> parameters related to the RTP BUGS listed in switch_types.h but it did not
> seem to fix the problem.
>
> I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>
> freeswitch sends the following digits to our carrier, and is received by the
> other end (also a freeswitch) missing the digits marked in black.
> pcap shows the missing digits as being part of the previous packet, so two
> 1's becomes one 1 with the duration sum of two.
>
>  1   -    1   -    2   -    1   -    1   -    #
>
> What are the relevant parameters I should set?
>
> _________________________________________________________________________
> 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://confluence.freeswitch.org<http://confluence.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://confluence.freeswitch.org<http://confluence.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://confluence.freeswitch.org<http://confluence.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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.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>
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/1c2d69d0/attachment-0001.html 

From brian at freeswitch.org  Tue Mar 24 01:05:32 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 23 Mar 2015 17:05:32 -0500
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CY1PR0501MB1098E4686FBDD3920CA8CD49840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
	<CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
	<CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<BD303FF5-3B4D-4773-A410-6B163F09557E@jerris.com>
	<CY1PR0501MB1098E4686FBDD3920CA8CD49840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAEJMVkDP+2HYkT+XdOSRCqCiXMw2P6G+Xi5miSGLfbF52B2xOw@mail.gmail.com>

The rubix cube of rtp_bugs may help, I can almost bet IGNORE_DTMF_DURATION
and GEN_ONE_GEN_ALL may be all you.

On Mon, Mar 23, 2015 at 4:16 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  That is not a good thing. So, if we are using freeswitch and have
> intermittent audio issues with PSTN calls, I guess there is no real
> solution except to change the carrier ?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Monday, March 23, 2015 4:52 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] DTMF missing digits
>
>
>
> Some sonus settings are VERY specific to specific sonus firmware and will
> completely break talking to others.  In fact, for the same reasons, some
> versions of sonus firmware will break when talking to different versions of
> sonus firmware.
>
>
>
>
>
>  On Mar 23, 2015, at 3:37 PM, Sharath Kumar <Sharath.Kumar at meZocliq.com>
> wrote:
>
>
>
> I am having similar issues with Sonus and occasional packet loss etc. I am
> using not the latest and greatest FS but was updated last Nov 2014.
>
>
>
> The wiki states
>
> ?Note that the version of FreeSWITCH which has this patch auto-detects
> Sonus, so you don't need to configure anything?
>
>
>
> Does it really apply **all** of the fixes for SONUS or only the DTMF fix.
>
>
>
> I was considering using the below param tailored for Sonus bugs.Do these
> have any effect ? or are they deprecated ?
>
>
>
> <param name="manual-rtp-bugs" value=" SONUS_SEND_INVALID_TIMESTAMP_2833
>
>  |SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER|IGNORE_DTMF_DURATION|
> |GEN_ONE_GEN_ALL"/>
>
>
>
> And clear the auto-detection param.
>
>
>
> Thanks
>
> Sharath
>
>
>
>
>
> *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 *Chris
> Tunbridge
> *Sent:* Friday, March 20, 2015 4:22 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] DTMF missing digits
>
>
>
> Aron its not that you're doing inband suppression, Sonus is buggy and if
> its in your media path it can cause lots of headaches.  Many carriers use
> Sonus, some as just SBC's and some that handle media.
>
>
>
> On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com>
> wrote:
>
>  I'm sending rfc2833, I don't do any inband suppression on the channel .
>
>
>
> On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>  how do you send DTMF to sonus? info or rfc2833?
>
>
>
> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
> <aronp at guaranteedplus.com> wrote:
> > I am experiencing problems with sonus dtmf. I tried setting different
> > parameters related to the RTP BUGS listed in switch_types.h but it did
> not
> > seem to fix the problem.
> >
> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
> >
> > freeswitch sends the following digits to our carrier, and is received by
> the
> > other end (also a freeswitch) missing the digits marked in black.
> > pcap shows the missing digits as being part of the previous packet, so
> two
> > 1's becomes one 1 with the duration sum of two.
> >
> >  1   -    1   -    2   -    1   -    1   -    #
> >
> > What are the relevant parameters I should set?
> >
>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/2ea7d460/attachment.html 

From aronp at guaranteedplus.com  Tue Mar 24 03:07:25 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Mon, 23 Mar 2015 20:07:25 -0400
Subject: [Freeswitch-users] DTMF missing digits
In-Reply-To: <CAEJMVkDP+2HYkT+XdOSRCqCiXMw2P6G+Xi5miSGLfbF52B2xOw@mail.gmail.com>
References: <CANJp-yipyQjs3QyqZBFVCULZ-+OD+kLCBChukOdsi3bUiqfa3w@mail.gmail.com>
	<CACOYK=gjGaw7A=y1d9G=0+-0tPFDs+utf=-iwZuvW9vULPko3g@mail.gmail.com>
	<CANJp-yj6nD_FLuUNpC54eSt3TBqmW7jaVBb6DSPKdEPrDLiHNg@mail.gmail.com>
	<CAO3hyzEMZ3sQid=RpsSUgZzDY7qRMe0SmPm2TZNFjeqbWYVrdA@mail.gmail.com>
	<CY1PR0501MB1098E9C568ADC4DE84FBC100840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<BD303FF5-3B4D-4773-A410-6B163F09557E@jerris.com>
	<CY1PR0501MB1098E4686FBDD3920CA8CD49840D0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAEJMVkDP+2HYkT+XdOSRCqCiXMw2P6G+Xi5miSGLfbF52B2xOw@mail.gmail.com>
Message-ID: <CANJp-ygwmcneQUoYB0cgpPwVFtUOPG+BjA_Sd3+sMmm-AyJNFQ@mail.gmail.com>

Setting the duration seems to solve the problem for me.

<param name="default-dtmf-duration" value="800"/>


On Mon, Mar 23, 2015 at 6:05 PM, Brian West <brian at freeswitch.org> wrote:

> The rubix cube of rtp_bugs may help, I can almost bet IGNORE_DTMF_DURATION
> and GEN_ONE_GEN_ALL may be all you.
>
> On Mon, Mar 23, 2015 at 4:16 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com
> > wrote:
>
>>  That is not a good thing. So, if we are using freeswitch and have
>> intermittent audio issues with PSTN calls, I guess there is no real
>> solution except to change the carrier ?
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Jerris
>> *Sent:* Monday, March 23, 2015 4:52 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] DTMF missing digits
>>
>>
>>
>> Some sonus settings are VERY specific to specific sonus firmware and will
>> completely break talking to others.  In fact, for the same reasons, some
>> versions of sonus firmware will break when talking to different versions of
>> sonus firmware.
>>
>>
>>
>>
>>
>>  On Mar 23, 2015, at 3:37 PM, Sharath Kumar <Sharath.Kumar at meZocliq.com>
>> wrote:
>>
>>
>>
>> I am having similar issues with Sonus and occasional packet loss etc. I
>> am using not the latest and greatest FS but was updated last Nov 2014.
>>
>>
>>
>> The wiki states
>>
>> ?Note that the version of FreeSWITCH which has this patch auto-detects
>> Sonus, so you don't need to configure anything?
>>
>>
>>
>> Does it really apply **all** of the fixes for SONUS or only the DTMF fix.
>>
>>
>>
>> I was considering using the below param tailored for Sonus bugs.Do these
>> have any effect ? or are they deprecated ?
>>
>>
>>
>> <param name="manual-rtp-bugs" value=" SONUS_SEND_INVALID_TIMESTAMP_2833
>>
>>  |SEND_LINEAR_TIMESTAMPS|START_SEQ_AT_ZERO|NEVER_SEND_MARKER|IGNORE_DTMF_DURATION|
>> |GEN_ONE_GEN_ALL"/>
>>
>>
>>
>> And clear the auto-detection param.
>>
>>
>>
>> Thanks
>>
>> Sharath
>>
>>
>>
>>
>>
>> *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 *Chris
>> Tunbridge
>> *Sent:* Friday, March 20, 2015 4:22 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] DTMF missing digits
>>
>>
>>
>> Aron its not that you're doing inband suppression, Sonus is buggy and if
>> its in your media path it can cause lots of headaches.  Many carriers use
>> Sonus, some as just SBC's and some that handle media.
>>
>>
>>
>> On Fri, Mar 20, 2015 at 1:43 PM, Podrigal, Aron <aronp at guaranteedplus.com>
>> wrote:
>>
>>  I'm sending rfc2833, I don't do any inband suppression on the channel .
>>
>>
>>
>> On Fri, Mar 20, 2015 at 6:33 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>  how do you send DTMF to sonus? info or rfc2833?
>>
>>
>>
>> On Thu, Mar 19, 2015 at 5:10 PM, Podrigal, Aron
>> <aronp at guaranteedplus.com> wrote:
>> > I am experiencing problems with sonus dtmf. I tried setting different
>> > parameters related to the RTP BUGS listed in switch_types.h but it did
>> not
>> > seem to fix the problem.
>> >
>> > I have freeswitch ----> [carrier(sonus), ...]  -----> freeswitch
>> >
>> > freeswitch sends the following digits to our carrier, and is received
>> by the
>> > other end (also a freeswitch) missing the digits marked in black.
>> > pcap shows the missing digits as being part of the previous packet, so
>> two
>> > 1's becomes one 1 with the duration sum of two.
>> >
>> >  1   -    1   -    2   -    1   -    1   -    #
>> >
>> > What are the relevant parameters I should set?
>> >
>>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150323/52c71629/attachment-0001.html 

From gabe at gundy.org  Tue Mar 24 07:51:39 2015
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 23 Mar 2015 22:51:39 -0600
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
In-Reply-To: <5509df43323ec_f3fe8bd32881237@resque-worker-ip-10-101-150-189.mail>
References: <5509df43323ec_f3fe8bd32881237@resque-worker-ip-10-101-150-189.mail>
Message-ID: <CAMwZ-t+Ri_EJj+_RBj-BSsJqqHDqWy1vr++tSpPM9H1J9B0KyA@mail.gmail.com>

The URL:

https://www.cluecon.com/speaking-proposals

...is referenced several times, but it's 404ing.

Best,
Gabe

On Wed, Mar 18, 2015 at 2:25 PM, Ken Rice <krice at freeswitch.org> wrote:
> New Post on freeswitch.org from krice387
> check it out at http://ift.tt/1GpR0Yu
> ClueCon Call For Speakers is Open!
>
> Call for speakers at ClueCon 2015 is open!
>
> Visit http://ift.tt/1FBM7gN to submit your speaking proposal Below!
>
> What makes a great ClueCon presentation? The tech savvy crowd that attends
> ClueCon loves technical presentations. In general, the more technical the
> presentation, the better.
>
> If you are thinking about a presentation then consider these points:
> * ClueCon talks are 30 minutes in length, including Q&A time with the
> audience (Lightning Talks are also open! 10 minute talks, 5 minutes Q&A)
> * ClueCon has a special focus on open source communications, VoIP, and
> telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and Kamailio
> * Attendees enjoy hearing about projects built with open source tools,
> especially those in a production environment
> * Highly technical discussions that show the nuts and bolts are especially
> well-liked
> * The audience appreciates seeing and participating in live demonstrations
>
> We are especially interested in Security-related talks and demonstrations
>
> Please register your proposals at http://ift.tt/1FBM7gN. To complete
> registration you need the following items:
> * Working title
> * Brief description of the talk (abstract)
> * Name of the presenter(s)
> * Bio and headshot of presenter(s)
> * Presenter?s contact information (including mobile phone the presenter will
> have with them at the conference)
>
> Don?t delay! Speaking proposals must be in by July 4, 2015 and scheduling
> requests are handled on a first come first serve basis.
>
> ClueCon 2015: See You There!
> For more details on ClueCon see https://cluecon.com.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From idokan at gmail.com  Tue Mar 24 08:04:24 2015
From: idokan at gmail.com (ik)
Date: Tue, 24 Mar 2015 07:04:24 +0200
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and WebRTC
In-Reply-To: <00bb01d06599$2d2d75a0$878860e0$@botecomm.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
	<00bb01d06599$2d2d75a0$878860e0$@botecomm.com>
Message-ID: <CAOccsKCuBBNToFt3p49sAquBv4qJwuM-zNiPFUECcE-i6zFYHA@mail.gmail.com>

There was a security update last weekend for Chrome, they might have solved
this and placed it on that version.

Ido
On Mar 23, 2015 8:45 PM, "Bote Man" <bote_radio at botecomm.com> wrote:

> I am running
>
> Chrome Version 41.0.2272.101 m
>
> on Windows 7 and checked the 888 FreeSWITCH conference yesterday, it had
> audio both directions. I tried it 1 or 2 weeks ago and it had no audio so
> it must have updated itself behind my back because I did not explicitly
> update Chrome.
>
>
>
> Firefox 36.0.1 did not have any audio last week and now 36.0.4 has no
> audio either.
>
>
>
> I am testing this on my home network using a Cisco RV042 router and the
> default checkboxes on the FS WebRTC page which apparently default to using
> STUN.
>
>
>
> Anthm has stated many times that WebRTC is a moving target so enjoy the
> exhilaration of these rapid developments! I believe this is why he insists
> that WebRTC users keep FreeSWITCH updated so that it tracks the changes in
> the major web browsers.
>
>
>
> Bote
>
>
>
>
>
>
>
> *From:* Oleg Stolyar
> *Sent:* Monday, 23 March, 2015 14:21
> *Subject:* [Freeswitch-users] Intermittent audio issue with Chrome 40-42
> and WebRTC
>
>
>
> Hi guys,
>
>
>
> Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
> then they complain that they cannot hear audio from FreeSWITCH or send
> audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
> several occurrences.  It happens on both Windows and Macs.  Here are some
> details"
>
>
>
> 1. Usually the problem is solved if we close Chrome and make sure all the
> Chrome processes are killed.
>
>
>
> 2. When the problem happens, FS logs this on connection:
>
> [ERR] switch_rtp.c:2976 audio DTLS packet not written
>
>
>
> 3. I tried using this WebRTC test on an affected computer:
>
> https://janus.conf.meetecho.com/echotest.html
>
> and that worked, so it is not a general WebRTC issue.
>
>
>
> 4. I tried FS WebRTC demo page
>
> https://webrtc.freeswitch.org/verto/index.html#page-main
>
> and on the affected computer the problem was still there - no audio.
>
>
>
> 5. This happened both on an old FS version from last May and on a very
> recent master FreeSWITCH Version
> 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
> 16:44:11Z 64bit)
>
>
>
> Any ideas?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/c7e3e836/attachment.html 

From max at nysolutions.com  Tue Mar 24 08:13:03 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 24 Mar 2015 05:13:03 +0000
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
In-Reply-To: <CAMwZ-t+Ri_EJj+_RBj-BSsJqqHDqWy1vr++tSpPM9H1J9B0KyA@mail.gmail.com>
References: <5509df43323ec_f3fe8bd32881237@resque-worker-ip-10-101-150-189.mail>
	<CAMwZ-t+Ri_EJj+_RBj-BSsJqqHDqWy1vr++tSpPM9H1J9B0KyA@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D39FC8@ex2.connex.local>

https://www.cluecon.com/call-for-speakers/ 

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 

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 Gabriel Gunderson
Sent: Tuesday, March 24, 2015 12:52 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ClueCon Call For Speakers is Open!

The URL:

https://www.cluecon.com/speaking-proposals

...is referenced several times, but it's 404ing.

Best,
Gabe

On Wed, Mar 18, 2015 at 2:25 PM, Ken Rice <krice at freeswitch.org> wrote:
> New Post on freeswitch.org from krice387 check it out at 
> http://ift.tt/1GpR0Yu ClueCon Call For Speakers is Open!
>
> Call for speakers at ClueCon 2015 is open!
>
> Visit http://ift.tt/1FBM7gN to submit your speaking proposal Below!
>
> What makes a great ClueCon presentation? The tech savvy crowd that 
> attends ClueCon loves technical presentations. In general, the more 
> technical the presentation, the better.
>
> If you are thinking about a presentation then consider these points:
> * ClueCon talks are 30 minutes in length, including Q&A time with the 
> audience (Lightning Talks are also open! 10 minute talks, 5 minutes 
> Q&A)
> * ClueCon has a special focus on open source communications, VoIP, and 
> telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and 
> Kamailio
> * Attendees enjoy hearing about projects built with open source tools, 
> especially those in a production environment
> * Highly technical discussions that show the nuts and bolts are 
> especially well-liked
> * The audience appreciates seeing and participating in live 
> demonstrations
>
> We are especially interested in Security-related talks and 
> demonstrations
>
> Please register your proposals at http://ift.tt/1FBM7gN. To complete 
> registration you need the following items:
> * Working title
> * Brief description of the talk (abstract)
> * Name of the presenter(s)
> * Bio and headshot of presenter(s)
> * Presenter?s contact information (including mobile phone the 
> presenter will have with them at the conference)
>
> Don?t delay! Speaking proposals must be in by July 4, 2015 and 
> scheduling requests are handled on a first come first serve basis.
>
> ClueCon 2015: See You There!
> For more details on ClueCon see https://cluecon.com.
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users 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://confluence.freeswitch.org
http://www.cluecon.com

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

From mkvonarx at gmail.com  Tue Mar 24 10:04:00 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Tue, 24 Mar 2015 08:04:00 +0100
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
Message-ID: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>

Hi

Can anyone tell me if FreeSWITCH supports silence suppression for SIP calls
that are inside a FreeSWITCH audio conference? If yes, how do I configure
mod_conference, mod_sofia and FreeSWITCH core to enable this feature?

More precisely, I try to enable/activate the behavior described in RFC 3389
for G.711 in such a way that there are only RTP packets of type 13 every 1
or 2 seconds. I tried to play around with some possible settings but could
never observe anything else then the regular G.711 PCMU RTP packets on the
wire. Even when I set the SIP call to 'deaf' via the FreeSWITCH console,
mod_conference/mod_sofia continue to send G.711 PCMU RTP packets every 20ms.

It's possible that I completly misunderstand RFC 3389 and the concepts of
silence suppression, comfort noise etc. In the end, what I try
to achieve is to reduce the network bandwidth of a G.711 SIP channel during
the periods when the FreeSWITCH only sends silence over the SIP channel.
Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to
another codec.

Thanks, Markus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/22872675/attachment-0001.html 

From fernando at softov.com.br  Tue Mar 24 10:36:53 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Tue, 24 Mar 2015 03:36:53 -0400
Subject: [Freeswitch-users] Transfering Attended Call
Message-ID: <CAKy26iPjxKBZ0Thj3v4eXuVCGDqpA0uRWcAjbJODXpSsO_mzhQ@mail.gmail.com>

Hi... First of all i make a software with freeswitch and FreeBSD, named
Voipr...

voipr.brbyte.com

It's a softswitch with Billing, Redirect, IVR, Routes, GSM, Gateway and
other things.

I am not using dialplan, just a outbound socket, with my own ESL Server
written in C.
I parse the events, and send comands back to make things. Like bridge, play
some music, hangup the call....

<extension name="VoiprExt">
    <condition>
        <action application="socket" data="127.0.0.1:3012 async full"/>
        <action application="hangup"/>
    </condition>
</extension>

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

My question, is how to make a transfer of a call...

Like A (1001) call to B (1002).

Then i make a BRIDGE between extensions

"execute", "set", "dialed_extension=1002"
"execute", "export", "dialed_extension=1002"
"execute", "set", "call_timeout=30"
"execute", "set", "continue_on_fail=true"
"execute", "set", "park_after_bridge=true"
"execute", "bridge", "user/${dialed_extension}"

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

Let's say B (1002) is a secretary who want to transfer the call to C (1003)

B (1002), press the key [*], and i play a music, like "enter the desired
number and #".

B (1002), press the keys 1003#

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

At this point, I have all information, and want to know how 3 things

Consider that: the call already bridged between A (1001) and B (1002)

- - - - -
1 - Making a call between B and C.
Then, when C answer B hangup. And the call return between A and C.


- - - - -
2 - Making a call between B and C.
Then, when C answer, B talks with C. B press * and B hangup.
And the call return between A and C


- - - - -
3 - Making a call between A and C.
Then, when C answer, A hangup.
And the call return to B and C


Thank you in advance, who can help me.

Sorry for my english.
If anyone need help with socket events, gsmopen or xml_curl send me a
e-mail. I will be happy to help.

Att,
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/820bb58d/attachment.html 

From steveayre at gmail.com  Tue Mar 24 13:43:31 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 24 Mar 2015 10:43:31 +0000
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
In-Reply-To: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>
References: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>
Message-ID: <CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>

https://wiki.freeswitch.org/wiki/VAD_and_CNG

On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:

> Hi
>
> Can anyone tell me if FreeSWITCH supports silence suppression for SIP
> calls that are inside a FreeSWITCH audio conference? If yes, how do I
> configure mod_conference, mod_sofia and FreeSWITCH core to enable this
> feature?
>
> More precisely, I try to enable/activate the behavior described in RFC
> 3389 for G.711 in such a way that there are only RTP packets of type 13
> every 1 or 2 seconds. I tried to play around with some possible settings
> but could never observe anything else then the regular G.711 PCMU RTP
> packets on the wire. Even when I set the SIP call to 'deaf' via the
> FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU
> RTP packets every 20ms.
>
> It's possible that I completly misunderstand RFC 3389 and the concepts of
> silence suppression, comfort noise etc. In the end, what I try
> to achieve is to reduce the network bandwidth of a G.711 SIP channel during
> the periods when the FreeSWITCH only sends silence over the SIP channel.
> Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to
> another codec.
>
> Thanks, Markus
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/dccc8d99/attachment.html 

From krice at freeswitch.org  Tue Mar 24 14:44:52 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 24 Mar 2015 06:44:52 -0500
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
In-Reply-To: <CAMwZ-t+Ri_EJj+_RBj-BSsJqqHDqWy1vr++tSpPM9H1J9B0KyA@mail.gmail.com>
Message-ID: <D136B864.991FB%krice@freeswitch.org>

Still getting that?


On 3/23/15, 11:51 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:

> The URL:

https://www.cluecon.com/speaking-proposals

...is referenced several
> times, but it's 404ing.

Best,
Gabe

On Wed, Mar 18, 2015 at 2:25 PM, Ken Rice
> <krice at freeswitch.org> wrote:
> New Post on freeswitch.org from krice387
>
> check it out at http://ift.tt/1GpR0Yu
> ClueCon Call For Speakers is Open!
>
>
> Call for speakers at ClueCon 2015 is open!
>
> Visit http://ift.tt/1FBM7gN to
> submit your speaking proposal Below!
>
> What makes a great ClueCon
> presentation? The tech savvy crowd that attends
> ClueCon loves technical
> presentations. In general, the more technical the
> presentation, the
> better.
>
> If you are thinking about a presentation then consider these
> points:
> * ClueCon talks are 30 minutes in length, including Q&A time with
> the
> audience (Lightning Talks are also open! 10 minute talks, 5 minutes
> Q&A)
> * ClueCon has a special focus on open source communications, VoIP,
> and
> telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and
> Kamailio
> * Attendees enjoy hearing about projects built with open source
> tools,
> especially those in a production environment
> * Highly technical
> discussions that show the nuts and bolts are especially
> well-liked
> * The
> audience appreciates seeing and participating in live demonstrations
>
> We
> are especially interested in Security-related talks and demonstrations
>
>
> Please register your proposals at http://ift.tt/1FBM7gN. To complete
>
> registration you need the following items:
> * Working title
> * Brief
> description of the talk (abstract)
> * Name of the presenter(s)
> * Bio and
> headshot of presenter(s)
> * Presenter?s contact information (including mobile
> phone the presenter will
> have with them at the conference)
>
> Don?t delay!
> Speaking proposals must be in by July 4, 2015 and scheduling
> requests are
> handled on a first come first serve basis.
>
> ClueCon 2015: See You There!
>
> For more details on ClueCon see https://cluecon.com.
>
>
>
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org

___________________________________________________
> ______________________
Professional FreeSWITCH Consulting Services:
> 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official
> FreeSWITCH 
> Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cl
> uecon.com

FreeSWITCH-users mailing
> list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman
> /listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt
> ions/freeswitch-users
http://www.freeswitch.org

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





From max at nysolutions.com  Tue Mar 24 15:34:59 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 24 Mar 2015 12:34:59 +0000
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
In-Reply-To: <D136B864.991FB%krice@freeswitch.org>
References: <CAMwZ-t+Ri_EJj+_RBj-BSsJqqHDqWy1vr++tSpPM9H1J9B0KyA@mail.gmail.com>
	<D136B864.991FB%krice@freeswitch.org>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D3AC2D@ex2.connex.local>

Ken this one is also broken https://www.cluecon.com/speaker-signup/ 

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 

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 Ken Rice
Sent: Tuesday, March 24, 2015 7:45 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ClueCon Call For Speakers is Open!

Still getting that?


On 3/23/15, 11:51 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:

> The URL:

https://www.cluecon.com/speaking-proposals

...is referenced several
> times, but it's 404ing.

Best,
Gabe

On Wed, Mar 18, 2015 at 2:25 PM, Ken Rice
> <krice at freeswitch.org> wrote:
> New Post on freeswitch.org from krice387
>
> check it out at http://ift.tt/1GpR0Yu
> ClueCon Call For Speakers is Open!
>
>
> Call for speakers at ClueCon 2015 is open!
>
> Visit http://ift.tt/1FBM7gN to
> submit your speaking proposal Below!
>
> What makes a great ClueCon
> presentation? The tech savvy crowd that attends ClueCon loves 
> technical presentations. In general, the more technical the 
> presentation, the better.
>
> If you are thinking about a presentation then consider these
> points:
> * ClueCon talks are 30 minutes in length, including Q&A time with the 
> audience (Lightning Talks are also open! 10 minute talks, 5 minutes
> Q&A)
> * ClueCon has a special focus on open source communications, VoIP, and 
> telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and 
> Kamailio
> * Attendees enjoy hearing about projects built with open source tools, 
> especially those in a production environment
> * Highly technical
> discussions that show the nuts and bolts are especially well-liked
> * The
> audience appreciates seeing and participating in live demonstrations
>
> We
> are especially interested in Security-related talks and demonstrations
>
>
> Please register your proposals at http://ift.tt/1FBM7gN. To complete
>
> registration you need the following items:
> * Working title
> * Brief
> description of the talk (abstract)
> * Name of the presenter(s)
> * Bio and
> headshot of presenter(s)
> * Presenter?s contact information (including mobile phone the 
> presenter will have with them at the conference)
>
> Don?t delay!
> Speaking proposals must be in by July 4, 2015 and scheduling requests 
> are handled on a first come first serve basis.
>
> ClueCon 2015: See You There!
>
> For more details on ClueCon see https://cluecon.com.
>
>
>
>
> ______________________________________________________________________
> ___> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
> FreeSWITCH-users 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://confluence.freeswitch.org
http://www.cl
> uecon.com

FreeSWITCH-users mailing
> list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman
> /listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt
> ions/freeswitch-users
http://www.freeswitch.org

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




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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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


From steven.szeto at mitel.com  Tue Mar 24 15:52:02 2015
From: steven.szeto at mitel.com (Szeto, Steven)
Date: Tue, 24 Mar 2015 08:52:02 -0400
Subject: [Freeswitch-users] internal versus external calls
Message-ID: <CALi6oUw+BZgLrwz09tpTR6R5WPSATt=2yCGGvAj4LU0jBTN2MQ@mail.gmail.com>

Hi Everyone,

When an incoming call comes through via SIP to Freeswitch, is there a way
to identify the call as an internal versus external call?

An internal call would be from a device hosted by a PBX.

An external call would be from the public network (e.g. a cell phone call
that has been transited through a PBX to Freeswitch.

Regards,
Steve

-- 
This e-mail (including any attachments) is for the sole use of the intended 
recipient(s) and may contain information that is confidential and/or 
protected by legal privilege. Any unauthorized review, use, copy, 
disclosure or distribution of this e-mail is strictly prohibited. If you 
are not the intended recipient, please notify Mitel immediately and destroy 
all copies of this e-mail.  Mitel does not accept any liability for breach 
of security, error or virus that may result from the transmission of this 
message.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/d4135587/attachment.html 

From mkvonarx at gmail.com  Tue Mar 24 15:57:42 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Tue, 24 Mar 2015 13:57:42 +0100
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
In-Reply-To: <CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
References: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>
	<CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
Message-ID: <CAHA5ALe=fXufoAnjCSdH1cG73f4w=na1STS+sxAwC-9ReE0FaQ@mail.gmail.com>

Hi Steven

Thanks for your reply. I actually already know that wiki page. But all
those configuration variables there don't work - at least not for SIP
channels that are connected to a mod_conference audio conference. Maybe
they do work for bridged calls, but that's not what I need. Also, the wiki
page does not mention conferences at all. And the sentence "When FreeSWITCH
does not detect speech, it stops transmitting RTP" seems not to apply to
mod_conference.
? I probably just have configured mod_conference incorrectly, but I don't
know where to check.

So any information or advice about SIP channels connected to a
mod_conference audio conference?

Thanks, Markus


2015-03-24 11:43 GMT+01:00 Steven Ayre <steveayre at gmail.com>:

> https://wiki.freeswitch.org/wiki/VAD_and_CNG
>
> On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:
>
>> Hi
>>
>> Can anyone tell me if FreeSWITCH supports silence suppression for SIP
>> calls that are inside a FreeSWITCH audio conference? If yes, how do I
>> configure mod_conference, mod_sofia and FreeSWITCH core to enable this
>> feature?
>>
>> More precisely, I try to enable/activate the behavior described in RFC
>> 3389 for G.711 in such a way that there are only RTP packets of type 13
>> every 1 or 2 seconds. I tried to play around with some possible settings
>> but could never observe anything else then the regular G.711 PCMU RTP
>> packets on the wire. Even when I set the SIP call to 'deaf' via the
>> FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU
>> RTP packets every 20ms.
>>
>> It's possible that I completly misunderstand RFC 3389 and the concepts of
>> silence suppression, comfort noise etc. In the end, what I try
>> to achieve is to reduce the network bandwidth of a G.711 SIP channel during
>> the periods when the FreeSWITCH only sends silence over the SIP channel.
>> Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to
>> another codec.
>>
>> Thanks, Markus
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/105ba0da/attachment.html 

From ssinyagin at gmail.com  Tue Mar 24 16:14:43 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 24 Mar 2015 14:14:43 +0100
Subject: [Freeswitch-users] internal versus external calls
In-Reply-To: <CALi6oUw+BZgLrwz09tpTR6R5WPSATt=2yCGGvAj4LU0jBTN2MQ@mail.gmail.com>
References: <CALi6oUw+BZgLrwz09tpTR6R5WPSATt=2yCGGvAj4LU0jBTN2MQ@mail.gmail.com>
Message-ID: <CACOYK=jyGji4jb0w6+YxjW6u5EkaMTk3mLVWWdhrG3JhWFLVug@mail.gmail.com>

they would normally come into different dialplan contexts, like
"public" and "default", and there you can distinguish them and define
the rules.


On Tue, Mar 24, 2015 at 1:52 PM, Szeto, Steven <steven.szeto at mitel.com> wrote:
> Hi Everyone,
>
> When an incoming call comes through via SIP to Freeswitch, is there a way to
> identify the call as an internal versus external call?
>
> An internal call would be from a device hosted by a PBX.
>
> An external call would be from the public network (e.g. a cell phone call
> that has been transited through a PBX to Freeswitch.
>
> Regards,
> Steve
>
> This e-mail (including any attachments) is for the sole use of the intended
> recipient(s) and may contain information that is confidential and/or
> protected by legal privilege. Any unauthorized review, use, copy, disclosure
> or distribution of this e-mail is strictly prohibited. If you are not the
> intended recipient, please notify Mitel immediately and destroy all copies
> of this e-mail.  Mitel does not accept any liability for breach of security,
> error or virus that may result from the transmission of this message.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Tue Mar 24 16:55:10 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 24 Mar 2015 08:55:10 -0500
Subject: [Freeswitch-users] ClueCon Call For Speakers is Open!
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315D3AC2D@ex2.connex.local>
Message-ID: <D136D6EE.99215%krice@freeswitch.org>

There we go... Should be all good now...


On 3/24/15, 7:34 AM, "Moishe Grunstein" <max at nysolutions.com> wrote:

> Ken this one is also broken https://www.cluecon.com/speaker-signup/
> 
> Thanks,
> 
> Moishe Grunstein
> Tornado Computer Systems, Inc.
> 212.400.7650 888.IPPBX.US
> Service Request Email: support at nysolutions.com
> 
> 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 Ken Rice
> Sent: Tuesday, March 24, 2015 7:45 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] ClueCon Call For Speakers is Open!
> 
> Still getting that?
> 
> 
> On 3/23/15, 11:51 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
> 
>> The URL:
> 
> https://www.cluecon.com/speaking-proposals
> 
> ...is referenced several
>> times, but it's 404ing.
> 
> Best,
> Gabe
> 
> On Wed, Mar 18, 2015 at 2:25 PM, Ken Rice
>> <krice at freeswitch.org> wrote:
>> New Post on freeswitch.org from krice387
>> 
>> check it out at http://ift.tt/1GpR0Yu
>> ClueCon Call For Speakers is Open!
>> 
>> 
>> Call for speakers at ClueCon 2015 is open!
>> 
>> Visit http://ift.tt/1FBM7gN to
>> submit your speaking proposal Below!
>> 
>> What makes a great ClueCon
>> presentation? The tech savvy crowd that attends ClueCon loves
>> technical presentations. In general, the more technical the
>> presentation, the better.
>> 
>> If you are thinking about a presentation then consider these
>> points:
>> * ClueCon talks are 30 minutes in length, including Q&A time with the
>> audience (Lightning Talks are also open! 10 minute talks, 5 minutes
>> Q&A)
>> * ClueCon has a special focus on open source communications, VoIP, and
>> telephony projects like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and
>> Kamailio
>> * Attendees enjoy hearing about projects built with open source tools,
>> especially those in a production environment
>> * Highly technical
>> discussions that show the nuts and bolts are especially well-liked
>> * The
>> audience appreciates seeing and participating in live demonstrations
>> 
>> We
>> are especially interested in Security-related talks and demonstrations
>> 
>> 
>> Please register your proposals at http://ift.tt/1FBM7gN. To complete
>> 
>> registration you need the following items:
>> * Working title
>> * Brief
>> description of the talk (abstract)
>> * Name of the presenter(s)
>> * Bio and
>> headshot of presenter(s)
>> * Presenter?s contact information (including mobile phone the
>> presenter will have with them at the conference)
>> 
>> Don?t delay!
>> Speaking proposals must be in by July 4, 2015 and scheduling requests
>> are handled on a first come first serve basis.
>> 
>> ClueCon 2015: See You There!
>> 
>> For more details on ClueCon see https://cluecon.com.
>> 
>> 
>> 
>> 
>> ______________________________________________________________________
>> ___> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> 
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> 
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> 
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users 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://confluence.freeswitch.org
> http://www.cl
>> uecon.com
> 
> FreeSWITCH-users mailing
>> list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman
>> /listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt
>> ions/freeswitch-users
> http://www.freeswitch.org
> 
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH





From royj at yandex.ru  Tue Mar 24 17:10:24 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Tue, 24 Mar 2015 17:10:24 +0300
Subject: [Freeswitch-users] internal versus external calls
In-Reply-To: <CACOYK=jyGji4jb0w6+YxjW6u5EkaMTk3mLVWWdhrG3JhWFLVug@mail.gmail.com>
References: <CALi6oUw+BZgLrwz09tpTR6R5WPSATt=2yCGGvAj4LU0jBTN2MQ@mail.gmail.com>
	<CACOYK=jyGji4jb0w6+YxjW6u5EkaMTk3mLVWWdhrG3JhWFLVug@mail.gmail.com>
Message-ID: <1068091427206224@web6o.yandex.ru>

Seems to mean come into different sofia profiles then dialplan contexts

24.03.2015, 16:18, "Stanislav Sinyagin" <ssinyagin at gmail.com>:
> they would normally come into different dialplan contexts, like
> "public" and "default", and there you can distinguish them and define
> the rules.
>
> On Tue, Mar 24, 2015 at 1:52 PM, Szeto, Steven <steven.szeto at mitel.com> wrote:
>> ?Hi Everyone,
>>
>> ?When an incoming call comes through via SIP to Freeswitch, is there a way to
>> ?identify the call as an internal versus external call?
>>
>> ?An internal call would be from a device hosted by a PBX.
>>
>> ?An external call would be from the public network (e.g. a cell phone call
>> ?that has been transited through a PBX to Freeswitch.
>>
>> ?Regards,
>> ?Steve
>>
>> ?This e-mail (including any attachments) is for the sole use of the intended
>> ?recipient(s) and may contain information that is confidential and/or
>> ?protected by legal privilege. Any unauthorized review, use, copy, disclosure
>> ?or distribution of this e-mail is strictly prohibited. If you are not the
>> ?intended recipient, please notify Mitel immediately and destroy all copies
>> ?of this e-mail. ?Mitel does not accept any liability for breach of security,
>> ?error or virus that may result from the transmission of this message.
>> ?_________________________________________________________________________
>> ?Professional FreeSWITCH Consulting Services:
>> ?consulting at freeswitch.org
>> ?http://www.freeswitchsolutions.com
>>
>> ?Official FreeSWITCH Sites
>> ?http://www.freeswitch.org
>> ?http://confluence.freeswitch.org
>> ?http://www.cluecon.com
>>
>> ?FreeSWITCH-users mailing list
>> ?FreeSWITCH-users at lists.freeswitch.org
>> ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ?http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tomp at tomp.uk  Tue Mar 24 17:26:26 2015
From: tomp at tomp.uk (tomp at tomp.uk)
Date: Tue, 24 Mar 2015 14:26:26 +0000
Subject: [Freeswitch-users] Freeswitch.org hacked
Message-ID: <15996d29237a43eb3293d26dce104511@tomp.uk>


Hi,

I think the freeswitch.org site has been hacked.

I am seeing adverts for viagra and other drugs top right hand side of 
the site.

Thanks
Tom


From gmaruzz at gmail.com  Tue Mar 24 17:39:33 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 24 Mar 2015 15:39:33 +0100
Subject: [Freeswitch-users] Freeswitch.org hacked
In-Reply-To: <15996d29237a43eb3293d26dce104511@tomp.uk>
References: <15996d29237a43eb3293d26dce104511@tomp.uk>
Message-ID: <CALXCt0oyG7LqRUh-FGgT=n6ktRxLCSyY6uky3JXV8fO7PDp0Rw@mail.gmail.com>

that's personalized advertising.

But the untold truth is what FreeSWITCH can do for moose enlargement

-giovanni


On Tue, Mar 24, 2015 at 3:26 PM, <tomp at tomp.uk> wrote:

>
> Hi,
>
> I think the freeswitch.org site has been hacked.
>
> I am seeing adverts for viagra and other drugs top right hand side of
> the site.
>
> Thanks
> Tom
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/5d5f00ac/attachment.html 

From stefan.kainz at 1012.at  Tue Mar 24 17:50:48 2015
From: stefan.kainz at 1012.at (Stefan Kainz)
Date: Tue, 24 Mar 2015 15:50:48 +0100
Subject: [Freeswitch-users] Freeswitch.org hacked
In-Reply-To: <CALXCt0oyG7LqRUh-FGgT=n6ktRxLCSyY6uky3JXV8fO7PDp0Rw@mail.gmail.com>
References: <15996d29237a43eb3293d26dce104511@tomp.uk>
	<CALXCt0oyG7LqRUh-FGgT=n6ktRxLCSyY6uky3JXV8fO7PDp0Rw@mail.gmail.com>
Message-ID: <551179C8.30201@1012.at>

lol


On 24.03.2015 15:39, Giovanni Maruzzelli wrote:
> that's personalized advertising.
>
> But the untold truth is what FreeSWITCH can do for moose enlargement
>
> -giovanni
>
>
> On Tue, Mar 24, 2015 at 3:26 PM, <tomp at tomp.uk <mailto:tomp at tomp.uk>> 
> wrote:
>
>
>     Hi,
>
>     I think the freeswitch.org <http://freeswitch.org> site has been
>     hacked.
>
>     I am seeing adverts for viagra and other drugs top right hand side of
>     the site.
>
>     Thanks
>     Tom
>
>     _________________________________________________________________________
>     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://confluence.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 <tel:%2B39-347-2665618>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*Stefan Kainz*
Softwareentwicklung
Systemadministration

*T*+43 / 13361012 - 2020*
F*+43 / 13361012 - 2022*
E***stefan.kainz at 1012.at <mailto:stefan.kainz at 1012.at>*
W***www. <http://www.1012.at/>1012.at

*1012**-Festnetz-Service GmbH*

FN287875x, Gerichtsstand: Handelsgericht Wien
Haydngasse 17 I 1060 Wien I Austria

Diese e-mail enth?lt vertrauliche und/oder rechtlich gesch?tzte 
Informationen.
Wenn Sie nicht der richtige Adressat sind oder diese e-mail irrt?mlich 
erhalten haben,
informieren Sie bitte sofort den Absender und vernichten Sie diese e-mail.
Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser e-mail ist 
nicht gestattet.

This e-mail may contain confidental and/or privileged information.
If you are not the intended recipient or have received this e-mail in 
error please notify the sender
immediately and destroy this e-mail. Any unauthorised copying, 
disclosure or distribution
of the material in this e-mail is strictly forbidden.

***Bitte pr?fen Sie der Umwelt zuliebe, ob der Ausdruck dieser E-Mail 
erforderlich ist.*


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/a85de255/attachment.html 

From bote_radio at botecomm.com  Tue Mar 24 17:56:13 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 24 Mar 2015 10:56:13 -0400
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
In-Reply-To: <CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
References: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>
	<CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
Message-ID: <000d01d06642$ac9efe40$05dcfac0$@botecomm.com>

I have moved that wiki page to Confluence and edited it.

 

https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG

 

Bote

 

 

From: Steven Ayre
Sent: Tuesday, 24 March, 2015 06:44
Subject: Re: [Freeswitch-users] Silence Suppression from an Audio Conference

 

https://wiki.freeswitch.org/wiki/VAD_and_CNG

 

On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:

Hi

 

Can anyone tell me if FreeSWITCH supports silence suppression for SIP calls that are inside a FreeSWITCH audio conference? If yes, how do I configure mod_conference, mod_sofia and FreeSWITCH core to enable this feature?

 

More precisely, I try to enable/activate the behavior described in RFC 3389 for G.711 in such a way that there are only RTP packets of type 13 every 1 or 2 seconds. I tried to play around with some possible settings but could never observe anything else then the regular G.711 PCMU RTP packets on the wire. Even when I set the SIP call to 'deaf' via the FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU RTP packets every 20ms.

 

It's possible that I completly misunderstand RFC 3389 and the concepts of silence suppression, comfort noise etc. In the end, what I try to achieve is to reduce the network bandwidth of a G.711 SIP channel during the periods when the FreeSWITCH only sends silence over the SIP channel. Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to another codec.

 

Thanks, Markus





 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/33000267/attachment-0001.html 

From bote_radio at botecomm.com  Tue Mar 24 18:06:35 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 24 Mar 2015 11:06:35 -0400
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
In-Reply-To: <CAHA5ALe=fXufoAnjCSdH1cG73f4w=na1STS+sxAwC-9ReE0FaQ@mail.gmail.com>
References: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>	<CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
	<CAHA5ALe=fXufoAnjCSdH1cG73f4w=na1STS+sxAwC-9ReE0FaQ@mail.gmail.com>
Message-ID: <001801d06644$1f31efc0$5d95cf40$@botecomm.com>

There is a setting listed in

 

https://freeswitch.org/confluence/display/FREESWITCH/mod_conference

 

energy-level

 

which acts as a noise gate. If you set this number high enough, the conference bridge will only admit audio from a conferee when it detects speech (or noise?) from him.

 

HOWEVER, there used to be a conference flag named ?waste? that told the conference to ?waste bandwidth? by transmitting packets all the time, even when there was no audio contained in them; now that flag has been eliminated and I understand that the conference bridge always sends packets. If I have this correct, then even the noise gate will not reduce your bandwidth.

 

I recommend you test this theory in case it is helpful and please report back with your findings. 

 

Thanks.

 

Bote

 

 

From: Markus von Arx
Sent: Tuesday, 24 March, 2015 08:58
Subject: Re: [Freeswitch-users] Silence Suppression from an Audio Conference

 

Hi Steven

 

Thanks for your reply. I actually already know that wiki page. But all those configuration variables there don't work - at least not for SIP channels that are connected to a mod_conference audio conference. Maybe they do work for bridged calls, but that's not what I need. Also, the wiki page does not mention conferences at all. And the sentence "When FreeSWITCH does not detect speech, it stops transmitting RTP" seems not to apply to mod_conference.

? I probably just have configured mod_conference incorrectly, but I don't know where to check.

 

So any information or advice about SIP channels connected to a mod_conference audio conference?

 

Thanks, Markus

 

 

2015-03-24 11:43 GMT+01:00 Steven Ayre <steveayre at gmail.com>:

https://wiki.freeswitch.org/wiki/VAD_and_CNG

 

On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:

Hi

 

Can anyone tell me if FreeSWITCH supports silence suppression for SIP calls that are inside a FreeSWITCH audio conference? If yes, how do I configure mod_conference, mod_sofia and FreeSWITCH core to enable this feature?

 

More precisely, I try to enable/activate the behavior described in RFC 3389 for G.711 in such a way that there are only RTP packets of type 13 every 1 or 2 seconds. I tried to play around with some possible settings but could never observe anything else then the regular G.711 PCMU RTP packets on the wire. Even when I set the SIP call to 'deaf' via the FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU RTP packets every 20ms.

 

It's possible that I completly misunderstand RFC 3389 and the concepts of silence suppression, comfort noise etc. In the end, what I try to achieve is to reduce the network bandwidth of a G.711 SIP channel during the periods when the FreeSWITCH only sends silence over the SIP channel. Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to another codec.

 

Thanks, Markus

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/7cf939a2/attachment.html 

From bote_radio at botecomm.com  Tue Mar 24 18:11:59 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Tue, 24 Mar 2015 11:11:59 -0400
Subject: [Freeswitch-users] internal versus external calls
In-Reply-To: <CACOYK=jyGji4jb0w6+YxjW6u5EkaMTk3mLVWWdhrG3JhWFLVug@mail.gmail.com>
References: <CALi6oUw+BZgLrwz09tpTR6R5WPSATt=2yCGGvAj4LU0jBTN2MQ@mail.gmail.com>
	<CACOYK=jyGji4jb0w6+YxjW6u5EkaMTk3mLVWWdhrG3JhWFLVug@mail.gmail.com>
Message-ID: <001d01d06644$e04f5850$a0ee08f0$@botecomm.com>

Yes, if by "internal" it means internal to your own FreeSWITCH installation
which you control. Then you can set a variable early in the dialplan that
indicates that this is an internal call. In fact, there might be an example
of this in the vanilla configuration files already.

If you need to test for this variable in the same pass through the dialplan
and not in later processing you should use the "inline" attribute so that it
takes effect immediately during the evaluation phase and not later during
the dialplan execution phase.

Otherwise, you could probably devise some clever means to detect the source
of calls by Caller*ID number or inbound carrier or profile as Stanislav
points out; there is a lot of flexibility available.

Bote


> -----Original Message-----
> From: Stanislav Sinyagin
> Sent: Tuesday, 24 March, 2015 09:15
> Subject: Re: [Freeswitch-users] internal versus external calls
> 
> they would normally come into different dialplan contexts, like
> "public" and "default", and there you can distinguish them and define
> the rules.
> 
> 
> On Tue, Mar 24, 2015 at 1:52 PM, Szeto, Steven <steven.szeto at mitel.com>
> wrote:
> > Hi Everyone,
> >
> > When an incoming call comes through via SIP to Freeswitch, is there a
way
> to
> > identify the call as an internal versus external call?
> >
> > An internal call would be from a device hosted by a PBX.
> >
> > An external call would be from the public network (e.g. a cell phone
call
> > that has been transited through a PBX to Freeswitch.
> >
> > Regards,
> > Steve
> >
> > This e-mail (including any attachments) is for the sole use of the
intended
> > recipient(s) and may contain information that is confidential and/or
> > protected by legal privilege. Any unauthorized review, use, copy,
disclosure
> > or distribution of this e-mail is strictly prohibited. If you are not
the
> > intended recipient, please notify Mitel immediately and destroy all
copies
> > of this e-mail.  Mitel does not accept any liability for breach of
security,
> > error or virus that may result from the transmission of this message.
> >



From jmesquita at freeswitch.org  Tue Mar 24 18:19:16 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 12:19:16 -0300
Subject: [Freeswitch-users] Polycom headache
Message-ID: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>

Guys, I am sure most of you have been through this but this is my first
time. I am having a HOLD problem with Polycom where they refuse to work
according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
is a known thing since I've found several hits on Google about it and they
even created an option for it on their provisioning material.
Unfortunately, it doesn't work. No matter what I set on the

<pre>
if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
!strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
"RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
back.....\n");
sendonly = 2; /* global sendonly always wins */
}
Jo?o Mesquita
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/25ba8a58/attachment.html 

From jmesquita at freeswitch.org  Tue Mar 24 18:25:11 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 12:25:11 -0300
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
Message-ID: <CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>

I am deeply sorry, I've hit send too early. Damn Gmail...

Guys, I am sure most of you have been through this but this is my first
time. I am having a HOLD problem with Polycom where they refuse to work
according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
is a known thing since I've found several hits on Google about it and they
even created an option for it on their provisioning material.
Unfortunately, it doesn't work. No matter what I set on the
voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:

v=0
o=- 1167669703 1167669704 IN IP4 0.0.0.0
s=Polycom IP Phone
c=IN IP4 0.0.0.0
t=0 0
m=audio 2228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

I've also looked into FS code and I've found something on the lines of, but
this is not kicking in either (granted, I am using 1.4.14 still and I
really don't believe FS should cope with this crap either).

if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->
> sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
> "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
> back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }


So, why am I coming to this ML for help? I wanted to know how did you guys
figure this out, is there an exact firmware version where Polycom has this
figured out or is it one of those long lasting ones that Polycom never
cares to address?

And btw, I tried with UC software version 4.0.8 and 4.0.7.

Thank you,
Jo?o Mesquita


On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
wrote:

>
> Guys, I am sure most of you have been through this but this is my first
> time. I am having a HOLD problem with Polycom where they refuse to work
> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
> is a known thing since I've found several hits on Google about it and they
> even created an option for it on their provisioning material.
> Unfortunately, it doesn't work. No matter what I set on the
>
> <pre>
> if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
> !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
> "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
> back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }
> Jo?o Mesquita
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/6388357d/attachment-0001.html 

From max at nysolutions.com  Tue Mar 24 18:29:15 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 24 Mar 2015 15:29:15 +0000
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D3BC02@ex2.connex.local>

You can set the polycom config to use the other rfc, we did that here.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: Tuesday, March 24, 2015 11:19 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Polycom headache


Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the

<pre>
if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
sendonly = 2; /* global sendonly always wins */
}
Jo?o Mesquita
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/25c3bfb8/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/25c3bfb8/attachment.jpg 

From mike at jerris.com  Tue Mar 24 18:43:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 24 Mar 2015 11:43:28 -0400
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
Message-ID: <1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>

Brian found this in the admin guide:

Enable Call Hold Parameter Function
Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) for outgoing hold signaling.
Specify whether to use sendonly hold signaling. Configure local call hold reminder options. Specify the music-on-hold URI.
Example Call Hold Configuration
template > parameter
sip-interop.cfg > voIpProt.SIP.useRFC2543hold
sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg > call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri

> On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> 
> I am deeply sorry, I've hit send too early. Damn Gmail...
> 
> Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
> 
> v=0
> o=- 1167669703 1167669704 IN IP4 0.0.0.0
> s=Polycom IP Phone
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 2228 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> 
> I've also looked into FS code and I've found something on the lines of, but this is not kicking in either (granted, I am using 1.4.14 still and I really don't believe FS should cope with this crap either).
> 
> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }
> 
> So, why am I coming to this ML for help? I wanted to know how did you guys figure this out, is there an exact firmware version where Polycom has this figured out or is it one of those long lasting ones that Polycom never cares to address?
> 
> And btw, I tried with UC software version 4.0.8 and 4.0.7.
> 
> Thank you,
> Jo?o Mesquita
> 
> 
> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org <mailto:jmesquita at freeswitch.org>> wrote:
> 
> Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the 
> 
> <pre>
> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }
> Jo?o Mesquita
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/19f257c1/attachment-0001.html 

From jmesquita at freeswitch.org  Tue Mar 24 19:27:14 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 13:27:14 -0300
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
Message-ID: <CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>

Moishe, what UC software version are you using?

MikeJ, I've set this on the web interface as well as provisioning and it's
like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware
version that will make this work would be awesome.

Jo?o Mesquita
FreeSWITCH? Solutions

On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Brian found this in the admin guide:
>
> Enable Call Hold Parameter Function
> Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
> a=inactive) for outgoing hold signaling.
> Specify whether to use sendonly hold signaling. Configure local call hold
> reminder options. Specify the music-on-hold URI.
> Example Call Hold Configuration
> template > parameter
> sip-interop.cfg > voIpProt.SIP.useRFC2543hold
> sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg >
> call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri
>
> On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
> I am deeply sorry, I've hit send too early. Damn Gmail...
>
> Guys, I am sure most of you have been through this but this is my first
> time. I am having a HOLD problem with Polycom where they refuse to work
> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
> is a known thing since I've found several hits on Google about it and they
> even created an option for it on their provisioning material.
> Unfortunately, it doesn't work. No matter what I set on the
> voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
>
> v=0
> o=- 1167669703 1167669704 IN IP4 0.0.0.0
> s=Polycom IP Phone
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 2228 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
>
> I've also looked into FS code and I've found something on the lines of,
> but this is not kicking in either (granted, I am using 1.4.14 still and I
> really don't believe FS should cope with this crap either).
>
> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp-
>> >sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING
>> , "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
>> back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>
>
> So, why am I coming to this ML for help? I wanted to know how did you guys
> figure this out, is there an exact firmware version where Polycom has this
> figured out or is it one of those long lasting ones that Polycom never
> cares to address?
>
> And btw, I tried with UC software version 4.0.8 and 4.0.7.
>
> Thank you,
> Jo?o Mesquita
>
>
> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
>>
>> Guys, I am sure most of you have been through this but this is my first
>> time. I am having a HOLD problem with Polycom where they refuse to work
>> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
>> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
>> is a known thing since I've found several hits on Google about it and they
>> even created an option for it on their provisioning material.
>> Unfortunately, it doesn't work. No matter what I set on the
>>
>> <pre>
>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
>> !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>> SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their
>> 0.0.0.0 hold method back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>> Jo?o Mesquita
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/60ecdcf0/attachment.html 

From spencer at whiteskycommunications.com  Tue Mar 24 19:48:40 2015
From: spencer at whiteskycommunications.com (Spencer Thomason)
Date: Tue, 24 Mar 2015 16:48:40 +0000
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
Message-ID: <34E78280-7368-4E54-BB8D-3278A554ACF7@whiteskycommunications.com>

Hey Guys,
This is working for us:
voIpProt.SIP.useRFC2543hold=?0"
voIpProt.SIP.useRFC3264HoldOnly=?1"
voIpProt.SIP.useSendonlyHold="1"

As of 4.0.3 Rev F, you need voIpProt.SIP.useRFC3264HoldOnly=?1? to force it for some reason.

See:
http://downloads.polycom.com/voice/voip/uc/UC_Software_Release_Notes_4_0_3_Rev_F.pdf
for details.

Thanks,
Spencer




On Mar 24, 2015, at 9:27 AM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

Moishe, what UC software version are you using?

MikeJ, I've set this on the web interface as well as provisioning and it's like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware version that will make this work would be awesome.

Jo?o Mesquita
FreeSWITCH? Solutions

On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Brian found this in the admin guide:

Enable Call Hold Parameter Function
Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) for outgoing hold signaling.
Specify whether to use sendonly hold signaling. Configure local call hold reminder options. Specify the music-on-hold URI.
Example Call Hold Configuration
template > parameter
sip-interop.cfg > voIpProt.SIP.useRFC2543hold
sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg > call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri

On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

I am deeply sorry, I've hit send too early. Damn Gmail...

Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:

v=0
o=- 1167669703 1167669704 IN IP4 0.0.0.0
s=Polycom IP Phone
c=IN IP4 0.0.0.0
t=0 0
m=audio 2228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

I've also looked into FS code and I've found something on the lines of, but this is not kicking in either (granted, I am using 1.4.14 still and I really don't believe FS should cope with this crap either).

if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
sendonly = 2; /* global sendonly always wins */
}

So, why am I coming to this ML for help? I wanted to know how did you guys figure this out, is there an exact firmware version where Polycom has this figured out or is it one of those long lasting ones that Polycom never cares to address?

And btw, I tried with UC software version 4.0.8 and 4.0.7.

Thank you,
Jo?o Mesquita


On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the

<pre>
if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
sendonly = 2; /* global sendonly always wins */
}
Jo?o Mesquita

_________________________________________________________________________
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://confluence.freeswitch.org<http://confluence.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://confluence.freeswitch.org<http://confluence.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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/8e3f8692/attachment-0001.html 

From mike at jerris.com  Tue Mar 24 19:52:02 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 24 Mar 2015 12:52:02 -0400
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
Message-ID: <B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>

which phone models?  what firmware are they on?


> On Mar 24, 2015, at 12:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
> 
> Moishe, what UC software version are you using?
> 
> MikeJ, I've set this on the web interface as well as provisioning and it's like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware version that will make this work would be awesome.
> 
> Jo?o Mesquita
> FreeSWITCH? Solutions
> 
> On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Brian found this in the admin guide:
> 
> Enable Call Hold Parameter Function
> Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) for outgoing hold signaling.
> Specify whether to use sendonly hold signaling. Configure local call hold reminder options. Specify the music-on-hold URI.
> Example Call Hold Configuration
> template > parameter
> sip-interop.cfg > voIpProt.SIP.useRFC2543hold
> sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg > call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri
> 
>> On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org <mailto:jmesquita at freeswitch.org>> wrote:
>> 
>> I am deeply sorry, I've hit send too early. Damn Gmail...
>> 
>> Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
>> 
>> v=0
>> o=- 1167669703 1167669704 IN IP4 0.0.0.0
>> s=Polycom IP Phone
>> c=IN IP4 0.0.0.0
>> t=0 0
>> m=audio 2228 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> 
>> I've also looked into FS code and I've found something on the lines of, but this is not kicking in either (granted, I am using 1.4.14 still and I really don't believe FS should cope with this crap either).
>> 
>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>> 
>> So, why am I coming to this ML for help? I wanted to know how did you guys figure this out, is there an exact firmware version where Polycom has this figured out or is it one of those long lasting ones that Polycom never cares to address?
>> 
>> And btw, I tried with UC software version 4.0.8 and 4.0.7.
>> 
>> Thank you,
>> Jo?o Mesquita
>> 
>> 
>> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org <mailto:jmesquita at freeswitch.org>> wrote:
>> 
>> Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the 
>> 
>> <pre>
>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>> Jo?o Mesquita

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/f9c327fd/attachment.html 

From jmesquita at freeswitch.org  Tue Mar 24 20:15:06 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 14:15:06 -0300
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
	<B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>
Message-ID: <CABH8QX1cSZBPkGBTJUQz2-4DpPY4oaHUAnDvNXwxKrVMvYeY0w@mail.gmail.com>

They are all IP331 and IP550 on firmware 4.0.8 (latest). I am giving
Spencer's suggestion for a spin and will report back ASAP.

Thanks Spencer!

Jo?o Mesquita
FreeSWITCH? Solutions

On Tue, Mar 24, 2015 at 1:52 PM, Michael Jerris <mike at jerris.com> wrote:

> which phone models?  what firmware are they on?
>
>
> On Mar 24, 2015, at 12:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
> Moishe, what UC software version are you using?
>
> MikeJ, I've set this on the web interface as well as provisioning and it's
> like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware
> version that will make this work would be awesome.
>
> Jo?o Mesquita
> FreeSWITCH? Solutions
>
> On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Brian found this in the admin guide:
>>
>> Enable Call Hold Parameter Function
>> Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
>> a=inactive) for outgoing hold signaling.
>> Specify whether to use sendonly hold signaling. Configure local call hold
>> reminder options. Specify the music-on-hold URI.
>> Example Call Hold Configuration
>> template > parameter
>> sip-interop.cfg > voIpProt.SIP.useRFC2543hold
>> sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg >
>> call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri
>>
>> On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org>
>> wrote:
>>
>> I am deeply sorry, I've hit send too early. Damn Gmail...
>>
>> Guys, I am sure most of you have been through this but this is my first
>> time. I am having a HOLD problem with Polycom where they refuse to work
>> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
>> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
>> is a known thing since I've found several hits on Google about it and they
>> even created an option for it on their provisioning material.
>> Unfortunately, it doesn't work. No matter what I set on the
>> voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
>>
>> v=0
>> o=- 1167669703 1167669704 IN IP4 0.0.0.0
>> s=Polycom IP Phone
>> c=IN IP4 0.0.0.0
>> t=0 0
>> m=audio 2228 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>>
>> I've also looked into FS code and I've found something on the lines of,
>> but this is not kicking in either (granted, I am using 1.4.14 still and I
>> really don't believe FS should cope with this crap either).
>>
>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp
>>> ->sdp_connection->c_address, "0.0.0.0"))) {
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>> SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their
>>> 0.0.0.0 hold method back.....\n");
>>> sendonly = 2; /* global sendonly always wins */
>>> }
>>
>>
>> So, why am I coming to this ML for help? I wanted to know how did you
>> guys figure this out, is there an exact firmware version where Polycom has
>> this figured out or is it one of those long lasting ones that Polycom never
>> cares to address?
>>
>> And btw, I tried with UC software version 4.0.8 and 4.0.7.
>>
>> Thank you,
>> Jo?o Mesquita
>>
>>
>> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org
>> > wrote:
>>
>>>
>>> Guys, I am sure most of you have been through this but this is my first
>>> time. I am having a HOLD problem with Polycom where they refuse to work
>>> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
>>> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
>>> is a known thing since I've found several hits on Google about it and they
>>> even created an option for it on their provisioning material.
>>> Unfortunately, it doesn't work. No matter what I set on the
>>>
>>> <pre>
>>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
>>> !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>>> SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their
>>> 0.0.0.0 hold method back.....\n");
>>> sendonly = 2; /* global sendonly always wins */
>>> }
>>> Jo?o Mesquita
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/fa5ab9ac/attachment-0001.html 

From krice at freeswitch.org  Tue Mar 24 20:22:14 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 24 Mar 2015 12:22:14 -0500
Subject: [Freeswitch-users] Freeswitch.org hacked
In-Reply-To: <551179C8.30201@1012.at>
Message-ID: <D1370776.99252%krice@freeswitch.org>

Its been fixed...


On 3/24/15, 9:50 AM, "Stefan Kainz" <stefan.kainz at 1012.at> wrote:

>    lol
>  
>  
>  
> On 24.03.2015 15:39, Giovanni Maruzzelli wrote:
>  
>  
>>  
>>  
>> that's personalized advertising.
>>  
>>  
>>  But the untold truth is what FreeSWITCH can do for moose enlargement
>>  
>>  
>> 
>>  
>>  
>> -giovanni
>>  
>>  
>> 
>>  
>> 
>>  
>> On Tue, Mar 24, 2015 at 3:26 PM, <tomp at tomp.uk> wrote:
>>  
>>> 
>>>  Hi,
>>>  
>>>  I think the freeswitch.org <http://freeswitch.org>  site has been hacked.
>>>  
>>>  I am seeing adverts for viagra and other drugs top right hand side of
>>>  the site.
>>>  
>>>  Thanks
>>>  Tom
>>>  
>>> _________________________________________________________________________
>>>  Professional FreeSWITCH Consulting Services:
>>>  consulting at freeswitch.org
>>>  http://www.freeswitchsolutions.com
>>>  
>>>  Official FreeSWITCH Sites
>>>  http://www.freeswitch.org
>>>  http://confluence.freeswitch.org
>>>  http://www.cluecon.com
>>>  
>>>  FreeSWITCH-users mailing list
>>>  FreeSWITCH-users at 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 <tel:%2B39-347-2665618>
>>  
>>  
>>  
>>  
>>  
>>  
>>   
>>  
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
Twitter: @FreeSWITCH


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/041314ff/attachment.html 

From max at nysolutions.com  Tue Mar 24 20:49:44 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 24 Mar 2015 17:49:44 +0000
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX1cSZBPkGBTJUQz2-4DpPY4oaHUAnDvNXwxKrVMvYeY0w@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
	<B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>
	<CABH8QX1cSZBPkGBTJUQz2-4DpPY4oaHUAnDvNXwxKrVMvYeY0w@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D3C87C@ex2.connex.local>

We are on various 4 and 5 revisions and it works.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita
Sent: Tuesday, March 24, 2015 1:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Polycom headache

They are all IP331 and IP550 on firmware 4.0.8 (latest). I am giving Spencer's suggestion for a spin and will report back ASAP.

Thanks Spencer!

Jo?o Mesquita
FreeSWITCH? Solutions

On Tue, Mar 24, 2015 at 1:52 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
which phone models?  what firmware are they on?


On Mar 24, 2015, at 12:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

Moishe, what UC software version are you using?

MikeJ, I've set this on the web interface as well as provisioning and it's like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware version that will make this work would be awesome.

Jo?o Mesquita
FreeSWITCH? Solutions

On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Brian found this in the admin guide:

Enable Call Hold Parameter Function
Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) for outgoing hold signaling.
Specify whether to use sendonly hold signaling. Configure local call hold reminder options. Specify the music-on-hold URI.
Example Call Hold Configuration
template > parameter
sip-interop.cfg > voIpProt.SIP.useRFC2543hold
sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg > call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri

On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

I am deeply sorry, I've hit send too early. Damn Gmail...

Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:

v=0
o=- 1167669703 1167669704 IN IP4 0.0.0.0
s=Polycom IP Phone
c=IN IP4 0.0.0.0
t=0 0
m=audio 2228 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

I've also looked into FS code and I've found something on the lines of, but this is not kicking in either (granted, I am using 1.4.14 still and I really don't believe FS should cope with this crap either).

if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
sendonly = 2; /* global sendonly always wins */
}

So, why am I coming to this ML for help? I wanted to know how did you guys figure this out, is there an exact firmware version where Polycom has this figured out or is it one of those long lasting ones that Polycom never cares to address?

And btw, I tried with UC software version 4.0.8 and 4.0.7.

Thank you,
Jo?o Mesquita

On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org<mailto:jmesquita at freeswitch.org>> wrote:

Guys, I am sure most of you have been through this but this is my first time. I am having a HOLD problem with Polycom where they refuse to work according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly) to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this is a known thing since I've found several hits on Google about it and they even created an option for it on their provisioning material. Unfortunately, it doesn't work. No matter what I set on the

<pre>
if ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back.....\n");
sendonly = 2; /* global sendonly always wins */
}
Jo?o Mesquita


_________________________________________________________________________
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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/065afee8/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/065afee8/attachment-0001.jpg 

From italorossib at gmail.com  Tue Mar 24 22:39:29 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 24 Mar 2015 16:39:29 -0300
Subject: [Freeswitch-users] Transfering Attended Call
In-Reply-To: <CAKy26iPjxKBZ0Thj3v4eXuVCGDqpA0uRWcAjbJODXpSsO_mzhQ@mail.gmail.com>
References: <CAKy26iPjxKBZ0Thj3v4eXuVCGDqpA0uRWcAjbJODXpSsO_mzhQ@mail.gmail.com>
Message-ID: <CAJR7f2H+3xWseV0V6uwzJtjc0_fw+obMr3NHyTk81_hWSkmPFQ@mail.gmail.com>

Ol? Luiz!

Take a look at
https://freeswitch.org/confluence/display/FREESWITCH/Attended+Transfer

On Tue, Mar 24, 2015 at 4:36 AM, Luiz Fernando Softov <
fernando at softov.com.br> wrote:

> Hi... First of all i make a software with freeswitch and FreeBSD, named
> Voipr...
>
> voipr.brbyte.com
>
> It's a softswitch with Billing, Redirect, IVR, Routes, GSM, Gateway and
> other things.
>
> I am not using dialplan, just a outbound socket, with my own ESL Server
> written in C.
> I parse the events, and send comands back to make things. Like bridge,
> play some music, hangup the call....
>
> <extension name="VoiprExt">
>     <condition>
>         <action application="socket" data="127.0.0.1:3012 async full"/>
>         <action application="hangup"/>
>     </condition>
> </extension>
>
> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
>
> My question, is how to make a transfer of a call...
>
> Like A (1001) call to B (1002).
>
> Then i make a BRIDGE between extensions
>
> "execute", "set", "dialed_extension=1002"
> "execute", "export", "dialed_extension=1002"
> "execute", "set", "call_timeout=30"
> "execute", "set", "continue_on_fail=true"
> "execute", "set", "park_after_bridge=true"
> "execute", "bridge", "user/${dialed_extension}"
>
> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
>
> Let's say B (1002) is a secretary who want to transfer the call to C (1003)
>
> B (1002), press the key [*], and i play a music, like "enter the desired
> number and #".
>
> B (1002), press the keys 1003#
>
> - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
>
> At this point, I have all information, and want to know how 3 things
>
> Consider that: the call already bridged between A (1001) and B (1002)
>
> - - - - -
> 1 - Making a call between B and C.
> Then, when C answer B hangup. And the call return between A and C.
>
>
> - - - - -
> 2 - Making a call between B and C.
> Then, when C answer, B talks with C. B press * and B hangup.
> And the call return between A and C
>
>
> - - - - -
> 3 - Making a call between A and C.
> Then, when C answer, A hangup.
> And the call return to B and C
>
>
> Thank you in advance, who can help me.
>
> Sorry for my english.
> If anyone need help with socket events, gsmopen or xml_curl send me a
> e-mail. I will be happy to help.
>
> Att,
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/48fa2bc1/attachment.html 

From jmesquita at freeswitch.org  Wed Mar 25 02:47:56 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 20:47:56 -0300
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315D3C87C@ex2.connex.local>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
	<B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>
	<CABH8QX1cSZBPkGBTJUQz2-4DpPY4oaHUAnDvNXwxKrVMvYeY0w@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315D3C87C@ex2.connex.local>
Message-ID: <CABH8QX3Np+vK=ET-aHWNn9mv5bDX_GH8z2aYiLopHt0kbdcwSQ@mail.gmail.com>

For future reference, Spencer was right. Without that obscure, never
mentioned option, it all works great!

Thank you Spencer!

Regards,
Jo?o Mesquita

On Tue, Mar 24, 2015 at 2:49 PM, Moishe Grunstein <max at nysolutions.com>
wrote:

>  We are on various 4 and 5 revisions and it works.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o
> Mesquita
> *Sent:* Tuesday, March 24, 2015 1:15 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Polycom headache
>
>
>
> They are all IP331 and IP550 on firmware 4.0.8 (latest). I am giving
> Spencer's suggestion for a spin and will report back ASAP.
>
>
>
> Thanks Spencer!
>
>
>   Jo?o Mesquita
> FreeSWITCH? Solutions
>
>
>
> On Tue, Mar 24, 2015 at 1:52 PM, Michael Jerris <mike at jerris.com> wrote:
>
>  which phone models?  what firmware are they on?
>
>
>
>
>
>  On Mar 24, 2015, at 12:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
>
>
> Moishe, what UC software version are you using?
>
>
>
> MikeJ, I've set this on the web interface as well as provisioning and it's
> like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware
> version that will make this work would be awesome.
>
>
>   Jo?o Mesquita
> FreeSWITCH? Solutions
>
>
>
> On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com> wrote:
>
>  Brian found this in the admin guide:
>
>
>
> Enable Call Hold Parameter Function
>
> Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
> a=inactive) for outgoing hold signaling.
>
> Specify whether to use sendonly hold signaling. Configure local call hold
> reminder options. Specify the music-on-hold URI.
>
> Example Call Hold Configuration
>
> template > parameter
>
> sip-interop.cfg > voIpProt.SIP.useRFC2543hold
>
> sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg >
> call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri
>
>
>
>   On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
>
>
> I am deeply sorry, I've hit send too early. Damn Gmail...
>
>
>
> Guys, I am sure most of you have been through this but this is my first
> time. I am having a HOLD problem with Polycom where they refuse to work
> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
> is a known thing since I've found several hits on Google about it and they
> even created an option for it on their provisioning material.
> Unfortunately, it doesn't work. No matter what I set on the
> voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
>
>
>
> v=0
>
> o=- 1167669703 1167669704 IN IP4 0.0.0.0
>
> s=Polycom IP Phone
>
> c=IN IP4 0.0.0.0
>
> t=0 0
>
> m=audio 2228 RTP/AVP 8 101
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
>
>
> I've also looked into FS code and I've found something on the lines of,
> but this is not kicking in either (granted, I am using 1.4.14 still and I
> really don't believe FS should cope with this crap either).
>
>
>
> *if* ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(
> sdp->sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
> "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
> back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }
>
>
>
> So, why am I coming to this ML for help? I wanted to know how did you guys
> figure this out, is there an exact firmware version where Polycom has this
> figured out or is it one of those long lasting ones that Polycom never
> cares to address?
>
>
>
> And btw, I tried with UC software version 4.0.8 and 4.0.7.
>
>
>
> Thank you,
>
> Jo?o Mesquita
>
>
>
> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
> wrote:
>
>
>
> Guys, I am sure most of you have been through this but this is my first
> time. I am having a HOLD problem with Polycom where they refuse to work
> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
> is a known thing since I've found several hits on Google about it and they
> even created an option for it on their provisioning material.
> Unfortunately, it doesn't work. No matter what I set on the
>
>
>
> <pre>
> if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
> !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING,
> "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
> back.....\n");
> sendonly = 2; /* global sendonly always wins */
> }
>
> Jo?o Mesquita
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/d49ceec3/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/d49ceec3/attachment-0001.jpg 

From jmesquita at freeswitch.org  Wed Mar 25 02:48:48 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Tue, 24 Mar 2015 20:48:48 -0300
Subject: [Freeswitch-users] Polycom headache
In-Reply-To: <CABH8QX3Np+vK=ET-aHWNn9mv5bDX_GH8z2aYiLopHt0kbdcwSQ@mail.gmail.com>
References: <CABH8QX0NzMUnwwbGCF+T2UxKqfFByJgCMu1oo0XDpOfaPSO0iQ@mail.gmail.com>
	<CABH8QX0Mjyo3c_OJQHtxQNbwBFMenF4bOs7GvSrbfc_5Lx2vQg@mail.gmail.com>
	<1B64E3ED-0A3A-4802-934B-9C047B684A6A@jerris.com>
	<CABH8QX24TJs-hT0L_cSpMAX0ZrXH0hkQuyopnP3qMHHYwtKHGQ@mail.gmail.com>
	<B42B3731-D72B-4449-8740-C1975BA12A5C@jerris.com>
	<CABH8QX1cSZBPkGBTJUQz2-4DpPY4oaHUAnDvNXwxKrVMvYeY0w@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315D3C87C@ex2.connex.local>
	<CABH8QX3Np+vK=ET-aHWNn9mv5bDX_GH8z2aYiLopHt0kbdcwSQ@mail.gmail.com>
Message-ID: <CABH8QX1rJRdqC+z_uSM0dx_bvTYuDKkgrTUnMpwhKhywki9Bxw@mail.gmail.com>

Sorry, with that obscure, never mentioned option, not the other way around.
I gotta start re-reading my emails *before* I hit send..

Jo?o Mesquita

On Tue, Mar 24, 2015 at 8:47 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
wrote:

> For future reference, Spencer was right. Without that obscure, never
> mentioned option, it all works great!
>
> Thank you Spencer!
>
> Regards,
> Jo?o Mesquita
>
> On Tue, Mar 24, 2015 at 2:49 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>>  We are on various 4 and 5 revisions and it works.
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Moishe Grunstein
>>
>> Tornado Computer Systems, Inc.
>>
>> 212.400.7650 888.IPPBX.US
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>>
>> Computer Networking * Managed Services * IP Video Surveillance * Network
>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>> Security * Site Surveys * CMS
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o
>> Mesquita
>> *Sent:* Tuesday, March 24, 2015 1:15 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Polycom headache
>>
>>
>>
>> They are all IP331 and IP550 on firmware 4.0.8 (latest). I am giving
>> Spencer's suggestion for a spin and will report back ASAP.
>>
>>
>>
>> Thanks Spencer!
>>
>>
>>   Jo?o Mesquita
>> FreeSWITCH? Solutions
>>
>>
>>
>> On Tue, Mar 24, 2015 at 1:52 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>  which phone models?  what firmware are they on?
>>
>>
>>
>>
>>
>>  On Mar 24, 2015, at 12:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
>> wrote:
>>
>>
>>
>> Moishe, what UC software version are you using?
>>
>>
>>
>> MikeJ, I've set this on the web interface as well as provisioning and
>> it's like it doesn't care, it will send 0.0.0.0 anyway. Knowing a firmware
>> version that will make this work would be awesome.
>>
>>
>>   Jo?o Mesquita
>> FreeSWITCH? Solutions
>>
>>
>>
>> On Tue, Mar 24, 2015 at 12:43 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>  Brian found this in the admin guide:
>>
>>
>>
>> Enable Call Hold Parameter Function
>>
>> Specify whether to use RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or
>> a=inactive) for outgoing hold signaling.
>>
>> Specify whether to use sendonly hold signaling. Configure local call hold
>> reminder options. Specify the music-on-hold URI.
>>
>> Example Call Hold Configuration
>>
>> template > parameter
>>
>> sip-interop.cfg > voIpProt.SIP.useRFC2543hold
>>
>> sip-interop.cfg > voIpProt.SIP.useSendonlyHold sip-interop.cfg >
>> call.hold.localReminder.* sip-interop.cfg > voIpProt.SIP.musicOnHold.uri
>>
>>
>>
>>   On Mar 24, 2015, at 11:25 AM, Jo?o Mesquita <jmesquita at freeswitch.org>
>> wrote:
>>
>>
>>
>> I am deeply sorry, I've hit send too early. Damn Gmail...
>>
>>
>>
>> Guys, I am sure most of you have been through this but this is my first
>> time. I am having a HOLD problem with Polycom where they refuse to work
>> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
>> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
>> is a known thing since I've found several hits on Google about it and they
>> even created an option for it on their provisioning material.
>> Unfortunately, it doesn't work. No matter what I set on the
>> voIpProt.SIP.useRFC2543hold option, it still sends 0.0.0.0 as follows:
>>
>>
>>
>> v=0
>>
>> o=- 1167669703 1167669704 IN IP4 0.0.0.0
>>
>> s=Polycom IP Phone
>>
>> c=IN IP4 0.0.0.0
>>
>> t=0 0
>>
>> m=audio 2228 RTP/AVP 8 101
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>>
>>
>> I've also looked into FS code and I've found something on the lines of,
>> but this is not kicking in either (granted, I am using 1.4.14 still and I
>> really don't believe FS should cope with this crap either).
>>
>>
>>
>> *if* ((sdp->sdp_connection && sdp->sdp_connection->c_address && !strcmp(
>> sdp->sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_WARNING
>> , "RFC2543 from March 1999 called; They want their 0.0.0.0 hold method
>> back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>>
>>
>>
>> So, why am I coming to this ML for help? I wanted to know how did you
>> guys figure this out, is there an exact firmware version where Polycom has
>> this figured out or is it one of those long lasting ones that Polycom never
>> cares to address?
>>
>>
>>
>> And btw, I tried with UC software version 4.0.8 and 4.0.7.
>>
>>
>>
>> Thank you,
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Tue, Mar 24, 2015 at 12:19 PM, Jo?o Mesquita <jmesquita at freeswitch.org>
>> wrote:
>>
>>
>>
>> Guys, I am sure most of you have been through this but this is my first
>> time. I am having a HOLD problem with Polycom where they refuse to work
>> according to the "new" RFC3624 using SDP a attributes (sendonly/recvonly)
>> to indicate hold and keep sending 0.0.0.0 as the old RFC2543. I guess this
>> is a known thing since I've found several hits on Google about it and they
>> even created an option for it on their provisioning material.
>> Unfortunately, it doesn't work. No matter what I set on the
>>
>>
>>
>> <pre>
>> if ((sdp->sdp_connection && sdp->sdp_connection->c_address &&
>> !strcmp(sdp->sdp_connection->c_address, "0.0.0.0"))) {
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
>> SWITCH_LOG_WARNING, "RFC2543 from March 1999 called; They want their
>> 0.0.0.0 hold method back.....\n");
>> sendonly = 2; /* global sendonly always wins */
>> }
>>
>> Jo?o Mesquita
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/c02348f9/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150324/c02348f9/attachment-0001.jpg 

From gmaruzz at gmail.com  Wed Mar 25 10:23:03 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 25 Mar 2015 08:23:03 +0100
Subject: [Freeswitch-users] T30 fax analysis from G711 pcap
Message-ID: <CALXCt0oNpKYcZBzr1qFqEHUg5z7pH2unNScncGewgjfrt_ZodQ@mail.gmail.com>

Hello FreeSWITCHers,

there is a way to extract T30 log from a g711 pcap?

I must check on various possible incompatibilities and anomalies on a large
number of different fax transmission, many of them in pure T30
(faxmachine<->faxmachine).

I would need something like
http://www.netgencommunications.com/products/faxtapng/...

Would it be possible to have a "replay" of the capture to be analysed by
spandsp, or some other opensource tool?

On T38 I can use wireshark, but on pure T30 it's useless...

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/a0eec597/attachment.html 

From miha at softnet.si  Wed Mar 25 10:40:07 2015
From: miha at softnet.si (Miha)
Date: Wed, 25 Mar 2015 08:40:07 +0100
Subject: [Freeswitch-users] ACK to wrong ip
Message-ID: <55126657.9090708@softnet.si>

Hi,

in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private 
ip) and not to ip from which request was recived (public ip)). I tried 
with <param name="NDLB-force-rport" value="true"/> on internal profile 
but still the same.

Must this be done on external profile or something else must be done?


tnx
miha


From gmaruzz at gmail.com  Wed Mar 25 11:04:59 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 25 Mar 2015 09:04:59 +0100
Subject: [Freeswitch-users] T30 fax analysis from G711 pcap
In-Reply-To: <CALXCt0oNpKYcZBzr1qFqEHUg5z7pH2unNScncGewgjfrt_ZodQ@mail.gmail.com>
References: <CALXCt0oNpKYcZBzr1qFqEHUg5z7pH2unNScncGewgjfrt_ZodQ@mail.gmail.com>
Message-ID: <CALXCt0p3QWo=mHB5Jt9M56UWLep56T--4xXJi42k9Mvixf9kmA@mail.gmail.com>

Hello all,

Steve Underwood graciously answered on IRC: there is a tool called
fax_decode in "tests" directory of spandsp.

Need to add "-lsndfile" to Makefile to compile it, after editing Makefile,
"make fax_decode".

Many thanks to Steve

-giovanni

On Wed, Mar 25, 2015 at 8:23 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Hello FreeSWITCHers,
>
> there is a way to extract T30 log from a g711 pcap?
>
> I must check on various possible incompatibilities and anomalies on a
> large number of different fax transmission, many of them in pure T30
> (faxmachine<->faxmachine).
>
> I would need something like
> http://www.netgencommunications.com/products/faxtapng/...
>
> Would it be possible to have a "replay" of the capture to be analysed by
> spandsp, or some other opensource tool?
>
> On T38 I can use wireshark, but on pure T30 it's useless...
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/f3f9f44f/attachment.html 

From royj at yandex.ru  Wed Mar 25 12:12:30 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 25 Mar 2015 12:12:30 +0300
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <55126657.9090708@softnet.si>
References: <55126657.9090708@softnet.si>
Message-ID: <752061427274750@web27o.yandex.ru>

try http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html

25.03.2015, 10:44, "Miha" <miha at softnet.si>:
> Hi,
>
> in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
> ip) and not to ip from which request was recived (public ip)). I tried
> with <param name="NDLB-force-rport" value="true"/> on internal profile
> but still the same.
>
> Must this be done on external profile or something else must be done?
>
> tnx
> miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From miha at softnet.si  Wed Mar 25 12:34:33 2015
From: miha at softnet.si (Miha)
Date: Wed, 25 Mar 2015 10:34:33 +0100
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <752061427274750@web27o.yandex.ru>
References: <55126657.9090708@softnet.si> <752061427274750@web27o.yandex.ru>
Message-ID: <55128129.7070305@softnet.si>

Hi tnx.

how to add this ({sip_sticky_contact=true}) if I have in bridge just 
${lcr_auto_route}' no not sofia/internal/... ?

tnx
miha

On 25/03/2015 10:12, royj at yandex.ru wrote:
> try http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>
> 25.03.2015, 10:44, "Miha" <miha at softnet.si>:
>> Hi,
>>
>> in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
>> ip) and not to ip from which request was recived (public ip)). I tried
>> with <param name="NDLB-force-rport" value="true"/> on internal profile
>> but still the same.
>>
>> Must this be done on external profile or something else must be done?
>>
>> tnx
>> miha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/e2bc6df7/attachment.html 

From royj at yandex.ru  Wed Mar 25 12:49:59 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Wed, 25 Mar 2015 12:49:59 +0300
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <55128129.7070305@softnet.si>
References: <55126657.9090708@softnet.si> <752061427274750@web27o.yandex.ru>
	<55128129.7070305@softnet.si>
Message-ID: <970951427276999@web26j.yandex.ru>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/3b02bb93/attachment.html 

From brian at freeswitch.org  Wed Mar 25 13:17:17 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 25 Mar 2015 05:17:17 -0500
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <970951427276999@web26j.yandex.ru>
References: <55126657.9090708@softnet.si> <752061427274750@web27o.yandex.ru>
	<55128129.7070305@softnet.si> <970951427276999@web26j.yandex.ru>
Message-ID: <CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>

Or fix your device to properly function? ;)

On Wednesday, March 25, 2015, <royj at yandex.ru> wrote:

> may be
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>
> 25.03.2015, 12:38, "Miha" <miha at softnet.si
> <javascript:_e(%7B%7D,'cvml','miha at softnet.si');>>:
>
> Hi tnx.
>
> how to add this ({sip_sticky_contact=true}) if I have in bridge just
> ${lcr_auto_route}' no not sofia/internal/... ?
>
> tnx
> miha
>
> On 25/03/2015 10:12, royj at yandex.ru
> <javascript:_e(%7B%7D,'cvml','royj at yandex.ru');> wrote:
>
> try http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>
> 25.03.2015, 10:44, "Miha" <miha at softnet.si> <javascript:_e(%7B%7D,'cvml','miha at softnet.si');>:
>
> Hi,
>
> in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
> ip) and not to ip from which request was recived (public ip)). I tried
> with <param name="NDLB-force-rport" value="true"/> on internal profile
> but still the same.
>
> Must this be done on external profile or something else must be done?
>
> tnx
> miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>http://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 <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>http://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
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/42d8e89e/attachment-0001.html 

From miha at softnet.si  Wed Mar 25 13:40:45 2015
From: miha at softnet.si (Miha)
Date: Wed, 25 Mar 2015 11:40:45 +0100
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>
References: <55126657.9090708@softnet.si>
	<752061427274750@web27o.yandex.ru>	<55128129.7070305@softnet.si>
	<970951427276999@web26j.yandex.ru>
	<CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>
Message-ID: <551290AD.40602@softnet.si>

Hi Brian,

on ther side is Asterisk which on prive IP behind nat, this is way I am 
trying to fix this as client on other side do not know how to do it :)

So what would be the best way to tell FS to send responses to request ip 
address?

tnx
miha

On 25/03/2015 11:17, Brian West wrote:
> Or fix your device to properly function? ;)
>
> On Wednesday, March 25, 2015, <royj at yandex.ru <mailto:royj at yandex.ru>> 
> wrote:
>
>     may be
>     https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>     25.03.2015, 12:38, "Miha" <miha at softnet.si>:
>>     Hi tnx.
>>
>>     how to add this ({sip_sticky_contact=true}) if I have in bridge
>>     just ${lcr_auto_route}' no not sofia/internal/... ?
>>
>>     tnx
>>     miha
>>
>>     On 25/03/2015 10:12, royj at yandex.ru wrote:
>>>     tryhttp://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>>>
>>>     25.03.2015, 10:44, "Miha"<miha at softnet.si>:
>>>>     Hi,
>>>>
>>>>     in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
>>>>     ip) and not to ip from which request was recived (public ip)). I tried
>>>>     with <param name="NDLB-force-rport" value="true"/> on internal profile
>>>>     but still the same.
>>>>
>>>>     Must this be done on external profile or something else must be done?
>>>>
>>>>     tnx
>>>>     miha
>>>>
>>>>     _________________________________________________________________________
>>>>     Professional FreeSWITCH Consulting Services:
>>>>     consulting at freeswitch.org
>>>>     http://www.freeswitchsolutions.com  <http://www.freeswitchsolutions.com/>
>>>>
>>>>     Official FreeSWITCH Sites
>>>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>>     http://confluence.freeswitch.org  <http://confluence.freeswitch.org/>
>>>>     http://www.cluecon.com  <http://www.cluecon.com/>
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at 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  <http://www.freeswitchsolutions.com/>
>>>
>>>     Official FreeSWITCH Sites
>>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>     http://confluence.freeswitch.org  <http://confluence.freeswitch.org/>
>>>     http://www.cluecon.com  <http://www.cluecon.com/>
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at 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
>>     <http://www.freeswitchsolutions.com/>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>     http://www.cluecon.com <http://www.cluecon.com/>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at 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/>
>>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers 
> <https://www.cluecon.com/call-for-speakers/> | Register 
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/e3627d2a/attachment.html 

From brian at freeswitch.org  Wed Mar 25 16:10:39 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 25 Mar 2015 08:10:39 -0500
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <551290AD.40602@softnet.si>
References: <55126657.9090708@softnet.si> <752061427274750@web27o.yandex.ru>
	<55128129.7070305@softnet.si> <970951427276999@web26j.yandex.ru>
	<CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>
	<551290AD.40602@softnet.si>
Message-ID: <CAEJMVkDcw4uvRm7JxJdAF8ihO0Hp1KYrsYziPnktv4f=UfSZiw@mail.gmail.com>

Asterisk has the same concept we do when it comes to that situation, there
is a local-network and an external ip setting for the sip peer/user, I'm of
the mindset that the client shouldn't be telling you lies.

On Wed, Mar 25, 2015 at 5:40 AM, Miha <miha at softnet.si> wrote:

>  Hi Brian,
>
> on ther side is Asterisk which on prive IP behind nat, this is way I am
> trying to fix this as client on other side do not know how to do it :)
>
> So what would be the best way to tell FS to send responses to request ip
> address?
>
> tnx
> miha
>
>
> On 25/03/2015 11:17, Brian West wrote:
>
> Or fix your device to properly function? ;)
>
> On Wednesday, March 25, 2015, <royj at yandex.ru> wrote:
>
>> may be
>> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>>
>> 25.03.2015, 12:38, "Miha" <miha at softnet.si>:
>>
>> Hi tnx.
>>
>> how to add this ({sip_sticky_contact=true}) if I have in bridge just
>> ${lcr_auto_route}' no not sofia/internal/... ?
>>
>> tnx
>> miha
>>
>> On 25/03/2015 10:12, royj at yandex.ru wrote:
>>
>> try http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>>
>> 25.03.2015, 10:44, "Miha" <miha at softnet.si>:
>>
>>  Hi,
>>
>> in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
>> ip) and not to ip from which request was recived (public ip)). I tried
>> with <param name="NDLB-force-rport" value="true"/> on internal profile
>> but still the same.
>>
>> Must this be done on external profile or something else must be done?
>>
>> tnx
>> miha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
>  *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/489e5315/attachment-0001.html 

From miha at softnet.si  Wed Mar 25 16:31:56 2015
From: miha at softnet.si (Miha)
Date: Wed, 25 Mar 2015 14:31:56 +0100
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <CAEJMVkDcw4uvRm7JxJdAF8ihO0Hp1KYrsYziPnktv4f=UfSZiw@mail.gmail.com>
References: <55126657.9090708@softnet.si>
	<752061427274750@web27o.yandex.ru>	<55128129.7070305@softnet.si>
	<970951427276999@web26j.yandex.ru>	<CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>	<551290AD.40602@softnet.si>
	<CAEJMVkDcw4uvRm7JxJdAF8ihO0Hp1KYrsYziPnktv4f=UfSZiw@mail.gmail.com>
Message-ID: <5512B8CC.9050908@softnet.si>

Brian tnx for this info.

Just one info know for next time. If I use "<param 
name="NDLB-force-rport" value="true"/>" on external profile this should 
work?

br
miha


On 25/03/2015 14:10, Brian West wrote:
> Asterisk has the same concept we do when it comes to that situation, 
> there is a local-network and an external ip setting for the sip 
> peer/user, I'm of the mindset that the client shouldn't be telling you 
> lies.
>
> On Wed, Mar 25, 2015 at 5:40 AM, Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     Hi Brian,
>
>     on ther side is Asterisk which on prive IP behind nat, this is way
>     I am trying to fix this as client on other side do not know how to
>     do it :)
>
>     So what would be the best way to tell FS to send responses to
>     request ip address?
>
>     tnx
>     miha
>
>
>     On 25/03/2015 11:17, Brian West wrote:
>>     Or fix your device to properly function? ;)
>>
>>     On Wednesday, March 25, 2015, <royj at yandex.ru
>>     <mailto:royj at yandex.ru>> wrote:
>>
>>         may be
>>         https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>>         25.03.2015, 12:38, "Miha" <miha at softnet.si>:
>>>         Hi tnx.
>>>
>>>         how to add this ({sip_sticky_contact=true}) if I have in
>>>         bridge just ${lcr_auto_route}' no not sofia/internal/... ?
>>>
>>>         tnx
>>>         miha
>>>
>>>         On 25/03/2015 10:12, royj at yandex.ru wrote:
>>>>         tryhttp://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>>>>
>>>>         25.03.2015, 10:44, "Miha"<miha at softnet.si>:
>>>>>         Hi,
>>>>>
>>>>>         in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
>>>>>         ip) and not to ip from which request was recived (public ip)). I tried
>>>>>         with <param name="NDLB-force-rport" value="true"/> on internal profile
>>>>>         but still the same.
>>>>>
>>>>>         Must this be done on external profile or something else must be done?
>>>>>
>>>>>         tnx
>>>>>         miha
>>>>>
>>>>>         _________________________________________________________________________
>>>>>         Professional FreeSWITCH Consulting Services:
>>>>>         consulting at freeswitch.org
>>>>>         http://www.freeswitchsolutions.com  <http://www.freeswitchsolutions.com/>
>>>>>
>>>>>         Official FreeSWITCH Sites
>>>>>         http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>>>         http://confluence.freeswitch.org  <http://confluence.freeswitch.org/>
>>>>>         http://www.cluecon.com  <http://www.cluecon.com/>
>>>>>
>>>>>         FreeSWITCH-users mailing list
>>>>>         FreeSWITCH-users at 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  <http://www.freeswitchsolutions.com/>
>>>>
>>>>         Official FreeSWITCH Sites
>>>>         http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>>         http://confluence.freeswitch.org  <http://confluence.freeswitch.org/>
>>>>         http://www.cluecon.com  <http://www.cluecon.com/>
>>>>
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at 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
>>>         <http://www.freeswitchsolutions.com/>
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>         http://confluence.freeswitch.org
>>>         <http://confluence.freeswitch.org/>
>>>         http://www.cluecon.com <http://www.cluecon.com/>
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at 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/>
>>>
>>
>>
>>     -- 
>>
>>     */Brian West/*
>>     brian at freeswitch.org <mailto:brian at freeswitch.org>
>>
>>
>>     */Twitter: @FreeSWITCH , @briankwest/*
>>     http://www.freeswitchbook.com
>>     http://www.freeswitchcookbook.com
>>
>>     ClueCon 2015 Call for Speakers
>>     <https://www.cluecon.com/call-for-speakers/> | Register
>>     <https://freeswitch.com/cart.php?gid=1> TODAY!
>>
>>     *T:*+19184209001 <tel:%2B19184209001> | *F:*+19184209002
>>     <tel:%2B19184209002> | *M:*+1918424WEST (9378)
>>     *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     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://confluence.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://confluence.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
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers 
> <https://www.cluecon.com/call-for-speakers/> | Register 
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/76b56e13/attachment.html 

From flokrrr at gmail.com  Wed Mar 25 17:15:06 2015
From: flokrrr at gmail.com (Florent Krieg)
Date: Wed, 25 Mar 2015 15:15:06 +0100
Subject: [Freeswitch-users] ACK to wrong ip
In-Reply-To: <5512B8CC.9050908@softnet.si>
References: <55126657.9090708@softnet.si> <752061427274750@web27o.yandex.ru>
	<55128129.7070305@softnet.si> <970951427276999@web26j.yandex.ru>
	<CAEJMVkAXixRRUBwZsi6jAGv7cNrz_FMy6Nmb5fmNfDSBwovCpg@mail.gmail.com>
	<551290AD.40602@softnet.si>
	<CAEJMVkDcw4uvRm7JxJdAF8ihO0Hp1KYrsYziPnktv4f=UfSZiw@mail.gmail.com>
	<5512B8CC.9050908@softnet.si>
Message-ID: <CAABKsuSnnVNNOrJ7Fdn_gg9nUjXqddrN6KGVjNqZw5HP46+6uw@mail.gmail.com>

Instructions externip can be used in Asterisk's global part of sip.conf to
fix this issue.

Florent
Le 25 mars 2015 14:36, "Miha" <miha at softnet.si> a ?crit :

>  Brian tnx for this info.
>
> Just one info know for next time. If I use "<param name="NDLB-force-rport"
> value="true"/>" on external profile this should work?
>
> br
> miha
>
>
> On 25/03/2015 14:10, Brian West wrote:
>
> Asterisk has the same concept we do when it comes to that situation, there
> is a local-network and an external ip setting for the sip peer/user, I'm of
> the mindset that the client shouldn't be telling you lies.
>
> On Wed, Mar 25, 2015 at 5:40 AM, Miha <miha at softnet.si> wrote:
>
>>  Hi Brian,
>>
>> on ther side is Asterisk which on prive IP behind nat, this is way I am
>> trying to fix this as client on other side do not know how to do it :)
>>
>> So what would be the best way to tell FS to send responses to request ip
>> address?
>>
>> tnx
>> miha
>>
>>
>> On 25/03/2015 11:17, Brian West wrote:
>>
>> Or fix your device to properly function? ;)
>>
>> On Wednesday, March 25, 2015, <royj at yandex.ru> wrote:
>>
>>> may be
>>> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>>>
>>> 25.03.2015, 12:38, "Miha" <miha at softnet.si>:
>>>
>>> Hi tnx.
>>>
>>> how to add this ({sip_sticky_contact=true}) if I have in bridge just
>>> ${lcr_auto_route}' no not sofia/internal/... ?
>>>
>>> tnx
>>> miha
>>>
>>> On 25/03/2015 10:12, royj at yandex.ru wrote:
>>>
>>> try http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/052888.html
>>>
>>> 25.03.2015, 10:44, "Miha" <miha at softnet.si>:
>>>
>>>  Hi,
>>>
>>> in my case FS send ACK to wrong ip (to ip in contact of 200 ok (private
>>> ip) and not to ip from which request was recived (public ip)). I tried
>>> with <param name="NDLB-force-rport" value="true"/> on internal profile
>>> but still the same.
>>>
>>> Must this be done on external profile or something else must be done?
>>>
>>> tnx
>>> miha
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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.orghttp://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>>  *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>  --
>
> *Brian West*
> brian at freeswitch.org
>
>
>  *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/3f066bb0/attachment-0001.html 

From olegstolyar at gmail.com  Wed Mar 25 21:51:42 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 25 Mar 2015 11:51:42 -0700
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and WebRTC
In-Reply-To: <CAOccsKCuBBNToFt3p49sAquBv4qJwuM-zNiPFUECcE-i6zFYHA@mail.gmail.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
	<00bb01d06599$2d2d75a0$878860e0$@botecomm.com>
	<CAOccsKCuBBNToFt3p49sAquBv4qJwuM-zNiPFUECcE-i6zFYHA@mail.gmail.com>
Message-ID: <CAC8JMO+fx8sL-MyvNyAcrfFAkpK3a79uYXCDJTe7fL0hR_Npkg@mail.gmail.com>

Hi Anthony,

Not sure if it will help but I am able to consistently reproduce this
problem on a Toshiba chromebook.  It happens with 41, 42 and 43 builds and
multiple reboots did not solve it.

If you need a way to reproduce the problem I'll be happy to ship this
chromebook to you to keep.

On Mon, Mar 23, 2015 at 10:04 PM, ik <idokan at gmail.com> wrote:

> There was a security update last weekend for Chrome, they might have
> solved this and placed it on that version.
>
> Ido
> On Mar 23, 2015 8:45 PM, "Bote Man" <bote_radio at botecomm.com> wrote:
>
>> I am running
>>
>> Chrome Version 41.0.2272.101 m
>>
>> on Windows 7 and checked the 888 FreeSWITCH conference yesterday, it had
>> audio both directions. I tried it 1 or 2 weeks ago and it had no audio so
>> it must have updated itself behind my back because I did not explicitly
>> update Chrome.
>>
>>
>>
>> Firefox 36.0.1 did not have any audio last week and now 36.0.4 has no
>> audio either.
>>
>>
>>
>> I am testing this on my home network using a Cisco RV042 router and the
>> default checkboxes on the FS WebRTC page which apparently default to using
>> STUN.
>>
>>
>>
>> Anthm has stated many times that WebRTC is a moving target so enjoy the
>> exhilaration of these rapid developments! I believe this is why he insists
>> that WebRTC users keep FreeSWITCH updated so that it tracks the changes in
>> the major web browsers.
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>>
>>
>> *From:* Oleg Stolyar
>> *Sent:* Monday, 23 March, 2015 14:21
>> *Subject:* [Freeswitch-users] Intermittent audio issue with Chrome 40-42
>> and WebRTC
>>
>>
>>
>> Hi guys,
>>
>>
>>
>> Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
>> then they complain that they cannot hear audio from FreeSWITCH or send
>> audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
>> several occurrences.  It happens on both Windows and Macs.  Here are some
>> details"
>>
>>
>>
>> 1. Usually the problem is solved if we close Chrome and make sure all the
>> Chrome processes are killed.
>>
>>
>>
>> 2. When the problem happens, FS logs this on connection:
>>
>> [ERR] switch_rtp.c:2976 audio DTLS packet not written
>>
>>
>>
>> 3. I tried using this WebRTC test on an affected computer:
>>
>> https://janus.conf.meetecho.com/echotest.html
>>
>> and that worked, so it is not a general WebRTC issue.
>>
>>
>>
>> 4. I tried FS WebRTC demo page
>>
>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>
>> and on the affected computer the problem was still there - no audio.
>>
>>
>>
>> 5. This happened both on an old FS version from last May and on a very
>> recent master FreeSWITCH Version
>> 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
>> 16:44:11Z 64bit)
>>
>>
>>
>> Any ideas?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/7aca2a82/attachment.html 

From olegstolyar at gmail.com  Wed Mar 25 22:13:24 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 25 Mar 2015 12:13:24 -0700
Subject: [Freeswitch-users] Intermittent audio issue with Chrome 40-42
	and WebRTC
In-Reply-To: <CAC8JMO+fx8sL-MyvNyAcrfFAkpK3a79uYXCDJTe7fL0hR_Npkg@mail.gmail.com>
References: <CAC8JMOJVE+E4mwrR9A93qwTV8AcXzUk180PCQEfwePuNmkujXQ@mail.gmail.com>
	<00bb01d06599$2d2d75a0$878860e0$@botecomm.com>
	<CAOccsKCuBBNToFt3p49sAquBv4qJwuM-zNiPFUECcE-i6zFYHA@mail.gmail.com>
	<CAC8JMO+fx8sL-MyvNyAcrfFAkpK3a79uYXCDJTe7fL0hR_Npkg@mail.gmail.com>
Message-ID: <CAC8JMO+exon2fkXbxcKUxT1yte5P5Uizj180D27OiMQE8sqkoA@mail.gmail.com>

Funny thing (kind of).  After I switched from wired network to wifi, it
started working,  Then when I disabled wifi and went back to wired, it kept
right on working.  But after a reboot was again broken on wired.  So far,
it all seems reproducible and matches what you said about Chrome mixing up
devices.

My offer to ship it to you still stands



On Wed, Mar 25, 2015 at 11:51 AM, Oleg Stolyar <olegstolyar at gmail.com>
wrote:

> Hi Anthony,
>
> Not sure if it will help but I am able to consistently reproduce this
> problem on a Toshiba chromebook.  It happens with 41, 42 and 43 builds and
> multiple reboots did not solve it.
>
> If you need a way to reproduce the problem I'll be happy to ship this
> chromebook to you to keep.
>
> On Mon, Mar 23, 2015 at 10:04 PM, ik <idokan at gmail.com> wrote:
>
>> There was a security update last weekend for Chrome, they might have
>> solved this and placed it on that version.
>>
>> Ido
>> On Mar 23, 2015 8:45 PM, "Bote Man" <bote_radio at botecomm.com> wrote:
>>
>>> I am running
>>>
>>> Chrome Version 41.0.2272.101 m
>>>
>>> on Windows 7 and checked the 888 FreeSWITCH conference yesterday, it had
>>> audio both directions. I tried it 1 or 2 weeks ago and it had no audio so
>>> it must have updated itself behind my back because I did not explicitly
>>> update Chrome.
>>>
>>>
>>>
>>> Firefox 36.0.1 did not have any audio last week and now 36.0.4 has no
>>> audio either.
>>>
>>>
>>>
>>> I am testing this on my home network using a Cisco RV042 router and the
>>> default checkboxes on the FS WebRTC page which apparently default to using
>>> STUN.
>>>
>>>
>>>
>>> Anthm has stated many times that WebRTC is a moving target so enjoy the
>>> exhilaration of these rapid developments! I believe this is why he insists
>>> that WebRTC users keep FreeSWITCH updated so that it tracks the changes in
>>> the major web browsers.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* Oleg Stolyar
>>> *Sent:* Monday, 23 March, 2015 14:21
>>> *Subject:* [Freeswitch-users] Intermittent audio issue with Chrome
>>> 40-42 and WebRTC
>>>
>>>
>>>
>>> Hi guys,
>>>
>>>
>>>
>>> Ever since my users switched to Chrome 40, 41 or 42 beta, every now and
>>> then they complain that they cannot hear audio from FreeSWITCH or send
>>> audio to FreeSWITCH.  I cannot reproduce this at will but was able to catch
>>> several occurrences.  It happens on both Windows and Macs.  Here are some
>>> details"
>>>
>>>
>>>
>>> 1. Usually the problem is solved if we close Chrome and make sure all
>>> the Chrome processes are killed.
>>>
>>>
>>>
>>> 2. When the problem happens, FS logs this on connection:
>>>
>>> [ERR] switch_rtp.c:2976 audio DTLS packet not written
>>>
>>>
>>>
>>> 3. I tried using this WebRTC test on an affected computer:
>>>
>>> https://janus.conf.meetecho.com/echotest.html
>>>
>>> and that worked, so it is not a general WebRTC issue.
>>>
>>>
>>>
>>> 4. I tried FS WebRTC demo page
>>>
>>> https://webrtc.freeswitch.org/verto/index.html#page-main
>>>
>>> and on the affected computer the problem was still there - no audio.
>>>
>>>
>>>
>>> 5. This happened both on an old FS version from last May and on a very
>>> recent master FreeSWITCH Version
>>> 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16
>>> 16:44:11Z 64bit)
>>>
>>>
>>>
>>> Any ideas?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/b3904efa/attachment-0001.html 

From ing.antonyam at gmail.com  Thu Mar 26 02:50:45 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Wed, 25 Mar 2015 17:50:45 -0600
Subject: [Freeswitch-users] Error Mod Nibblebill.
Message-ID: <CABFbAiUx5xu_1LfNM3Vt-MToLrByAsp7X2PXJdU8ZVwZRQkZmA@mail.gmail.com>

I have a problem with my configuration of nibblebill module, is configured
so that when the cash = 0 send the call to hangup, but brand sip Error 483
- Too many hops. My configuration is:

mod_nibblebill:
<param name = "nobal_amt" value = "0" />
<param name = "nobal_action" value = "hangup XML default" />

I have created in the dialplan extencion the following:

<extension name = "hangup">
   <condition field = "destination_number" expression = "^ (hangup) $">
     <action application = "playback" data = "voicemail / vm-continue.wav"
/>
     <action application = "hangup" />
   </ condition>
</ extension>

but does not work in my error log I find the following:

Dialplan: sofia/internal/8117 at 1.1.1.1 Regex (PASS) [mierda]
destination_number(mierda) =~ /^(mierda)$/ break=on-false
Dialplan: sofia/internal/8117 at 1.1.1.1 Action
playback(voicemail/vm-continue.wav)
Dialplan: sofia/internal/8117 at 1.1.1.1 Action hangup()
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/8117 at 1.1.1.1) State Change CS_ROUTING -> CS_EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING going to sleep
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] switch_channel.c:2169 (sofia/internal/
8117 at 1.1.1.1) Callstate Change RINGING -> EARLY
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/8117 at 1.1.1.1) State EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] mod_sofia.c:243 sofia/internal/
8117 at 1.1.1.1 SOFIA EXECUTE
2015-03-25 12:16:38.365165 [INFO] switch_core_session.c:1511 sofia/internal/
8117 at 1.1.1.1 setting session heartbeat to 60 second(s).
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:209
sofia/internal/8117 at 1.1.1.1 Standard EXECUTE
EXECUTE sofia/internal/8117 at 1.1.1.1 set(open=true)
2015-03-25 12:16:38.365165 [DEBUG] mod_dptools.c:1402 sofia/internal/
8117 at 1.1.1.1 SET [open]=[true]
EXECUTE sofia/internal/8117 at 1.1.1.1 playback(voicemail/vm-continue.wav)
2015-03-25 12:16:38.365165 [DEBUG] switch_ivr_play_say.c:1305 Codec
Activated L16 at 8000hz 1 channels 20ms
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:643 Received request
via SESSION_HEARTBEAT!
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.365165 [DEBUG] switch_ivr.c:1834 (sofia/internal/
8117 at 1.1.1.1) State Change CS_EXECUTE -> CS_ROUTING
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:871 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.365165 [NOTICE] switch_ivr.c:1841 Transfer
sofia/internal/8117 at 1.1.1.1 to XML[mierda at default]
2015-03-25 12:16:38.385162 [DEBUG] switch_ivr_play_say.c:1708 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-continue.wav
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/8117 at 1.1.1.1) State EXECUTE going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_ROUTING
2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:2165 (sofia/internal/
8117 at 1.1.1.1) Callstate Change EARLY -> RINGING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:150 sofia/internal/
8117 at 1.1.1.1 SOFIA ROUTING
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.385162 [NOTICE] switch_ivr.c:1736 Hangup sofia/internal/
8117 at 1.1.1.1 [CS_ROUTING] [EXCHANGE_ROUTING_ERROR]
2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:3189 Send signal
sofia/internal/8117 at 1.1.1.1 [KILL]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_HANGUP
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:681
(sofia/internal/8117 at 1.1.1.1) Callstate Change RINGING -> HANGUP
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683
(sofia/internal/8117 at 1.1.1.1) State HANGUP
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:506 Channel sofia/internal/
8117 at 1.1.1.1 hanging up, cause: EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:640 Responding to INVITE
with: 483
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:48
sofia/internal/8117 at 1.1.1.1 Standard HANGUP, cause: EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683
(sofia/internal/8117 at 1.1.1.1) State HANGUP going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:450
(sofia/internal/8117 at 1.1.1.1) State Change CS_HANGUP -> CS_REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767
(sofia/internal/8117 at 1.1.1.1) State REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:92
sofia/internal/8117 at 1.1.1.1 Standard REPORTING, cause:
EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767
(sofia/internal/8117 at 1.1.1.1) State REPORTING going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:444
(sofia/internal/8117 at 1.1.1.1) State Change CS_REPORTING -> CS_DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1 [BREAK]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1559 Session 201
(sofia/internal/8117 at 1.1.1.1) Locked, Waiting on external entities
2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1577 Session 201
(sofia/internal/8117 at 1.1.1.1) Ended
2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1581 Close
Channel sofia/internal/8117 at 1.1.1.1 [CS_DESTROY]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:572
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582
(sofia/internal/8117 at 1.1.1.1) State DESTROY
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:399 sofia/internal/
8117 at 1.1.1.1 SOFIA DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:99
sofia/internal/8117 at 1.1.1.1 Standard DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582
(sofia/internal/8117 at 1.1.1.1) State DESTROY going to sleep


any ideas?

The design is based on a cost for all of the following linkhttps://
freeswitch.org/confluence/display/FREESWITCH/mod_nibblebill#mod_nibblebill-Installationandconfiguration
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150325/9974510c/attachment.html 

From ashokkuanar at gmail.com  Thu Mar 26 07:12:51 2015
From: ashokkuanar at gmail.com (Ashok kumar Kuanar)
Date: Thu, 26 Mar 2015 09:42:51 +0530
Subject: [Freeswitch-users] Preferred Database and billing software for
	freeSwitch1.4.17
Message-ID: <CADzzAs5Hd+uZOn=u6WW-phbLcc61XtBjvZueBCxyEOWfRikwLA@mail.gmail.com>

Hi all,

I am New to FreeSwitch , i have installed FreeSwitch 1.4.17 on CentOS 7, it
will be highly appreciated if  any one suggest me which billing software(
A2 billing / ASTPP) with versionis more suitable for FreeSwitch as well as
which Database is preferred for this version,



Regards,
Ashok
+91-8796111401



>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/4afab670/attachment.html 

From k.presler at megafit.su  Thu Mar 26 13:09:20 2015
From: k.presler at megafit.su (=?koi8-r?B?68nSyczMIPDSxdPMxdI=?=)
Date: Thu, 26 Mar 2015 10:09:20 +0000
Subject: [Freeswitch-users] Change SIP Contact header  on b-leg
Message-ID: <HE1PR06MB11309B44ECF52155228F7A7B8D080@HE1PR06MB1130.eurprd06.prod.outlook.com>

Provider not accepting calls because of "bad" Contact header
sip:gw+ss at 10.50.244.15:5060;transport=udp;gw=ss

ITSP needs sip:username at 10.50.244.15:5060, where username should be passed from dialplan for every call and there is no params.


I've tried following variables with set and export, but nothing seems to work.


sip_contact_user
sip_outgoing_contact_uri
sip_invite_contact_params


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/d826c8f8/attachment-0001.html 

From asilva at wirelessmundi.com  Thu Mar 26 14:01:17 2015
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Thu, 26 Mar 2015 12:01:17 +0100
Subject: [Freeswitch-users] Change SIP Contact header  on b-leg
In-Reply-To: <HE1PR06MB11309B44ECF52155228F7A7B8D080@HE1PR06MB1130.eurprd06.prod.outlook.com>
References: <HE1PR06MB11309B44ECF52155228F7A7B8D080@HE1PR06MB1130.eurprd06.prod.outlook.com>
Message-ID: <5513E6FD.7010005@wirelessmundi.com>

Not sure but...

You can force the contact extension in the gateway... instead of having 
the "gw+ss"

The parameter is

extension = your_provider_expected_extension
extension-in-contact  = true

see:

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Gateway+Authentication+Params

Regards,
Ant?nio

On 03/26/2015 11:09 AM, ?????? ??????? wrote:
>
> Provider not accepting calls because of ?bad? Contact header
>
> sip:gw+ss at 10.50.244.15:5060;transport=udp;gw=ss
>
> ITSP needs sip:username at 10.50.244.15:5060 
> <sip:username at 10.50.244.15:5060>, where username should be passed from 
> dialplan for every call and there is no params.
>
> I?ve tried following variables with set and export, but nothing seems 
> to work.
>
> sip_contact_user
>
> sip_outgoing_contact_uri
>
> sip_invite_contact_params
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
---
Ant?nio Silva

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/2daf0a3f/attachment.html 

From lists at telefaks.de  Thu Mar 26 15:11:19 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 26 Mar 2015 13:11:19 +0100
Subject: [Freeswitch-users] eliminating fax header line
In-Reply-To: <CAJNDQt67cmqH8=B-iSPh-cRvZP9PLziG+OpWVHXt2YarrwLkcA@mail.gmail.com>
References: <CAJNDQt7J6hnPt6B4S7GM03X9BMFt_6WH7ydLKT0zYF2SUiyFkw@mail.gmail.com>
	<CAJNDQt7rBZWSR=Y5+r8Gmk4GKXydahgEn9voYAgNRxQeR=bbGA@mail.gmail.com>
	<CAJNDQt67cmqH8=B-iSPh-cRvZP9PLziG+OpWVHXt2YarrwLkcA@mail.gmail.com>
Message-ID: <5513F767.9050007@telefaks.de>

Just for the records:

With current Freeswitch I could eleminate the Fax header line completely
by patching

/usr/src/freeswitch/src/mod/applications/mod_spandsp/mod_spandsp.c
        spandsp_globals.ident = "SpanDSP Fax Ident";
        spandsp_globals.header = "SpanDSP Fax Header";

to
        spandsp_globals.ident = "";
        spandsp_globals.header = "";

Then you can set fax_ident='', fax_header='' in your dialstring and get
an empty header

Diff for this:

diff mod_spandsp.c mod_spandsp.c.org
518,519c518,519
<       spandsp_globals.ident = "";
<       spandsp_globals.header = "";
---
>       spandsp_globals.ident = "SpanDSP Fax Ident";
>       spandsp_globals.header = "SpanDSP Fax Header";


Best regards
Peter


On 05/28/14 22:29, Bruce Lefko wrote:
> Hi Michael.
>
> I tried the '  ', but same thing: the parameter does not show up as
> parsed and the harcoded default is what is used.
>
>
>
> On Wed, May 28, 2014 at 2:58 PM, Bruce Lefko <blefko5361 at gmail.com
> <mailto:blefko5361 at gmail.com>> wrote:
>
>     Hi Steve.
>
>     I checked into the legality of removing the header line, and it
>     turns out this is only a problem for advertisements, which we are
>     not doing.
>
>     As far as getting the header removed, I tried doing the following:
>
>     originate
>     {ignore_early_media=true,absolute_codec_string='PCMU,PCMA',fax_enable_t38=true,fax_verbose=true,fax_use_ecm=true,fax_enable_t38_request=true,origination_caller_id_number='A
>     CALLER ID',origination_caller_id_name='A CALLER ID
>     NAME',call_timeout=120,fax_ident='a fax
>     ident',fax_header=}sofia/gateway/outbound/+15555555555
>     &txFax('/tmp/sample.tiff')
>
>     Even if I remove the default headers in spandsp.conf.xml and
>     fax.conf.xml I still get "SpanDSP Fax Header".  I noticed that
>     this is hardcoded in
>     "src/mod/applications/mod_spandsp/mod_spandsp.c" line 527.
>
>     Am I not passing the null string properly somehow via originate?
>      I noticed in my logs that there are many lines for
>     "switch_event.c:1661 Parsing variable" for my other parameters,
>     but not for my fax_header parameter.
>
>     Thanks!
>
>
>
>     On Thu, May 8, 2014 at 3:16 PM, Bruce Lefko <blefko5361 at gmail.com
>     <mailto:blefko5361 at gmail.com>> wrote:
>
>         I'd like to completely eliminate the header line sent at the
>         top of each fax page using spandsp.  i've set fax-header to a
>         null string, but that doesn't seem to be removing it, and it
>         defaults back to 'Spandsp fax header'.  This seems to be the
>         default in the source code in 1.2.
>
>         What do I have to do to eliminate the header line?  Do I have
>         to make changes to freeswitch source?
>
>         Thanks!
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/079c6c83/attachment.html 

From italorossib at gmail.com  Thu Mar 26 16:37:31 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 26 Mar 2015 10:37:31 -0300
Subject: [Freeswitch-users] eliminating fax header line
In-Reply-To: <5513F767.9050007@telefaks.de>
References: <CAJNDQt7J6hnPt6B4S7GM03X9BMFt_6WH7ydLKT0zYF2SUiyFkw@mail.gmail.com>
	<CAJNDQt7rBZWSR=Y5+r8Gmk4GKXydahgEn9voYAgNRxQeR=bbGA@mail.gmail.com>
	<CAJNDQt67cmqH8=B-iSPh-cRvZP9PLziG+OpWVHXt2YarrwLkcA@mail.gmail.com>
	<5513F767.9050007@telefaks.de>
Message-ID: <CAJR7f2E4hGqpqAfO5=Nh78nRHuWk6M=RQhAY2RvzScLMaBO=Pw@mail.gmail.com>

Current code allows you to set _undef_ in config file:

} else if (!strcmp(name, "ident")) {
                    if (!strcmp(value, "_undef_")) {
                        spandsp_globals.ident = "";
                    } else {
                        spandsp_globals.ident =
switch_core_strdup(spandsp_globals.config_pool, value);
                    }


On Thu, Mar 26, 2015 at 9:11 AM, Peter Steinbach <lists at telefaks.de> wrote:

>  Just for the records:
>
> With current Freeswitch I could eleminate the Fax header line completely
> by patching
>
> /usr/src/freeswitch/src/mod/applications/mod_spandsp/mod_spandsp.c
>         spandsp_globals.ident = "SpanDSP Fax Ident";
>         spandsp_globals.header = "SpanDSP Fax Header";
>
> to
>         spandsp_globals.ident = "";
>         spandsp_globals.header = "";
>
> Then you can set fax_ident='', fax_header='' in your dialstring and get an
> empty header
>
> Diff for this:
>
> diff mod_spandsp.c mod_spandsp.c.org
> 518,519c518,519
> <       spandsp_globals.ident = "";
> <       spandsp_globals.header = "";
> ---
> >       spandsp_globals.ident = "SpanDSP Fax Ident";
> >       spandsp_globals.header = "SpanDSP Fax Header";
>
>
> Best regards
> Peter
>
>
>
> On 05/28/14 22:29, Bruce Lefko wrote:
>
> Hi Michael.
>
>  I tried the '  ', but same thing: the parameter does not show up as
> parsed and the harcoded default is what is used.
>
>
>
> On Wed, May 28, 2014 at 2:58 PM, Bruce Lefko <blefko5361 at gmail.com> wrote:
>
>> Hi Steve.
>>
>>  I checked into the legality of removing the header line, and it turns
>> out this is only a problem for advertisements, which we are not doing.
>>
>>  As far as getting the header removed, I tried doing the following:
>>
>>  originate
>> {ignore_early_media=true,absolute_codec_string='PCMU,PCMA',fax_enable_t38=true,fax_verbose=true,fax_use_ecm=true,fax_enable_t38_request=true,origination_caller_id_number='A
>> CALLER ID',origination_caller_id_name='A CALLER ID
>> NAME',call_timeout=120,fax_ident='a fax
>> ident',fax_header=}sofia/gateway/outbound/+15555555555
>> &txFax('/tmp/sample.tiff')
>>
>>  Even if I remove the default headers in spandsp.conf.xml and
>> fax.conf.xml I still get "SpanDSP Fax Header".  I noticed that this is
>> hardcoded in "src/mod/applications/mod_spandsp/mod_spandsp.c" line 527.
>>
>>  Am I not passing the null string properly somehow via originate?  I
>> noticed in my logs that there are many lines for "switch_event.c:1661
>> Parsing variable" for my other parameters, but not for my fax_header
>> parameter.
>>
>>  Thanks!
>>
>>
>>
>> On Thu, May 8, 2014 at 3:16 PM, Bruce Lefko <blefko5361 at gmail.com> wrote:
>>
>>> I'd like to completely eliminate the header line sent at the top of each
>>> fax page using spandsp.  i've set fax-header to a null string, but that
>>> doesn't seem to be removing it, and it defaults back to 'Spandsp fax
>>> header'.  This seems to be the default in the source code in 1.2.
>>>
>>>  What do I have to do to eliminate the header line?  Do I have to make
>>> changes to freeswitch source?
>>>
>>>  Thanks!
>>>
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.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
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/570f25b3/attachment-0001.html 

From mkvonarx at gmail.com  Thu Mar 26 17:20:25 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Thu, 26 Mar 2015 15:20:25 +0100
Subject: [Freeswitch-users] Silence Suppression from an Audio Conference
In-Reply-To: <001801d06644$1f31efc0$5d95cf40$@botecomm.com>
References: <CAHA5ALf769OHaFbpkML5QNxn6X+2wDyzEm7W1PwsPJ+5gpLg8A@mail.gmail.com>
	<CAFiqYummY=9RXvsBALFpTEEYbhM_8KF3BsqAW2Br9YHcfTyVdg@mail.gmail.com>
	<CAHA5ALe=fXufoAnjCSdH1cG73f4w=na1STS+sxAwC-9ReE0FaQ@mail.gmail.com>
	<001801d06644$1f31efc0$5d95cf40$@botecomm.com>
Message-ID: <CAHA5ALfmXh0PnxnT14p2PiS2WZsCTU=-E_DZAKu-jL1HJGcVEg@mail.gmail.com>

Hi Bote

I did play around with those settings some more and came to the conclusion,
that VAD and CNG do indeed work. Just not in the way I want. I observed
that the FreeSWITCH always continues to send G.711 PCMU RTP packets every
20 milliseconds no matter if the RTP packets carry user voice or CNG
payload, and that the FreeSWITCH never uses RTP type 13 for sending CNG,
and that the FreeSWITCH never does actual silence suppression. I'll open
another discussion with more specific questions about silence suppression
without the "distraction" of audio conferencing.

Markus


2015-03-24 16:06 GMT+01:00 Bote Man <bote_radio at botecomm.com>:

> There is a setting listed in
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>
>
>
> energy-level
>
>
>
> which acts as a noise gate. If you set this number high enough, the
> conference bridge will only admit audio from a conferee when it detects
> speech (or noise?) from him.
>
>
>
> HOWEVER, there used to be a conference flag named ?waste? that told the
> conference to ?waste bandwidth? by transmitting packets all the time, even
> when there was no audio contained in them; now that flag has been
> eliminated and I understand that the conference bridge always sends
> packets. If I have this correct, then even the noise gate will not reduce
> your bandwidth.
>
>
>
> I recommend you test this theory in case it is helpful and please report
> back with your findings.
>
>
>
> Thanks.
>
>
>
> Bote
>
>
>
>
>
> *From:* Markus von Arx
> *Sent:* Tuesday, 24 March, 2015 08:58
> *Subject:* Re: [Freeswitch-users] Silence Suppression from an Audio
> Conference
>
>
>
> Hi Steven
>
>
>
> Thanks for your reply. I actually already know that wiki page. But all
> those configuration variables there don't work - at least not for SIP
> channels that are connected to a mod_conference audio conference. Maybe
> they do work for bridged calls, but that's not what I need. Also, the wiki
> page does not mention conferences at all. And the sentence "When FreeSWITCH
> does not detect speech, it stops transmitting RTP" seems not to apply to
> mod_conference.
>
> ? I probably just have configured mod_conference incorrectly, but I don't
> know where to check.
>
>
>
> So any information or advice about SIP channels connected to a
> mod_conference audio conference?
>
>
>
> Thanks, Markus
>
>
>
>
>
> 2015-03-24 11:43 GMT+01:00 Steven Ayre <steveayre at gmail.com>:
>
> https://wiki.freeswitch.org/wiki/VAD_and_CNG
>
>
>
> On 24 March 2015 at 07:04, Markus von Arx <mkvonarx at gmail.com> wrote:
>
> Hi
>
>
>
> Can anyone tell me if FreeSWITCH supports silence suppression for SIP
> calls that are inside a FreeSWITCH audio conference? If yes, how do I
> configure mod_conference, mod_sofia and FreeSWITCH core to enable this
> feature?
>
>
>
> More precisely, I try to enable/activate the behavior described in RFC
> 3389 for G.711 in such a way that there are only RTP packets of type 13
> every 1 or 2 seconds. I tried to play around with some possible settings
> but could never observe anything else then the regular G.711 PCMU RTP
> packets on the wire. Even when I set the SIP call to 'deaf' via the
> FreeSWITCH console, mod_conference/mod_sofia continue to send G.711 PCMU
> RTP packets every 20ms.
>
>
>
> It's possible that I completly misunderstand RFC 3389 and the concepts of
> silence suppression, comfort noise etc. In the end, what I try
> to achieve is to reduce the network bandwidth of a G.711 SIP channel during
> the periods when the FreeSWITCH only sends silence over the SIP channel.
> Unfortunately, we're stuck with G.711 at the moment, so I cannot switch to
> another codec.
>
>
>
> Thanks, Markus
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/33ec18f0/attachment.html 

From mkvonarx at gmail.com  Thu Mar 26 17:33:15 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Thu, 26 Mar 2015 15:33:15 +0100
Subject: [Freeswitch-users] Support for actual Silence Suppression
Message-ID: <CAHA5ALdHOXni_91JuHwaQn53QQxLOX+0rRGitPp-2ZkT1bQm_Q@mail.gmail.com>

Hi

I'd like to configure FreeSWITCH in such a way that it does actual real
silence suppression and reduce the used network bandwidth during silence
periods.

By actual real silence suppression I mean that it uses one of the two
options described in RFC 3389 chapter 5, so either
1) Sending RTP packets with payload 13 that carry CNG description instead
of PCMU RTP packets during silence periods, and send those RTP payload 13
packets much less often than every 20 milliseconds, or
2) Use discontinuous RTP transmission during silence periods, or in other
words send PCMU RTP packets much less often than every 20 milliseconds.

So my questions:
1) Does FreeSWITCH support on of those two options described above?
2) If yes, how do I configure FreeSWITCH to actually do it?

Btw: I do know about
https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG, and I
have tried most of the things described there. What I found (by using
Wireshark) is that FreeSWITCH does indeed support VAD and CNG, but that the
FreeSWITCH does not use that information to send RTP payload 13 CNG packets
nor to do discontinuous RTP transmission, but instead the FreeSWITCH
continues to send PCMU RTP packets every 20 milliseconds and puts the CNG
audio inside these PCMU packets. This "works" in the way that the remote
party hears CNG, but this is not silence suppression as it does not reduce
the network bandwidth.

Thanks a lot,
Markus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/7cea52f9/attachment.html 

From ing.antonyam at gmail.com  Thu Mar 26 18:23:52 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Thu, 26 Mar 2015 09:23:52 -0600
Subject: [Freeswitch-users] Help Mod_Nibblebill
Message-ID: <CABFbAiVrFpLGCypo+fBbG58GeRX35rJGpPj7DdSMc0ZRD5YO5g@mail.gmail.com>

I have a problem with my configuration of nibblebill module, is configured
so that when the cash = 0 send the call to hangup, but brand sip Error 483
- Too many hops. My configuration is:

mod_nibblebill:
<param name = "nobal_amt" value = "0" />
<param name = "nobal_action" value = "hangup XML default" />

I have created in the dialplan extencion the following:

<extension name = "hangup">
   <condition field = "destination_number" expression = "^ (hangup) $">
     <action application = "playback" data = "voicemail / vm-continue.wav"
/>
     <action application = "hangup" />
   </ condition>
</ extension>

but does not work in my error log I find the following:

Dialplan: sofia/internal/8117 at 1.1.1.1 Regex (PASS) [mierda]
destination_number(mierda) =~ /^(mierda)$/ break=on-false
Dialplan: sofia/internal/8117 at 1.1.1.1 Action
playback(voicemail/vm-continue.wav)
Dialplan: sofia/internal/8117 at 1.1.1.1 Action hangup()
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:167
(sofia/internal/8117 at 1.1.1.1) State Change CS_ROUTING -> CS_EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING going to sleep
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] switch_channel.c:2169 (sofia/internal/
8117 at 1.1.1.1) Callstate Change RINGING -> EARLY
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/8117 at 1.1.1.1) State EXECUTE
2015-03-25 12:16:38.365165 [DEBUG] mod_sofia.c:243 sofia/internal/
8117 at 1.1.1.1 SOFIA EXECUTE
2015-03-25 12:16:38.365165 [INFO] switch_core_session.c:1511 sofia/internal/
8117 at 1.1.1.1 setting session heartbeat to 60 second(s).
2015-03-25 12:16:38.365165 [DEBUG] switch_core_state_machine.c:209
sofia/internal/8117 at 1.1.1.1 Standard EXECUTE
EXECUTE sofia/internal/8117 at 1.1.1.1 set(open=true)
2015-03-25 12:16:38.365165 [DEBUG] mod_dptools.c:1402 sofia/internal/
8117 at 1.1.1.1 SET [open]=[true]
EXECUTE sofia/internal/8117 at 1.1.1.1 playback(voicemail/vm-continue.wav)
2015-03-25 12:16:38.365165 [DEBUG] switch_ivr_play_say.c:1305 Codec
Activated L16 at 8000hz 1 channels 20ms
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:643 Received request
via SESSION_HEARTBEAT!
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.365165 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.365165 [DEBUG] switch_ivr.c:1834 (sofia/internal/
8117 at 1.1.1.1) State Change CS_EXECUTE -> CS_ROUTING
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.365165 [DEBUG] switch_core_session.c:871 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.365165 [NOTICE] switch_ivr.c:1841 Transfer
sofia/internal/8117 at 1.1.1.1 to XML[mierda at default]
2015-03-25 12:16:38.385162 [DEBUG] switch_ivr_play_say.c:1708 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-continue.wav
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:481
(sofia/internal/8117 at 1.1.1.1) State EXECUTE going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_ROUTING
2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:2165 (sofia/internal/
8117 at 1.1.1.1) Callstate Change EARLY -> RINGING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:150 sofia/internal/
8117 at 1.1.1.1 SOFIA ROUTING
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.385162 [NOTICE] switch_ivr.c:1736 Hangup sofia/internal/
8117 at 1.1.1.1 [CS_ROUTING] [EXCHANGE_ROUTING_ERROR]
2015-03-25 12:16:38.385162 [DEBUG] switch_channel.c:3189 Send signal
sofia/internal/8117 at 1.1.1.1 [KILL]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:474
(sofia/internal/8117 at 1.1.1.1) State ROUTING going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_HANGUP
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:681
(sofia/internal/8117 at 1.1.1.1) Callstate Change RINGING -> HANGUP
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683
(sofia/internal/8117 at 1.1.1.1) State HANGUP
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:506 Channel sofia/internal/
8117 at 1.1.1.1 hanging up, cause: EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:640 Responding to INVITE
with: 483
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:486 Attempting to bill
at $1 per minute to account 1
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:498 Not billing 1 -
call is not in answered state
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:502 Comparing 0.000000
to hangup balance of 0.000000
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:505 Balance of 0.000000
fell below allowed amount of 0.000000! (Account 1)
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:418 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='1']
2015-03-25 12:16:38.385162 [DEBUG] mod_nibblebill.c:426 Retrieved current
balance for account 1 (balance = 0.000000)
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:48
sofia/internal/8117 at 1.1.1.1 Standard HANGUP, cause: EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:683
(sofia/internal/8117 at 1.1.1.1) State HANGUP going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:450
(sofia/internal/8117 at 1.1.1.1) State Change CS_HANGUP -> CS_REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:418
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767
(sofia/internal/8117 at 1.1.1.1) State REPORTING
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:92
sofia/internal/8117 at 1.1.1.1 Standard REPORTING, cause:
EXCHANGE_ROUTING_ERROR
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:767
(sofia/internal/8117 at 1.1.1.1) State REPORTING going to sleep
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:444
(sofia/internal/8117 at 1.1.1.1) State Change CS_REPORTING -> CS_DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1351 Send signal
sofia/internal/8117 at 1.1.1.1[BREAK]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_session.c:1559 Session 201
(sofia/internal/8117 at 1.1.1.1) Locked, Waiting on external entities
2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1577 Session 201
(sofia/internal/8117 at 1.1.1.1) Ended
2015-03-25 12:16:38.385162 [NOTICE] switch_core_session.c:1581 Close
Channel sofia/internal/8117 at 1.1.1.1[CS_DESTROY]
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:572
(sofia/internal/8117 at 1.1.1.1) Running State Change CS_DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582
(sofia/internal/8117 at 1.1.1.1) State DESTROY
2015-03-25 12:16:38.385162 [DEBUG] mod_sofia.c:399 sofia/internal/
8117 at 1.1.1.1 SOFIA DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:99
sofia/internal/8117 at 1.1.1.1 Standard DESTROY
2015-03-25 12:16:38.385162 [DEBUG] switch_core_state_machine.c:582
(sofia/internal/8117 at 1.1.1.1) State DESTROY going to sleep


any ideas?

The design is based on a cost for all of the following linkhttps://
freeswitch.org/confluence/display/FREESWITCH/mod_nibblebill#mod_nibblebill-Installationandconfiguration
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/4ab43726/attachment-0001.html 

From david.villasmil at gmail.com  Thu Mar 26 19:30:28 2015
From: david.villasmil at gmail.com (David Villasmil Govea)
Date: Thu, 26 Mar 2015 17:30:28 +0100
Subject: [Freeswitch-users] possible bug?
Message-ID: <CAPTi0gpfDeKSm4O8Q6SJJincAz_izaSbVkWDK=SOjcz8nA=xgg@mail.gmail.com>

Hello guys,

I had this situation in which I'm generating calls to FS from port 5061,
but on the actual INVITE it says port 5060. FS is responding to port 5060
instead of port 5061 which is the port from which it received the requests.

Is this by design?

Regards,

David

-- 
DVG

-- 
Imagination is more important than knowledge
Albert Einstein
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/ef2c254b/attachment.html 

From ben at langfeld.co.uk  Thu Mar 26 19:37:46 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 26 Mar 2015 13:37:46 -0300
Subject: [Freeswitch-users] possible bug?
In-Reply-To: <CAPTi0gpfDeKSm4O8Q6SJJincAz_izaSbVkWDK=SOjcz8nA=xgg@mail.gmail.com>
References: <CAPTi0gpfDeKSm4O8Q6SJJincAz_izaSbVkWDK=SOjcz8nA=xgg@mail.gmail.com>
Message-ID: <CAAyX+KFLTEHpDdW0vaWsfAcV9gZx9qS9ROfP5PEZ-p3ex6avow@mail.gmail.com>

You're calling FreeSWITCH from what, exactly? You left out a lot of
important detail about your scenario.

On 26 March 2015 at 13:30, David Villasmil Govea <david.villasmil at gmail.com>
wrote:

> Hello guys,
>
> I had this situation in which I'm generating calls to FS from port 5061,
> but on the actual INVITE it says port 5060. FS is responding to port 5060
> instead of port 5061 which is the port from which it received the requests.
>
> Is this by design?
>
> Regards,
>
> David
>
> --
> DVG
>
> --
> Imagination is more important than knowledge
> Albert Einstein
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/d7db2eae/attachment.html 

From krice at freeswitch.org  Thu Mar 26 19:39:20 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 26 Mar 2015 16:39:20 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
Message-ID: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>

New Post on freeswitch.org from krice387
check it out at http://ift.tt/1Izjq3R
FreeSWITCH 1.4.18 Released!
FreeSWITCH 1.4.18 has been released!

This is routine maintenance release.

Source Tarball available at http://ift.tt/1Izjssi

Debian and Yum Repos have been updated as well.

See the release notes below for a list of notable changes.

For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO .

FreeSWITCH 1.4.18 Release Notes

FreeSWITCH 1.4.18 is a routine maintenance release.

New features that were added:

FS-7201 Set ANI plan and ANI type for ftmod_libpri

FS-7209 If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type.

FS-7265 Add mongo_find_n API

FS-7269 Add error logs in mod_java

FS-7284 A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs.

FS-7285 Allow eavesdrop to bridge only one leg

FS-7164 Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly.

FS-7299 Implement cookie-file option for mod_xml_cdr

FS-7302 Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir

FS-7309 Allow removal of User-Agent header from the sip message

FS-7304 Multiple and reversed ranges for XML dialplan date and time conditions

FS-7312 Update mod_verto to proxy additional variables

FS-7323 Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.

FS-7354 Filter feature ported from mod_event_socket to mod_erlang_event

?

Improvements to the documentation:

FS-7362 Add minimal configuration for configuring FreeSWITCH from scratch

?

Improvements in build system, cross platform support, and packaging:

FS-7149 Update Windows build to use flite-2.0.0-release

FS-7346 Update mod_mongo driver to 1.1.0

FS-7122 Fixed issues building on CentOS 5 and other distributions with older autotools

FS-6520 Fix for libv8 build issue using MSVC 2013

FS-7245 Don?t rebuild core on mod_foo-clean targets

FS-7270 Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile

FS-7318 Debian rules update to handle a pre-bootstrapped orig file

FS-7149 Update freeswitch.spec for flite-2.0.0

FS-7236 Fix code before declaration in mod_conference

FS-7264 Fix signed/unsigned warnings on Windows building ws.c

FS-7294 Enable -Werror when building with clang compiler

FS-7296 Fix build error on newest gcc

FS-7314 Fix for configure error caused by a broken openssl 1.0.2 includes

FS-7322 Fix for issues building on CentOS 5 and other distributions with older autotools

FS-7340 Remove json-c dependency in favor of our own json code

FS-7350 Add ?enable-address-sanitizer configure flag to enable clang address sanitizer

FS-7355 Fix rpl_realloc symbol missing link error that can occur when using clang

?

The following bugs were fixed:

FS-7193 Fix for sofia contact being encoded which makes it impossible to call a registered user

FS-7191 Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems

FS-7205 Do not url encode unless an ?@? is in the uri

FS-7211 Fix for sofia_contact returns unable to locate registered user

FS-7208 _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp

FS-7214 Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache

FS-7217 Use upper case when you query

FS-7197 If the span has been already fully stopped and ftdm is not running, return success from the span stop function.

FS-7235 Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC

FS-7236 Added lock to prevent a race condition and segfault in mod_conference

FS-7236 Fix mutex use before init error caused by 27c8622

0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed.

FS-7256 Fix for being unable to load mod_java

FS-7252 Fix for 6-year-old regression from commit 525f1ac back in 2008

FS-7260 Fix for L16 at 16000h with Asterisk negotiation issue

FS-7236 Re-factor to fix audio problem from commit 7c63670

FS-7250 Removed the FreeSWITCH core handler for SIG_CHLD because it isn?t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted.

FS-7066 Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231

FS-7298 Fix race condition when callcenter member cancels the call

FS-7301 Fix for issue faxing to numbers with a pass through tone

FS-7192 Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia

FS-7308 Only log SLA SQL query SQL when debugging is enabled in mod_sofia

FS-7306 Fix for fs_encode in mod_spandsp sleeping too much

FS-7230 Fixed a memory leak in mod_conference

FS-7307 Fixed buffering issue when recording calls in native format

FS-7126 Fixed coredump when calling the translate application

FS-7313 Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail

FS-7339 Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time.

FS-7342 Fixed a crash regression in mod_conference caused by FS-7230

FS-7305 Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav.

FS-5570 Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.

FS-7300 Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/831e4577/attachment.html 

From ashwinrath at gmail.com  Thu Mar 26 19:39:30 2015
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Thu, 26 Mar 2015 22:09:30 +0530
Subject: [Freeswitch-users] FSComm not building in windows
Message-ID: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>

Hi

The FSComm app seems to have multiple issues while building on windows.

1) The mod_qsettings ins included in the project but not present on
filesystem
2) the ISettings and AccountManager classes are not included in the project
3) Including the above classes causes linker errors for missing QT related
methods such as qt_metacast, metaObject etc

is FScomm actively maintained or a deprecated project?

-- 
Ashwin Rath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/a74d19a2/attachment-0001.html 

From spencer at whiteskycommunications.com  Thu Mar 26 19:56:04 2015
From: spencer at whiteskycommunications.com (Spencer Thomason)
Date: Thu, 26 Mar 2015 16:56:04 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
Message-ID: <3D4EE2F7-1750-46DA-9B7E-0F945D0D25BA@whiteskycommunications.com>

Hi Ken,
Is there anyway to get commit 36e1840d6306887e72ab3c74c29a2204a7d0fcf7 in v1.4?  Without this, we?ve been unable to upgrade past 1.4.12 as T38 has been broken on several devices.

Thanks for all the hard work!
Spencer



On Mar 26, 2015, at 9:39 AM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:

New Post on freeswitch.org<http://freeswitch.org> from krice387
check it out at http://ift.tt/1Izjq3R
FreeSWITCH 1.4.18 Released!

FreeSWITCH 1.4.18 has been released!

This is routine maintenance release.

Source Tarball available at http://ift.tt/1Izjssi

Debian and Yum Repos have been updated as well.

See the release notes below for a list of notable changes.

For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO .


FreeSWITCH 1.4.18 Release Notes

FreeSWITCH 1.4.18 is a routine maintenance release.

New features that were added:

  *   FS-7201<http://ift.tt/1Izjssm> Set ANI plan and ANI type for ftmod_libpri
  *   FS-7209<http://ift.tt/1E59QHb> If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type.
  *   FS-7265<http://ift.tt/1CBB8Bj> Add mongo_find_n API
  *   FS-7269<http://ift.tt/1DZDaxd> Add error logs in mod_java
  *   FS-7284<http://ift.tt/1DLZjjG> A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs.
  *   FS-7285<http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one leg
  *   FS-7164<http://ift.tt/1DLZhbq> Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly.
  *   FS-7299<http://ift.tt/1Flzgwx> Implement cookie-file option for mod_xml_cdr
  *   FS-7302<http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir
  *   FS-7309<http://ift.tt/1FlzgwB> Allow removal of User-Agent header from the sip message
  *   FS-7304<http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML dialplan date and time conditions
  *   FS-7312<http://ift.tt/1vUSFoy> Update mod_verto to proxy additional variables
  *   FS-7323<http://ift.tt/1GJ3SJN> Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.
  *   FS-7354<http://ift.tt/1GeoZmD> Filter feature ported from mod_event_socket to mod_erlang_event



Improvements to the documentation:

  *   FS-7362<http://ift.tt/1O0xNC0> Add minimal configuration for configuring FreeSWITCH from scratch



Improvements in build system, cross platform support, and packaging:

  *   FS-7149<http://ift.tt/1Gl1nRp> Update Windows build to use flite-2.0.0-release
  *   FS-7346<http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
  *   FS-7122<http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5 and other distributions with older autotools
  *   FS-6520<http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC 2013
  *   FS-7245<http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean targets
  *   FS-7270<http://ift.tt/1CBB8Bo> Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
  *   FS-7318<http://ift.tt/1DLZjA3> Debian rules update to handle a pre-bootstrapped orig file

  *   FS-7149<http://ift.tt/1Gl1nRp> Update freeswitch.spec for flite-2.0.0
  *   FS-7236<http://ift.tt/1AbGMuF> Fix code before declaration in mod_conference
  *   FS-7264<http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on Windows building ws.c
  *   FS-7294<http://ift.tt/1Flzio8> Enable -Werror when building with clang compiler
  *   FS-7296<http://ift.tt/1DLZjA5> Fix build error on newest gcc
  *   FS-7314<http://ift.tt/1vUSEkv> Fix for configure error caused by a broken openssl 1.0.2 includes
  *   FS-7322<http://ift.tt/1vUSFER> Fix for issues building on CentOS 5 and other distributions with older autotools
  *   FS-7340<http://ift.tt/1BjWYKt> Remove json-c dependency in favor of our own json code
  *   FS-7350<http://ift.tt/18vTmJL> Add ?enable-address-sanitizer configure flag to enable clang address sanitizer
  *   FS-7355<http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link error that can occur when using clang



The following bugs were fixed:

  *   FS-7193<http://ift.tt/1Akme4d> Fix for sofia contact being encoded which makes it impossible to call a registered user
  *   FS-7191<http://ift.tt/1yw2rsQ> Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems
  *   FS-7205<http://ift.tt/1Akmccp> Do not url encode unless an ?@? is in the uri
  *   FS-7211<http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable to locate registered user
  *   FS-7208<http://ift.tt/1Akmekt> _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp
  *   FS-7214<http://ift.tt/1yw2u7Y> Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache
  *   FS-7217<http://ift.tt/1yw2u84> Use upper case when you query
  *   FS-7197<http://ift.tt/1Izjqkp> If the span has been already fully stopped and ftdm is not running, return success from the span stop function.
  *   FS-7235<http://ift.tt/1AbGLqx> Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC
  *   FS-7236<http://ift.tt/1AbGMuF> Added lock to prevent a race condition and segfault in mod_conference

  *   FS-7236<http://ift.tt/1AbGMuF> Fix mutex use before init error caused by 27c8622
  *   0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed.
  *   FS-7256<http://ift.tt/1CBB5FN> Fix for being unable to load mod_java
  *   FS-7252<http://ift.tt/1DZDaNr> Fix for 6-year-old regression from commit 525f1ac back in 2008
  *   FS-7260<http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk negotiation issue
  *   FS-7236<http://ift.tt/1AbGMuF> Re-factor to fix audio problem from commit 7c63670

  *   FS-7250<http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler for SIG_CHLD because it isn?t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted.
  *   FS-7066<http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231
  *   FS-7298<http://ift.tt/1DLZjA5> Fix race condition when callcenter member cancels the call
  *   FS-7301<http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with a pass through tone
  *   FS-7192<http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia
  *   FS-7308<http://ift.tt/1Flziok> Only log SLA SQL query SQL when debugging is enabled in mod_sofia
  *   FS-7306<http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp sleeping too much
  *   FS-7230<http://ift.tt/1FlzgwN> Fixed a memory leak in mod_conference

  *   FS-7307<http://ift.tt/1vUSEkp> Fixed buffering issue when recording calls in native format
  *   FS-7126<http://ift.tt/1GJ3R8O> Fixed coredump when calling the translate application
  *   FS-7313<http://ift.tt/1vUSFoD> Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail

  *   FS-7339<http://ift.tt/1BjWWCs> Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time.
  *   FS-7342<http://ift.tt/18vTol4> Fixed a crash regression in mod_conference caused by FS-7230
  *   FS-7305<http://ift.tt/18vTmJN> Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav.
  *   FS-5570<http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.

  *   FS-7300<http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs

_________________________________________________________________________
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://confluence.freeswitch.org
http://www.cluecon.com

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/9b67cbf0/attachment.html 

From max at nysolutions.com  Thu Mar 26 19:57:53 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Thu, 26 Mar 2015 16:57:53 +0000
Subject: [Freeswitch-users] FSComm not building in windows
In-Reply-To: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>
References: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>

Did you follow the wiki? https://wiki.freeswitch.org/wiki/FSComm

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashwin Rath
Sent: Thursday, March 26, 2015 12:40 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FSComm not building in windows

Hi
The FSComm app seems to have multiple issues while building on windows.
1) The mod_qsettings ins included in the project but not present on filesystem
2) the ISettings and AccountManager classes are not included in the project
3) Including the above classes causes linker errors for missing QT related methods such as qt_metacast, metaObject etc
is FScomm actively maintained or a deprecated project?

--
Ashwin Rath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/48103a44/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/48103a44/attachment-0001.jpg 

From brian at freeswitch.org  Thu Mar 26 20:03:40 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 Mar 2015 12:03:40 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <3D4EE2F7-1750-46DA-9B7E-0F945D0D25BA@whiteskycommunications.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<3D4EE2F7-1750-46DA-9B7E-0F945D0D25BA@whiteskycommunications.com>
Message-ID: <CAEJMVkCX8nJv6BtcdkPn1GPYyQAs=_jvcefUvzNtJ=XxkT36Yg@mail.gmail.com>

If you need that in 1.4.x then you can wait till the next release, Or you
can pull the 1.4.18 tarball and apply the patch yourself locally and
rebuild the packages.  We have published the official release process today
with the release of 1.4.18

For more information about the release process please see
https://freeswitch.org/confluence/display/FREESWITCH/Release+Process

Thanks,

On Thu, Mar 26, 2015 at 11:56 AM, Spencer Thomason <
spencer at whiteskycommunications.com> wrote:

>  Hi Ken,
> Is there anyway to get commit 36e1840d6306887e72ab3c74c29a2204a7d0fcf7 in
> v1.4?  Without this, we?ve been unable to upgrade past 1.4.12 as T38 has
> been broken on several devices.
>
>  Thanks for all the hard work!
> Spencer
>
>
>
>  On Mar 26, 2015, at 9:39 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> New Post on freeswitch.org from krice387
> check it out at http://ift.tt/1Izjq3R
> FreeSWITCH 1.4.18 Released!
>
> FreeSWITCH 1.4.18 has been released!
>
> This is routine maintenance release.
>
> Source Tarball available at http://ift.tt/1Izjssi
>
> Debian and Yum Repos have been updated as well.
>
> See the release notes below for a list of notable changes.
>
> For additional information about the FreeSWITCH release process, please
> see http://ift.tt/1E59SyO .
>
>  FreeSWITCH 1.4.18 Release Notes
>
> FreeSWITCH 1.4.18 is a routine maintenance release.
>
> New features that were added:
>
>    - FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for
>    ftmod_libpri
>    - FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted
>    correctly by libpri, fallback to calling TON/type.
>    - FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
>    - FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
>    - FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line
>    arguments to allow for using -base instead of specifying each directory
>    when using alternate configs.
>    - FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one
>    leg
>    - FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in
>    the grammar tag for mod_rayo. This is useful for MRCP engines to look up
>    their grammars directly.
>    - FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for
>    mod_xml_cdr
>    - FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for
>    path to conf_dir -k for path to log_dir -m for path to mod_dir
>    - FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header
>    from the sip message
>    - FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML
>    dialplan date and time conditions
>    - FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy additional
>    variables
>    - FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in
>    mod_http_cache using {refresh=true} parameter that can be prefixed to a URL
>    to force refresh when using http:// https:// file formats or the
>    http_get API. And added http_remove_cache API call to manually expire a
>    cached URL.
>    - FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from
>    mod_event_socket to mod_erlang_event
>
>
>
> Improvements to the documentation:
>
>    - FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for
>    configuring FreeSWITCH from scratch
>
>
>
> Improvements in build system, cross platform support, and packaging:
>
>    - FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use
>    flite-2.0.0-release
>    - FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
>    - FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5
>    and other distributions with older autotools
>    - FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC
>    2013
>    - FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean
>    targets
>    - FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for
>    libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
>    - FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a
>    pre-bootstrapped orig file
>
>
>    - FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for
>    flite-2.0.0
>    - FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in
>    mod_conference
>    - FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on
>    Windows building ws.c
>    - FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with
>    clang compiler
>    - FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
>    - FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a
>    broken openssl 1.0.2 includes
>    - FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5
>    and other distributions with older autotools
>    - FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor of
>    our own json code
>    - FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer
>    configure flag to enable clang address sanitizer
>    - FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link
>    error that can occur when using clang
>
>
>
> The following bugs were fixed:
>
>    - FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded
>    which makes it impossible to call a registered user
>    - FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection string
>    to remove unnecessary option that may cause a failure on some systems
>    - FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is
>    in the uri
>    - FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable
>    to locate registered user
>    - FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident
>    will make it be an empty string which is the same you were doing on your
>    local builds in mod_spandsp
>    - FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command
>    argument bounds checking for flush and delete in mod_memcache
>    - FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
>    - FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully
>    stopped and ftdm is not running, return success from the span stop
>    function.
>    - FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting
>    recorded files in append mode if appended data is shorter than
>    RECORD_MIN_SEC
>    - FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race
>    condition and segfault in mod_conference
>
>
>    - FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error
>    caused by 27c8622
>    - 0dc48df Fix for a bug from original implementation, cannot send call
>    state about state destroy, this is an internal state and the session is
>    already destroyed.
>    - FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load
>    mod_java
>    - FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from
>    commit 525f1ac back in 2008
>    - FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk
>    negotiation issue
>    - FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from
>    commit 7c63670
>
>
>    - FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler
>    for SIG_CHLD because it isn?t necessary anymore and it causes dependent
>    libraries that tried to start a child process to hang waiting on a signal
>    that FreeSWITCH core intercepted.
>    - FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load
>    averages on older kernels with related bugs FS-7253 and FS-7231
>    - FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter
>    member cancels the call
>    - FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with
>    a pass through tone
>    - FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs
>    responding to an auth challenge in mod_sofia
>    - FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when
>    debugging is enabled in mod_sofia
>    - FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp
>    sleeping too much
>    - FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in
>    mod_conference
>
>
>    - FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when recording
>    calls in native format
>    - FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the
>    translate application
>    - FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing
>    invalid params to the vm_fsdb_msg_email api in mod_voicemail
>
>
>    - FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql
>    statements for basic_calls and detailed_calls to happen after the creation
>    of the tables so the creation works and won?t have to be run a second time.
>    - FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in
>    mod_conference caused by FS-7230
>    - FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of
>    FS startup and shutdown faster, like in the case of tone2wav.
>    - FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to
>    group api command. When the ?multi? parameter is present, the group command
>    will return a list of group members delimited by :_: which allows for
>    multiply-registered endpoints to participate in a group.
>
>
>    - FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in
>    SPEECH-COMPLETE event and validate load input grammar URLs
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/72094d73/attachment.html 

From mike at jerris.com  Thu Mar 26 20:05:15 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 26 Mar 2015 13:05:15 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <3D4EE2F7-1750-46DA-9B7E-0F945D0D25BA@whiteskycommunications.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<3D4EE2F7-1750-46DA-9B7E-0F945D0D25BA@whiteskycommunications.com>
Message-ID: <059B4DEF-F631-4502-8D74-C1BE24D88647@jerris.com>

As noted in the jira:

https://freeswitch.org/jira/browse/FS-6954 <https://freeswitch.org/jira/browse/FS-6954>

That fix will be in 1.4.19.  If you would like to handle your own builds, that fix is safe to cherry-pick on top of the 1.4.18 tag.

> On Mar 26, 2015, at 12:56 PM, Spencer Thomason <spencer at whiteskycommunications.com> wrote:
> 
> Hi Ken,
> Is there anyway to get commit 36e1840d6306887e72ab3c74c29a2204a7d0fcf7 in v1.4?  Without this, we?ve been unable to upgrade past 1.4.12 as T38 has been broken on several devices.
> 
> Thanks for all the hard work!
> Spencer
> 
>  
> 
> On Mar 26, 2015, at 9:39 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> 
>> New Post on freeswitch.org <http://freeswitch.org/> from krice387
>> check it out at http://ift.tt/1Izjq3R <http://ift.tt/1Izjq3R>
>> FreeSWITCH 1.4.18 Released!
>> FreeSWITCH 1.4.18 has been released!
>> 
>> This is routine maintenance release.
>> 
>> Source Tarball available at http://ift.tt/1Izjssi <http://ift.tt/1Izjssi>
>> Debian and Yum Repos have been updated as well.
>> 
>> See the release notes below for a list of notable changes.
>> 
>> For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO <http://ift.tt/1E59SyO> .
>> 
>> 
>> FreeSWITCH 1.4.18 Release Notes
>> 
>> FreeSWITCH 1.4.18 is a routine maintenance release.
>> 
>> New features that were added:
>> 
>> FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for ftmod_libpri
>> FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type.
>> FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
>> FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
>> FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs.
>> FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one leg
>> FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly.
>> FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for mod_xml_cdr
>> FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir
>> FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header from the sip message
>> FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML dialplan date and time conditions
>> FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy additional variables
>> FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.
>> FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from mod_event_socket to mod_erlang_event
>>  
>> Improvements to the documentation:
>> 
>> FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for configuring FreeSWITCH from scratch
>>  
>> Improvements in build system, cross platform support, and packaging:
>> 
>> FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use flite-2.0.0-release
>> FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
>> FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5 and other distributions with older autotools
>> FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC 2013
>> FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean targets
>> FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
>> FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a pre-bootstrapped orig file
>> FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for flite-2.0.0
>> FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in mod_conference
>> FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on Windows building ws.c
>> FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with clang compiler
>> FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
>> FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a broken openssl 1.0.2 includes
>> FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5 and other distributions with older autotools
>> FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor of our own json code
>> FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer configure flag to enable clang address sanitizer
>> FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link error that can occur when using clang
>>  
>> The following bugs were fixed:
>> 
>> FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded which makes it impossible to call a registered user
>> FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems
>> FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is in the uri
>> FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable to locate registered user
>> FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp
>> FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache
>> FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
>> FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully stopped and ftdm is not running, return success from the span stop function.
>> FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC
>> FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race condition and segfault in mod_conference
>> FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error caused by 27c8622
>> 0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed.
>> FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load mod_java
>> FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from commit 525f1ac back in 2008
>> FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk negotiation issue
>> FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from commit 7c63670
>> FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler for SIG_CHLD because it isn?t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted.
>> FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231
>> FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter member cancels the call
>> FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with a pass through tone
>> FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia
>> FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when debugging is enabled in mod_sofia
>> FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp sleeping too much
>> FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in mod_conference
>> FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when recording calls in native format
>> FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the translate application
>> FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail
>> FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time.
>> FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in mod_conference caused by FS-7230
>> FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav.
>> FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.
>> FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/6e1f6686/attachment-0001.html 

From andretodd at verizon.net  Thu Mar 26 20:07:22 2015
From: andretodd at verizon.net (Andre DeMattia)
Date: Thu, 26 Mar 2015 13:07:22 -0400
Subject: [Freeswitch-users] Call originate suggestion needed
Message-ID: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>

HI, I need some suggestions on how to use Limit on an outbound gateway(s)
and count the calls that make it past limit.

 

Currently I have my limit fire on_originate and it works but it cause a Leg
B cdr. (not a big deal) and I need to count the calls but when I count them
on on_Post Originate only the successful ringing calls are counted, 503's
etc are not counted. I can't count the calls on originate because that
doesn't remove the exceeded limit calls

 

So what I want to do is call my dialplan and before the calls start check
limit and move on to the next bridge. Any call that makes it to the provider
I need to count to keep track (cps, ports etc)

 

Any suggestions? I don't want to do loopback since i'm trying to get the
calls out of the switch as fast as possible. 


thanks all

Andre

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/382dff0e/attachment.html 

From spa at syntec.co.uk  Thu Mar 26 20:05:01 2015
From: spa at syntec.co.uk (Sebastian Auriol)
Date: Thu, 26 Mar 2015 17:05:01 -0000
Subject: [Freeswitch-users] How do I bridge to a different profile AND
	gateway?
Message-ID: <!&!AAAAAAAAAAAuAAAAAAAAAL60wriLM9cRsTVojW0AAAABAFEdQVQs6tMRsLEAoMxarIMAAAABk2cAABAAAABH/e7kLfGlQaOxe9CSRZZ0AQAAAAA=@syntec.co.uk>

Hi list,
 
It's my first post: be gentle! ;-)
 
I've read the wiki and I've searched the mailing list but I can't find out
how to do what I want to do and that is:
 
Bridge an incoming call to a different profile AND specify the gateway?  I
can easily see how to do one or the other but not both.  Why would I want to
do this?  The incoming call is on the internal profile.  The internal
profile uses UDP and ACLs as it is all connected to a single switch.  But
the call from internal needs to be bridged to an external gateway using TLS
- on a different network interface.  So it should be on a different profile.
However, I don't want to specify the IP address in the dialplan because the
destination is another gateway, so it should be a gateway.
 
Any help appreciated - I'm fairly new to FS.

Kind regards,

SebA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/9c2662c1/attachment.html 

From mdalepiane at gmail.com  Thu Mar 26 20:08:34 2015
From: mdalepiane at gmail.com (Mateus Dalepiane)
Date: Thu, 26 Mar 2015 14:08:34 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
Message-ID: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>

We have the following scenario: The session is established between WebRTC
and FreeSWITCH using Websockets.

Once the session is established, if the websocket connection drops the
media continues to flow util FreeSWITCH tries to send a re-INVITE to the
client. At this point it realizes that the connection was closed and hangs
up the call.

Now, if the websocket connection drops and is re-established, would it be
possible to inform FreeSWITCH that the new connection should be used for
the previously established session?

If the WebRTC client sends an INVITE message with the old session
parameters, FreeSWITCH will be able to understand that it belongs to the
old session?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/e7755d8a/attachment.html 

From tfred31 at yahoo.com  Thu Mar 26 20:55:00 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Thu, 26 Mar 2015 10:55:00 -0700
Subject: [Freeswitch-users] transfer_fallback_extension
Message-ID: <1427392500.92705.YahooMailBasic@web160205.mail.bf1.yahoo.com>

On a number of websites and in my conf/directory/default.xml file I see:   
     <variable name = "transfer_fallback_extension" value = "operator"/>

And in my    conf/dialplan/default.xml   file I see     <extension name = "operator">    where the operator 'extension' is defined as a single extension number.

But I don't want the failed transfer attempt to go to a single pre-defined extension 
I want the failed transfer attempt to go back to the extension which initially launched the transfer attempt

How/where would I modify the variable   transfer_fallback_extension   to get this to work as needed?

Or should this be handled in a different manner somewhere else within FreeSWITCH?
If so, where/how?

Thanks




    


From victor.chukalovskiy at gmail.com  Thu Mar 26 21:07:02 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Thu, 26 Mar 2015 14:07:02 -0400
Subject: [Freeswitch-users] How do I bridge to a different profile AND
 gateway?
In-Reply-To: <!&!AAAAAAAAAAAuAAAAAAAAAL60wriLM9cRsTVojW0AAAABAFEdQVQs6tMRsLEAoMxarIMAAAABk2cAABAAAABH/e7kLfGlQaOxe9CSRZZ0AQAAAAA=@syntec.co.uk>
References: <!&!AAAAAAAAAAAuAAAAAAAAAL60wriLM9cRsTVojW0AAAABAFEdQVQs6tMRsLEAoMxarIMAAAABk2cAABAAAABH/e7kLfGlQaOxe9CSRZZ0AQAAAAA=@syntec.co.uk>
Message-ID: <55144AC6.6010909@gmail.com>

Suppose your gateway is named "gw_1" and it's defined withing "gateways" 
section of your "external" profile.
As I understand it, you can't have "gw_1" also defined in profile 
"internal" at the same time. So, gateway name is unique across all profiles.

So, when you bridge:
<action application="bridge" data="sofia/gateway/gw_1/5555555555"/>

You specify "gw_1" gateway and it implies the profile is "external"



On 15-03-26 01:05 PM, Sebastian Auriol wrote:
> Hi list,
> It's my first post: be gentle! ;-)
> I've read the wiki and I've searched the mailing list but I can't find 
> out how to do what I want to do and that is:
> Bridge an incoming call to a different profile AND specify 
> the gateway?  I can easily see how to do one or the other but not 
> both.  Why would I want to do this?  The incoming call is on 
> the internal profile. The internal profile uses UDP and ACLs as it is 
> all connected to a single switch.  But the call from internal needs to 
> be bridged to an external gateway using TLS - on a different network 
> interface.  So it should be on a different profile.  However, I don't 
> want to specify the IP address in the dialplan because the destination 
> is another gateway, so it should be a gateway.
> Any help appreciated - I'm fairly new to FS.
>
> Kind regards,
>
> SebA
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/c227be43/attachment-0001.html 

From brian at freeswitch.org  Thu Mar 26 21:15:08 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 Mar 2015 13:15:08 -0500
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
Message-ID: <CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>

Have you taken a look at Verto?

On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
wrote:

> We have the following scenario: The session is established between WebRTC
> and FreeSWITCH using Websockets.
>
> Once the session is established, if the websocket connection drops the
> media continues to flow util FreeSWITCH tries to send a re-INVITE to the
> client. At this point it realizes that the connection was closed and hangs
> up the call.
>
> Now, if the websocket connection drops and is re-established, would it be
> possible to inform FreeSWITCH that the new connection should be used for
> the previously established session?
>
> If the WebRTC client sends an INVITE message with the old session
> parameters, FreeSWITCH will be able to understand that it belongs to the
> old session?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/2cb7019f/attachment.html 

From spa at syntec.co.uk  Thu Mar 26 21:28:09 2015
From: spa at syntec.co.uk (SebA)
Date: Thu, 26 Mar 2015 18:28:09 -0000
Subject: [Freeswitch-users] How do I bridge to a different profile AND
	gateway?
In-Reply-To: <55144AC6.6010909@gmail.com>
References: <!&!AAAAAAAAAAAuAAAAAAAAAL60wriLM9cRsTVojW0AAAABAFEdQVQs6tMRsLEAoMxarIMAAAABk2cAABAAAABH/e7kLfGlQaOxe9CSRZZ0AQAAAAA=@syntec.co.uk>
	<55144AC6.6010909@gmail.com>
Message-ID: <!&!AAAAAAAAAAAuAAAAAAAAAL60wriLM9cRsTVojW0AAAABAFEdQVQs6tMRsLEAoMxarIMAAAABk2cAABAAAAAhZf4QkqTlRpnk3/3FifnLAQAAAAA=@syntec.co.uk>

Ah!  I thought I tried that initially, but maybe I had something else wrong
at the time.  I can confirm that that works.  Many thanks!

Kind regards, 

SebA  


 


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Victor
Chukalovskiy
Sent: 26 March 2015 18:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How do I bridge to a different profile AND
gateway?


Suppose your gateway is named "gw_1" and it's defined withing "gateways"
section of your "external" profile.
As I understand it, you can't have "gw_1" also defined in profile "internal"
at the same time. So, gateway name is unique across all profiles.

So, when you bridge:
<action application="bridge" data="sofia/gateway/gw_1/5555555555"/>

You specify "gw_1" gateway and it implies the profile is "external"




On 15-03-26 01:05 PM, Sebastian Auriol wrote:


Hi list,
 
It's my first post: be gentle! ;-)
 
I've read the wiki and I've searched the mailing list but I can't find out
how to do what I want to do and that is:
 
Bridge an incoming call to a different profile AND specify the gateway?  I
can easily see how to do one or the other but not both.  Why would I want to
do this?  The incoming call is on the internal profile.  The internal
profile uses UDP and ACLs as it is all connected to a single switch.  But
the call from internal needs to be bridged to an external gateway using TLS
- on a different network interface.  So it should be on a different profile.
However, I don't want to specify the IP address in the dialplan because the
destination is another gateway, so it should be a gateway.
 
Any help appreciated - I'm fairly new to FS.

Kind regards,

SebA

 

_________________________________________________________________________

Professional FreeSWITCH Consulting Services: 

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/171a11ec/attachment.html 

From victor.chukalovskiy at gmail.com  Thu Mar 26 21:42:26 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Thu, 26 Mar 2015 14:42:26 -0400
Subject: [Freeswitch-users] Call originate suggestion needed
In-Reply-To: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
References: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
Message-ID: <55145312.2010100@gmail.com>

I use mod_lcr for routing calls to different providers based on their 
rate-sheets. It allows to set limit on per-provider basis in terms of 
ports.

cps not really, but believe it's more of a carriers concern than 
yours... unless you are pumping some crazy values

Maybe this makes no sense in your use scenario, so just an idea

On 15-03-26 01:07 PM, Andre DeMattia wrote:
>
> HI, I need some suggestions on how to use Limit on an outbound 
> gateway(s) and count the calls that make it past limit.
>
> Currently I have my limit fire on_originate and it works but it cause 
> a Leg B cdr. (not a big deal) and I need to count the calls but when I 
> count them on on_Post Originate only the successful ringing calls are 
> counted, 503?s etc are not counted. I can?t count the calls on 
> originate because that doesn?t remove the exceeded limit calls
>
> So what I want to do is call my dialplan and before the calls start 
> check limit and move on to the next bridge. Any call that makes it to 
> the provider I need to count to keep track (cps, ports etc)
>
> **
>
> Any suggestions? I don't want to do loopback since i'm trying to get 
> the calls out of the switch as fast as possible.
>
>
> thanks all
>
> Andre
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/b43b3b70/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Thu Mar 26 21:48:11 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Thu, 26 Mar 2015 18:48:11 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
Message-ID: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>

All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML requests but I do see for dialplan so the call does succeed. I don?t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn?t seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app?s cdr and not the bridge apps. I need the bridge app?s cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/28326195/attachment.html 

From mishehu at freeswitch.org  Thu Mar 26 22:15:41 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 26 Mar 2015 14:15:41 -0500
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <55145ADD.1060002@freeswitch.org>

Did you try just having mod_xml_cdr log out to files on the local system 
to see if you are experiencing the same issues or not?

-- 
Yossi Neiman

On 03/26/2015 01:48 PM, Sharath Kumar wrote:

> All,
>
> I use mod_cdr_xml with xml_curl and receive messages on the backend 
> PHP for call control logic. I am a little mystified by the XML CDR 
> requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t 
> always see the XML requests but I do see for dialplan so the call does 
> succeed. I don?t understand why the CDRs XML requests are 
> intermittent? Also, I changed the config to log the b-leg cdr as well, 
> it didn?t seem to make any impact.
>
> So for each leg do I expect a CDR ? Also,if I do a redirect and then a 
> bridge I only see the redirect app?s cdr and not the bridge apps. I 
> need the bridge app?s cdr since It contains QOS metrics for that call.
>
> A little help would be appreciated.
>
> Thank you
>
> Sharath
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/43eb7dab/attachment.html 

From steveayre at gmail.com  Thu Mar 26 22:39:49 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 Mar 2015 19:39:49 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>

Are you logging b-leg CDRs as well as a-leg?

On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com> wrote:

>      All,
>
>
>
> I use mod_cdr_xml with xml_curl and receive messages on the backend PHP
> for call control logic. I am a little mystified by the XML CDR requests. I
> see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML
> requests but I do see for dialplan so the call does succeed. I don?t
> understand why the CDRs XML requests are intermittent? Also, I changed the
> config to log the b-leg cdr as well, it didn?t seem to make any impact.
>
> So for each leg do I expect a CDR ? Also,if I do a redirect and then a
> bridge I only see the redirect app?s cdr and not the bridge apps. I need
> the bridge app?s cdr since It contains QOS metrics for that call.
>
>
>
> A little help would be appreciated.
>
>
>
> Thank you
>
> Sharath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/eb55d088/attachment.html 

From Sharath.Kumar at meZocliq.com  Thu Mar 26 22:52:38 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Thu, 26 Mar 2015 19:52:38 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <55145ADD.1060002@freeswitch.org>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<55145ADD.1060002@freeswitch.org>
Message-ID: <CY1PR0501MB1098D35E357F8F09EAA0D79784080@CY1PR0501MB1098.namprd05.prod.outlook.com>

Okay I tried doing that. It did not create log file though. I assume you enable it by adding  "log-dir" value="/var/log/freeswitch" in the xml_cdr.conf file. Or is there any other way to achieve this ?

Thank you Yossi

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of I put the Who? in Mishehu
Sent: Thursday, March 26, 2015 3:16 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Did you try just having mod_xml_cdr log out to files on the local system to see if you are experiencing the same issues or not?



--

Yossi Neiman



On 03/26/2015 01:48 PM, Sharath Kumar wrote:
All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don't always see the XML requests but I do see for dialplan so the call does succeed. I don't understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn't seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app's cdr and not the bridge apps. I need the bridge app's cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath




_________________________________________________________________________

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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/198489e5/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Thu Mar 26 22:59:40 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Thu, 26 Mar 2015 19:59:40 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
Message-ID: <CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>

I am logging both legs. I believe the default behavior is to generate cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by commenting the entry. Neither seemed to matter. If I do see a CDR I only see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2 cdr request correct ?

Thank you Steven
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Thursday, March 26, 2015 3:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Are you logging b-leg CDRs as well as a-leg?

On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML requests but I do see for dialplan so the call does succeed. I don?t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn?t seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app?s cdr and not the bridge apps. I need the bridge app?s cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath

_________________________________________________________________________
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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/44817239/attachment.html 

From anthony.minessale at gmail.com  Thu Mar 26 23:06:43 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 Mar 2015 15:06:43 -0500
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>

Only if you actually have 2 legs, a failed call where the number is wrong
etc will only have 1

On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  I am logging both legs. I believe the default behavior is to generate
> cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by
> commenting the entry. Neither seemed to matter. If I do see a CDR I only
> see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2
> cdr request correct ?
>
>
>
> Thank you Steven
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, March 26, 2015 3:40 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> Are you logging b-leg CDRs as well as a-leg?
>
>
>
> On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>      All,
>
>
>
> I use mod_cdr_xml with xml_curl and receive messages on the backend PHP
> for call control logic. I am a little mystified by the XML CDR requests. I
> see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML
> requests but I do see for dialplan so the call does succeed. I don?t
> understand why the CDRs XML requests are intermittent? Also, I changed the
> config to log the b-leg cdr as well, it didn?t seem to make any impact.
>
> So for each leg do I expect a CDR ? Also,if I do a redirect and then a
> bridge I only see the redirect app?s cdr and not the bridge apps. I need
> the bridge app?s cdr since It contains QOS metrics for that call.
>
>
>
> A little help would be appreciated.
>
>
>
> Thank you
>
> Sharath
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/3487be39/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Thu Mar 26 23:18:04 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Thu, 26 Mar 2015 20:18:04 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
Message-ID: <CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>

Of course. I am only talking about successful calls and with the voice path present.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, March 26, 2015 4:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Only if you actually have 2 legs, a failed call where the number is wrong etc will only have 1

On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
I am logging both legs. I believe the default behavior is to generate cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by commenting the entry. Neither seemed to matter. If I do see a CDR I only see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2 cdr request correct ?

Thank you Steven
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 Steven Ayre
Sent: Thursday, March 26, 2015 3:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Are you logging b-leg CDRs as well as a-leg?

On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML requests but I do see for dialplan so the call does succeed. I don?t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn?t seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app?s cdr and not the bridge apps. I need the bridge app?s cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath

_________________________________________________________________________
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://confluence.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://confluence.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       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/3c6cb6ca/attachment.html 

From mishehu at freeswitch.org  Thu Mar 26 23:42:35 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 26 Mar 2015 15:42:35 -0500
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
	<CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <55146F3B.3090605@freeswitch.org>

Ok, so if you set the log dir path, you are not seeing any actual files 
in that path?  There should be 1 xml file for each call channel that 
made it all the way to the CS_REPORTING state.  If you do not see that, 
then I'd have to guess that there is something else in your 
configuration that is impacting this.  If I am correct, then it would 
help if you would provide the config and the debug logs from the FS 
console or fs_cli.

-- 
Yossi Neiman

On 03/26/2015 03:18 PM, Sharath Kumar wrote:

> Of course. I am only talking about successful calls and with the voice 
> path present.
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Anthony Minessale
> *Sent:* Thursday, March 26, 2015 4:07 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
> Only if you actually have 2 legs, a failed call where the number is 
> wrong etc will only have 1
>
> On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar 
> <Sharath.Kumar at mezocliq.com <mailto:Sharath.Kumar at mezocliq.com>> wrote:
>
>     I am logging both legs. I believe the default behavior is to
>     generate cdrs for b-leg as well. I tried by adding ?log-b-leg? to
>     true and also by commenting the entry. Neither seemed to matter.
>     If I do see a CDR I only see 1 request. I thought by enabling both
>     ?a? and ?b? legs I should see 2 cdr request correct ?
>
>     Thank you Steven
>
>     *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 *Steven Ayre
>     *Sent:* Thursday, March 26, 2015 3:40 PM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>     Are you logging b-leg CDRs as well as a-leg?
>
>     On 26 March 2015 at 18:48, Sharath Kumar
>     <Sharath.Kumar at mezocliq.com <mailto:Sharath.Kumar at mezocliq.com>>
>     wrote:
>
>         All,
>
>         I use mod_cdr_xml with xml_curl and receive messages on the
>         backend PHP for call control logic. I am a little mystified by
>         the XML CDR requests. I see in the logs CS_HANGUP to
>         CS_REPORTING but I don?t always see the XML requests but I do
>         see for dialplan so the call does succeed. I don?t understand
>         why the CDRs XML requests are intermittent? Also, I changed
>         the config to log the b-leg cdr as well, it didn?t seem to
>         make any impact.
>
>         So for each leg do I expect a CDR ? Also,if I do a redirect
>         and then a bridge I only see the redirect app?s cdr and not
>         the bridge apps. I need the bridge app?s cdr since It contains
>         QOS metrics for that call.
>
>         A little help would be appreciated.
>
>         Thank you
>
>         Sharath
>
>
>         _________________________________________________________________________
>         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://confluence.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://confluence.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 ? @anthmfs ? @FreeSWITCH ?
>
> ? http://freeswitch.org/ ? http://cluecon.com/ ? 
> http://twitter.com/FreeSWITCH
>
> ? irc.freenode.net <http://irc.freenode.net> #freeswitch ? 
> _http://freeswitch.org/g+_
>
> ClueCon Weekly Development Call
>
> ? sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org> ? +19193869900
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/5e380c56/attachment-0001.html 

From Sharath.Kumar at meZocliq.com  Thu Mar 26 23:52:21 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Thu, 26 Mar 2015 20:52:21 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <55146F3B.3090605@freeswitch.org>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
	<CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<55146F3B.3090605@freeswitch.org>
Message-ID: <CY1PR0501MB109845898A997BD2C9F913AF84080@CY1PR0501MB1098.namprd05.prod.outlook.com>

Thanks Yossi. I will dig a little deeper and let you know.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of I put the Who? in Mishehu
Sent: Thursday, March 26, 2015 4:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Ok, so if you set the log dir path, you are not seeing any actual files in that path?  There should be 1 xml file for each call channel that made it all the way to the CS_REPORTING state.  If you do not see that, then I'd have to guess that there is something else in your configuration that is impacting this.  If I am correct, then it would help if you would provide the config and the debug logs from the FS console or fs_cli.



--

Yossi Neiman



On 03/26/2015 03:18 PM, Sharath Kumar wrote:
Of course. I am only talking about successful calls and with the voice path present.

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 Anthony Minessale
Sent: Thursday, March 26, 2015 4:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Only if you actually have 2 legs, a failed call where the number is wrong etc will only have 1

On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
I am logging both legs. I believe the default behavior is to generate cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by commenting the entry. Neither seemed to matter. If I do see a CDR I only see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2 cdr request correct ?

Thank you Steven
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 Steven Ayre
Sent: Thursday, March 26, 2015 3:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Are you logging b-leg CDRs as well as a-leg?

On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML requests but I do see for dialplan so the call does succeed. I don?t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn?t seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app?s cdr and not the bridge apps. I need the bridge app?s cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath

_________________________________________________________________________
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://confluence.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://confluence.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       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900





_________________________________________________________________________

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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/f2c77224/attachment-0001.html 

From ssinyagin at gmail.com  Fri Mar 27 04:40:33 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 27 Mar 2015 02:40:33 +0100
Subject: [Freeswitch-users] Call originate suggestion needed
In-Reply-To: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
References: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
Message-ID: <CACOYK=gnwthtnP-ra=79smuWaPZLKM0Te9Yj-7zZp8_3Hj3Uxg@mail.gmail.com>

collect call setup and hangup events via ESL, and keep count of them
in a database. Then the dialplan would check that database to make
routing decisions.



On Thu, Mar 26, 2015 at 6:07 PM, Andre DeMattia <andretodd at verizon.net> wrote:
> HI, I need some suggestions on how to use Limit on an outbound gateway(s)
> and count the calls that make it past limit.
>
>
>
> Currently I have my limit fire on_originate and it works but it cause a Leg
> B cdr. (not a big deal) and I need to count the calls but when I count them
> on on_Post Originate only the successful ringing calls are counted, 503?s
> etc are not counted. I can?t count the calls on originate because that
> doesn?t remove the exceeded limit calls
>
>
>
> So what I want to do is call my dialplan and before the calls start check
> limit and move on to the next bridge. Any call that makes it to the provider
> I need to count to keep track (cps, ports etc)
>
>
>
> Any suggestions? I don't want to do loopback since i'm trying to get the
> calls out of the switch as fast as possible.
>
>
> thanks all
>
> Andre
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From italorossib at gmail.com  Fri Mar 27 05:24:57 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 26 Mar 2015 23:24:57 -0300
Subject: [Freeswitch-users] Call originate suggestion needed
In-Reply-To: <CACOYK=gnwthtnP-ra=79smuWaPZLKM0Te9Yj-7zZp8_3Hj3Uxg@mail.gmail.com>
References: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
	<CACOYK=gnwthtnP-ra=79smuWaPZLKM0Te9Yj-7zZp8_3Hj3Uxg@mail.gmail.com>
Message-ID: <CAJR7f2HqSP-7bF2zQEpy_HNK40s_WNoy-tjNQBCBVsadvv2wag@mail.gmail.com>

What about
https://freeswitch.org/confluence/display/FREESWITCH/Limit#Limit-limit_execute
?

On Thu, Mar 26, 2015 at 10:40 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> collect call setup and hangup events via ESL, and keep count of them
> in a database. Then the dialplan would check that database to make
> routing decisions.
>
>
>
> On Thu, Mar 26, 2015 at 6:07 PM, Andre DeMattia <andretodd at verizon.net>
> wrote:
> > HI, I need some suggestions on how to use Limit on an outbound gateway(s)
> > and count the calls that make it past limit.
> >
> >
> >
> > Currently I have my limit fire on_originate and it works but it cause a
> Leg
> > B cdr. (not a big deal) and I need to count the calls but when I count
> them
> > on on_Post Originate only the successful ringing calls are counted, 503?s
> > etc are not counted. I can?t count the calls on originate because that
> > doesn?t remove the exceeded limit calls
> >
> >
> >
> > So what I want to do is call my dialplan and before the calls start check
> > limit and move on to the next bridge. Any call that makes it to the
> provider
> > I need to count to keep track (cps, ports etc)
> >
> >
> >
> > Any suggestions? I don't want to do loopback since i'm trying to get the
> > calls out of the switch as fast as possible.
> >
> >
> > thanks all
> >
> > Andre
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/73c41b57/attachment.html 

From brian at freeswitch.org  Fri Mar 27 05:28:05 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 Mar 2015 21:28:05 -0500
Subject: [Freeswitch-users] Call originate suggestion needed
In-Reply-To: <CAJR7f2HqSP-7bF2zQEpy_HNK40s_WNoy-tjNQBCBVsadvv2wag@mail.gmail.com>
References: <BAY403-EAS17EC112B6B554BE1489221BF080@phx.gbl>
	<CACOYK=gnwthtnP-ra=79smuWaPZLKM0Te9Yj-7zZp8_3Hj3Uxg@mail.gmail.com>
	<CAJR7f2HqSP-7bF2zQEpy_HNK40s_WNoy-tjNQBCBVsadvv2wag@mail.gmail.com>
Message-ID: <CAEJMVkCNDBe42N4mNvRtS2FZsAqC+VPxGYcayBSkBggdqGbo+Q@mail.gmail.com>

limit_execute is probably the best option in this case.  Per ?talo's email.

On Thu, Mar 26, 2015 at 9:24 PM, ?talo Rossi <italorossib at gmail.com> wrote:

> What about
> https://freeswitch.org/confluence/display/FREESWITCH/Limit#Limit-limit_execute
> ?
>
> On Thu, Mar 26, 2015 at 10:40 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> collect call setup and hangup events via ESL, and keep count of them
>> in a database. Then the dialplan would check that database to make
>> routing decisions.
>>
>>
>>
>> On Thu, Mar 26, 2015 at 6:07 PM, Andre DeMattia <andretodd at verizon.net>
>> wrote:
>> > HI, I need some suggestions on how to use Limit on an outbound
>> gateway(s)
>> > and count the calls that make it past limit.
>> >
>> >
>> >
>> > Currently I have my limit fire on_originate and it works but it cause a
>> Leg
>> > B cdr. (not a big deal) and I need to count the calls but when I count
>> them
>> > on on_Post Originate only the successful ringing calls are counted,
>> 503?s
>> > etc are not counted. I can?t count the calls on originate because that
>> > doesn?t remove the exceeded limit calls
>> >
>> >
>> >
>> > So what I want to do is call my dialplan and before the calls start
>> check
>> > limit and move on to the next bridge. Any call that makes it to the
>> provider
>> > I need to count to keep track (cps, ports etc)
>> >
>> >
>> >
>> > Any suggestions? I don't want to do loopback since i'm trying to get the
>> > calls out of the switch as fast as possible.
>> >
>> >
>> > thanks all
>> >
>> > Andre
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150326/7250a2fc/attachment.html 

From ashwinrath at gmail.com  Fri Mar 27 09:31:11 2015
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Fri, 27 Mar 2015 12:01:11 +0530
Subject: [Freeswitch-users] FSComm not building in windows
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>
References: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>
Message-ID: <CAA_Nwu474WTRDjAAi-ndtvaPzX-svA-czmKASjpxs-73FtQ+WA@mail.gmail.com>

Yes i did

But even then i get errors like missing files or missing function
definitions.

On Thu, Mar 26, 2015 at 10:27 PM, Moishe Grunstein <max at nysolutions.com>
wrote:

>  Did you follow the wiki? https://wiki.freeswitch.org/wiki/FSComm
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashwin Rath
> *Sent:* Thursday, March 26, 2015 12:40 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] FSComm not building in windows
>
>
>
> Hi
>
> The FSComm app seems to have multiple issues while building on windows.
>
> 1) The mod_qsettings ins included in the project but not present on
> filesystem
>
> 2) the ISettings and AccountManager classes are not included in the project
>
> 3) Including the above classes causes linker errors for missing QT related
> methods such as qt_metacast, metaObject etc
>
> is FScomm actively maintained or a deprecated project?
>
>
> --
>
> Ashwin Rath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ashwin Kumar Rath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/7c3c0a82/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/7c3c0a82/attachment-0001.jpg 

From raphael.lechner at gmail.com  Fri Mar 27 12:21:01 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Fri, 27 Mar 2015 10:21:01 +0100
Subject: [Freeswitch-users] transfer_fallback_extension
In-Reply-To: <1427392500.92705.YahooMailBasic@web160205.mail.bf1.yahoo.com>
References: <1427392500.92705.YahooMailBasic@web160205.mail.bf1.yahoo.com>
Message-ID: <CFF69A91-8CB3-4542-880C-0507559DEC30@gmail.com>

Hi,

I have created the following dial plan for that.

If the call is not answered after 30 seconds then the call is going back to the initial caller and on the display is showing (NO_ANSWER)>>orig_caller_id_name>>orig_caller_id_number.

<extension name="Local_Extension">
  <condition field="destination_number" expression="^([0-9][0-9])$">
    <action application="export" data="dialed_extension=$1" />
    <action application="bind_meta_app" data="1 b s execute_extension::dx XML features" />
    <action application="bind_meta_app"
      data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav" />
    <action application="bind_meta_app" data="3 b s execute_extension::cf XML features" />
    <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features" />
    <action application="set" data="transfer_ringback=$${hold_music}" />
    <action application="set" data="call_timeout=30" />
    <action application="set" data="hangup_after_bridge=true" />
    <action application="set" data="continue_on_fail=NO_ANSWER" />
    <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}" />
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}" />
    <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}" />
    <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}" />
    <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}" />
    <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}" />
    <action application="set"
      data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}" />
    <action application="set" data="effective_callee_id_number=${dialed_extension}" />
    <action application="log" data="INFO effective_callee_id_number=${dialed_extension}" />
    <action application="log"
      data="INFO effective_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}" />
    <action application="export" data="alert_info=http://localhost/;info=internal" />
    <action application="log" data="INFO alert_info=${alert_info}" />

    <condition field="${last_bridge_hangup_cause}" expression="^NO_ANSWER$">
      <action application="log"
        data="INFO NO_ANSWER -- ${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})" />
      <action application="export"
        data="origination_caller_id_name=${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})" />
      <anti-action application="set"
        data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}" />
    </condition>

    <action application="bridge"
      data="{effective_callee_id_name=${effective_callee_id_name},effective_callee_id_number=${effective_callee_id_number},origination_caller_id_name=${origination_caller_id_name}}user/${dialed_extension}@${domain_name}" />
    <action application="transfer" data="${last_sent_callee_id_number}" />
  </condition>
</extension>

Raphael

> On 26 Mar 2015, at 18:55, T Fred Farmington <tfred31 at yahoo.com> wrote:
> 
> On a number of websites and in my conf/directory/default.xml file I see:   
>     <variable name = "transfer_fallback_extension" value = "operator"/>
> 
> And in my    conf/dialplan/default.xml   file I see     <extension name = "operator">    where the operator 'extension' is defined as a single extension number.
> 
> But I don't want the failed transfer attempt to go to a single pre-defined extension 
> I want the failed transfer attempt to go back to the extension which initially launched the transfer attempt
> 
> How/where would I modify the variable   transfer_fallback_extension   to get this to work as needed?
> 
> Or should this be handled in a different manner somewhere else within FreeSWITCH?
> If so, where/how?
> 
> Thanks
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jayesh1017 at gmail.com  Fri Mar 27 12:43:05 2015
From: jayesh1017 at gmail.com (Jayesh Nambiar)
Date: Fri, 27 Mar 2015 09:43:05 +0000
Subject: [Freeswitch-users] Swap on att_xfer application
Message-ID: <CANAYoztEfEHt1ySZfUw0cX_GDt8SoA_zDXdMVnYegnCNOGbTCw@mail.gmail.com>

Hello,
I was exploring the usage of att_xfer and it works as expected. I was just
wondering that is there a way when the last party answers, I can hold that
party and get back to initial call placing the last party on hold using
some DTMF action??
I see a DTMF action like # to end the last call and get back to the initial
call, but what I am looking at is hold the last call and get back to
initial call before I actually transfer the call. Is there anything similar
available which can be used to swap between calls using the att_xfer
application.

Thanks,

--- Jayesh
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/608eb0da/attachment.html 

From steveayre at gmail.com  Fri Mar 27 13:38:31 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 27 Mar 2015 10:38:31 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <55146F3B.3090605@freeswitch.org>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
	<CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<55146F3B.3090605@freeswitch.org>
Message-ID: <CAFiqYuk8kmyatFy-SzoRtG6xiasqasicaTaCTwtfL4Bcjz=rsA@mail.gmail.com>

If you have submitting to a web server enabled you'll want to
set log-http-and-disk to true (the default is false).

On 26 March 2015 at 20:42, I put the Who? in Mishehu <mishehu at freeswitch.org
> wrote:

>  Ok, so if you set the log dir path, you are not seeing any actual files
> in that path?  There should be 1 xml file for each call channel that made
> it all the way to the CS_REPORTING state.  If you do not see that, then I'd
> have to guess that there is something else in your configuration that is
> impacting this.  If I am correct, then it would help if you would provide
> the config and the debug logs from the FS console or fs_cli.
>
> --
> Yossi Neiman
>
> On 03/26/2015 03:18 PM, Sharath Kumar wrote:
>
>   Of course. I am only talking about successful calls and with the voice
> path present.
>
>
>
> *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 *Anthony
> Minessale
> *Sent:* Thursday, March 26, 2015 4:07 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> Only if you actually have 2 legs, a failed call where the number is wrong
> etc will only have 1
>
>
>
> On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>  I am logging both legs. I believe the default behavior is to generate
> cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by
> commenting the entry. Neither seemed to matter. If I do see a CDR I only
> see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2
> cdr request correct ?
>
>
>
> Thank you Steven
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, March 26, 2015 3:40 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> Are you logging b-leg CDRs as well as a-leg?
>
>
>
> On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>      All,
>
>
>
> I use mod_cdr_xml with xml_curl and receive messages on the backend PHP
> for call control logic. I am a little mystified by the XML CDR requests. I
> see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML
> requests but I do see for dialplan so the call does succeed. I don?t
> understand why the CDRs XML requests are intermittent? Also, I changed the
> config to log the b-leg cdr as well, it didn?t seem to make any impact.
>
> So for each leg do I expect a CDR ? Also,if I do a redirect and then a
> bridge I only see the redirect app?s cdr and not the bridge apps. I need
> the bridge app?s cdr since It contains QOS metrics for that call.
>
>
>
> A little help would be appreciated.
>
>
>
> Thank you
>
> Sharath
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
>
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
>
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
>
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/8b25c428/attachment-0001.html 

From denis at ringme.ru  Fri Mar 27 14:24:22 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Fri, 27 Mar 2015 14:24:22 +0300
Subject: [Freeswitch-users] bind_meta_app - bind to * AND #
Message-ID: <55153DE6.60006@ringme.ru>

Hello.

I want use bind_meta_app, how i can bind on *2 and #1? If i set 
bind_meta_key=# - bind on 2 BEFORE this set - bound to #2.
I don`t want use bind_digit_action - it's so strange and 
incomprehensible, especially bind to leg B

My last try:
         <action application="bind_digit_action" 
data="att_xfer,*2,exec:execute_extension,att_xfer XML features"/>
         <action application="bind_meta_app" data="2 b b 
execute_extension::att_xfer XML features"/>
         <action application="set" data="bind_meta_key=#"/>
         <action application="bind_digit_action" 
data="blind_xfer,#1,exec:execute_extension,dx2 XML features"/>
         <action application="bind_meta_app" data="1 b b 
execute_extension::dx XML features"/>

Who have working configs for transfer (blind+att) with A+B legs?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/0f9d5fa4/attachment.html 

From denis at ringme.ru  Fri Mar 27 14:35:08 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Fri, 27 Mar 2015 14:35:08 +0300
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
Message-ID: <5515406C.9030208@ringme.ru>

Where is 7385 fix?
https://freeswitch.org/jira/browse/FS-7385

On 26.03.2015 19:39, Ken Rice wrote:
> New Post on freeswitch.org from krice387
> check it out at http://ift.tt/1Izjq3R
> FreeSWITCH 1.4.18 Released!


From nasida at live.ru  Fri Mar 27 15:41:28 2015
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 27 Mar 2015 15:41:28 +0300
Subject: [Freeswitch-users] unexpected segfault with latest debian and
	libmyodbc.so
Message-ID: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>

Hi guys,

I just got unexpected segfault I try to understand if anybody had similar problems.

Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at 500 ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in libmyodbc.so[7fd3a8340000+3c000]


# lsb_release -a
Distributor ID:    Debian
Description:    Debian GNU/Linux 7.8 (wheezy)
Release:    7.8
Codename:    wheezy

freeswitch at internal> version 
FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)

# apt-cache show libmyodbc
Package: libmyodbc
Source: myodbc
Version: 5.1.10-2+deb7u1

Please advice,
Thanks.

 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/6dfd3337/attachment.html 

From italorossib at gmail.com  Fri Mar 27 15:43:33 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 27 Mar 2015 09:43:33 -0300
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <5515406C.9030208@ringme.ru>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<5515406C.9030208@ringme.ru>
Message-ID: <CAJR7f2HwNNEHrTFBCwhhajaEyKTHrN1CnCoMu3eNjbNA5qqEgA@mail.gmail.com>

I think it's not included in this version, please someone correct me if i'm
wrong

This is the fix:
https://freeswitch.org/fisheye/changelog/freeswitch?cs=ed0a434b95efc54dbc01017fd6ff33dab1582371

On Fri, Mar 27, 2015 at 8:35 AM, ????? <denis at ringme.ru> wrote:

> Where is 7385 fix?
> https://freeswitch.org/jira/browse/FS-7385
>
> On 26.03.2015 19:39, Ken Rice wrote:
> > New Post on freeswitch.org from krice387
> > check it out at http://ift.tt/1Izjq3R
> > FreeSWITCH 1.4.18 Released!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/af71706c/attachment.html 

From brian at freeswitch.org  Fri Mar 27 15:45:14 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 07:45:14 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <5515406C.9030208@ringme.ru>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<5515406C.9030208@ringme.ru>
Message-ID: <CAEJMVkCKCA+Yu=bks=Bbn=5p9LVMOnGSbBZOs87kk0c4w9sdHA@mail.gmail.com>

The release process for 1.4.18 started on Thursday March 12th.  The release
process takes time and testing.

On Fri, Mar 27, 2015 at 6:35 AM, ????? <denis at ringme.ru> wrote:

> Where is 7385 fix?
> https://freeswitch.org/jira/browse/FS-7385
>
> On 26.03.2015 19:39, Ken Rice wrote:
> > New Post on freeswitch.org from krice387
> > check it out at http://ift.tt/1Izjq3R
> > FreeSWITCH 1.4.18 Released!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/43324e93/attachment.html 

From switcherfree at gmail.com  Fri Mar 27 09:12:02 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Thu, 26 Mar 2015 23:12:02 -0700
Subject: [Freeswitch-users] Missing modules and configs in fresh install
Message-ID: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>

Hello,
I'm trying to create a new installation of freeswitch. I'm running
CentOS 6.6 and picked the easy path to install pre-built binaries
using yum. Looking at /usr/lib64/freeswitch after the install, I
noticed that mod_conference.so wasn't there. Also noticed that no
config files were installed in /etc/freeswitch. Is this expected?

I downloaded configs from the latest stash repository and I was then
able to start freeswitch with vanilla config. I see several startup
messages about missing mods. Which modules should I expect to see as
part of the pre-built binaries? How/where does one download additional
mods? Here is what I have in the /usr/lib64/freeswitch :

mod_cdr_csv.so
mod_commands.so
mod_console.so
mod_dialplan_directory.so
mod_dialplan_xml.so
mod_dptools.so
mod_event_socket.so
mod_logfile.so
mod_loopback.so
mod_native_file.so
mod_sndfile.so
mod_sofia.so
mod_spandsp.so
mod_syslog.so
mod_tone_stream.so
mod_xml_rpc.so


Any help is appreciated.
Thanks,
Andy


From italorossib at gmail.com  Fri Mar 27 15:48:10 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 27 Mar 2015 09:48:10 -0300
Subject: [Freeswitch-users] unexpected segfault with latest debian and
	libmyodbc.so
In-Reply-To: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>
References: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>
Message-ID: <CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>

I have seen a lot of threading issues with mysql + odbc, make sure you're
using unixodbc >= 2.3.

If you're using an older version you can set Threading = 2 in your
/etc/odbcinst.ini as a workaround, but this is *not* recommended for
production/high volume, upgrade as soon as possible.

On Fri, Mar 27, 2015 at 9:41 AM, Yuriy Nasida <nasida at live.ru> wrote:

> Hi guys,
>
> I just got unexpected segfault I try to understand if anybody had similar
> problems.
>
> *Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at 500
> ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in
> libmyodbc.so[7fd3a8340000+3c000]*
>
>
> # lsb_release -a
> Distributor ID:    Debian
> Description:    Debian GNU/Linux 7.8 (wheezy)
> Release:    7.8
> Codename:    wheezy
>
> freeswitch at internal> version
> FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git
> 507a0f2 2014-12-29 18:59:51Z 64bit)
>
> # apt-cache show libmyodbc
> Package: libmyodbc
> Source: myodbc
> Version: 5.1.10-2+deb7u1
>
> Please advice,
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/b38866f9/attachment-0001.html 

From aqsyounas at gmail.com  Fri Mar 27 15:57:34 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 27 Mar 2015 17:57:34 +0500
Subject: [Freeswitch-users] freeswitch memory leakage.
Message-ID: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>

Hi, users

We are using 6 freeswitch instances, 5 freeswitch instances for playing
streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
taking more than 3gb of momory and even though calls are not more than 25.

Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
Everytime we have to restart the freeswitch to release the captured memory.

But is believed, freeswitch must release the momory when there are no
calls. But freeswitch still keeps captured memory.

I have attactted the top command result along with this email.
Willing to perform any test if it helps tackle the problem.

Thanks for your help.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/b25b9925/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: top-result.png
Type: image/png
Size: 319968 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/b25b9925/attachment-0001.png 

From brian at freeswitch.org  Fri Mar 27 16:06:18 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 08:06:18 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
Message-ID: <CAEJMVkCNDFBmqrh04Z7TGnKQ01jQnPJ5aDOiETEBMZYCqexWfg@mail.gmail.com>

You sure you have to restart?  Does the process eat up all the ram and
segfault due to resource starvation? Or stop working at all?

https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-CollectionInformationWithValgrind(Linux/Unix)

It may or may not be a leak based on your description, I do know mod_vlc
does use some memory up, but it may just be memory pool swelling.

Also what version of FreeSWITCH are you running exactly?

On Fri, Mar 27, 2015 at 7:57 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, users
>
> We are using 6 freeswitch instances, 5 freeswitch instances for playing
> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
> taking more than 3gb of momory and even though calls are not more than 25.
>
> Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
> Everytime we have to restart the freeswitch to release the captured memory.
>
> But is believed, freeswitch must release the momory when there are no
> calls. But freeswitch still keeps captured memory.
>
> I have attactted the top command result along with this email.
> Willing to perform any test if it helps tackle the problem.
>
> Thanks for your help.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/2ded07fa/attachment.html 

From mishehu at freeswitch.org  Fri Mar 27 16:19:12 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 08:19:12 -0500
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
Message-ID: <551558D0.7090803@freeswitch.org>

Depending on what repo you got those packages from, they could be 
seriously old packages.  Send back the output of:

rpm -qa |grep -i freeswitch

Chances are you only installed the base package.  you can also do `yum 
search freeswitch` to see if other packages are offered. After you 
provide the output from the above command I can comment on the age of 
the packages you installed.

-- 
Yossi Neiman

On 03/27/2015 01:12 AM, Free Switcher wrote:

> Hello,
> I'm trying to create a new installation of freeswitch. I'm running
> CentOS 6.6 and picked the easy path to install pre-built binaries
> using yum. Looking at /usr/lib64/freeswitch after the install, I
> noticed that mod_conference.so wasn't there. Also noticed that no
> config files were installed in /etc/freeswitch. Is this expected?
>
> I downloaded configs from the latest stash repository and I was then
> able to start freeswitch with vanilla config. I see several startup
> messages about missing mods. Which modules should I expect to see as
> part of the pre-built binaries? How/where does one download additional
> mods? Here is what I have in the /usr/lib64/freeswitch :
>
> mod_cdr_csv.so
> mod_commands.so
> mod_console.so
> mod_dialplan_directory.so
> mod_dialplan_xml.so
> mod_dptools.so
> mod_event_socket.so
> mod_logfile.so
> mod_loopback.so
> mod_native_file.so
> mod_sndfile.so
> mod_sofia.so
> mod_spandsp.so
> mod_syslog.so
> mod_tone_stream.so
> mod_xml_rpc.so
>
>
> Any help is appreciated.
> Thanks,
> Andy
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Mar 27 16:22:29 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 08:22:29 -0500
Subject: [Freeswitch-users] FSComm not building in windows
In-Reply-To: <CAA_Nwu474WTRDjAAi-ndtvaPzX-svA-czmKASjpxs-73FtQ+WA@mail.gmail.com>
References: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>	<BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>
	<CAA_Nwu474WTRDjAAi-ndtvaPzX-svA-czmKASjpxs-73FtQ+WA@mail.gmail.com>
Message-ID: <55155995.8070606@freeswitch.org>

If you think you've found a bug please file a jira for it.  Though it's 
possible you are simply missing a dependency on your system.  But either 
way, we'll not be able to advise without seeing what those errors 
actually are.

-- 
Yossi Neiman

On 03/27/2015 01:31 AM, Ashwin Rath wrote:
> Yes i did
>
> But even then i get errors like missing files or missing function 
> definitions.
>
> On Thu, Mar 26, 2015 at 10:27 PM, Moishe Grunstein 
> <max at nysolutions.com <mailto:max at nysolutions.com>> wrote:
>
>     Did you follow the wiki? https://wiki.freeswitch.org/wiki/FSComm
>
>     Thanks,
>
>     Moishe Grunstein
>
>     Tornado Computer Systems, Inc.
>
>     212.400.7650 <tel:212.400.7650> 888.IPPBX.US <http://888.IPPBX.US>
>     *Service Request Email: support at nysolutions.com
>     <mailto:support at nysolutions.com> *
>
>     cid:image001.jpg at 01C72F94.9EE45D60 <http://www.nysolutions.com/>
>
>     Computer Networking * Managed Services * IP Video Surveillance *
>     Network Assessments * Web Solutions * Voice over IP * Disaster
>     Recovery * Network Security * Site Surveys * CMS
>
>     *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 *Ashwin Rath
>     *Sent:* Thursday, March 26, 2015 12:40 PM
>     *To:* FreeSWITCH Users Help
>     *Subject:* [Freeswitch-users] FSComm not building in windows
>
>     Hi
>
>     The FSComm app seems to have multiple issues while building on
>     windows.
>
>     1) The mod_qsettings ins included in the project but not present
>     on filesystem
>
>     2) the ISettings and AccountManager classes are not included in
>     the project
>
>     3) Including the above classes causes linker errors for missing QT
>     related methods such as qt_metacast, metaObject etc
>
>     is FScomm actively maintained or a deprecated project?
>
>
>     -- 
>
>     Ashwin Rath
>
>
>     _________________________________________________________________________
>     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://confluence.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
>
>
>
>
> -- 
> Ashwin Kumar Rath
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/e60feecf/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/e60feecf/attachment-0001.jpe 

From brian at freeswitch.org  Fri Mar 27 16:23:04 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 08:23:04 -0500
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <551558D0.7090803@freeswitch.org>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
Message-ID: <CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>

If you downloaded and built the source its all going to be in
/usr/local/freeswitch unless you configured it with the same configure args
as the previous package install.

On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu <
mishehu at freeswitch.org> wrote:

> Depending on what repo you got those packages from, they could be
> seriously old packages.  Send back the output of:
>
> rpm -qa |grep -i freeswitch
>
> Chances are you only installed the base package.  you can also do `yum
> search freeswitch` to see if other packages are offered. After you
> provide the output from the above command I can comment on the age of
> the packages you installed.
>
> --
> Yossi Neiman
>
> On 03/27/2015 01:12 AM, Free Switcher wrote:
>
> > Hello,
> > I'm trying to create a new installation of freeswitch. I'm running
> > CentOS 6.6 and picked the easy path to install pre-built binaries
> > using yum. Looking at /usr/lib64/freeswitch after the install, I
> > noticed that mod_conference.so wasn't there. Also noticed that no
> > config files were installed in /etc/freeswitch. Is this expected?
> >
> > I downloaded configs from the latest stash repository and I was then
> > able to start freeswitch with vanilla config. I see several startup
> > messages about missing mods. Which modules should I expect to see as
> > part of the pre-built binaries? How/where does one download additional
> > mods? Here is what I have in the /usr/lib64/freeswitch :
> >
> > mod_cdr_csv.so
> > mod_commands.so
> > mod_console.so
> > mod_dialplan_directory.so
> > mod_dialplan_xml.so
> > mod_dptools.so
> > mod_event_socket.so
> > mod_logfile.so
> > mod_loopback.so
> > mod_native_file.so
> > mod_sndfile.so
> > mod_sofia.so
> > mod_spandsp.so
> > mod_syslog.so
> > mod_tone_stream.so
> > mod_xml_rpc.so
> >
> >
> > Any help is appreciated.
> > Thanks,
> > Andy
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/19e2273a/attachment.html 

From brian at freeswitch.org  Fri Mar 27 16:23:44 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 08:23:44 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <CAJR7f2HwNNEHrTFBCwhhajaEyKTHrN1CnCoMu3eNjbNA5qqEgA@mail.gmail.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<5515406C.9030208@ringme.ru>
	<CAJR7f2HwNNEHrTFBCwhhajaEyKTHrN1CnCoMu3eNjbNA5qqEgA@mail.gmail.com>
Message-ID: <CAEJMVkD5EOg9KhmENNLhtSuJ_FBSepYB_s3vvwvfzmrxuiSLCg@mail.gmail.com>

Yes, its not in 1.4.18.

On Fri, Mar 27, 2015 at 7:43 AM, ?talo Rossi <italorossib at gmail.com> wrote:

> I think it's not included in this version, please someone correct me if
> i'm wrong
>
> This is the fix:
>
> https://freeswitch.org/fisheye/changelog/freeswitch?cs=ed0a434b95efc54dbc01017fd6ff33dab1582371
>
> On Fri, Mar 27, 2015 at 8:35 AM, ????? <denis at ringme.ru> wrote:
>
>> Where is 7385 fix?
>> https://freeswitch.org/jira/browse/FS-7385
>>
>> On 26.03.2015 19:39, Ken Rice wrote:
>> > New Post on freeswitch.org from krice387
>> > check it out at http://ift.tt/1Izjq3R
>> > FreeSWITCH 1.4.18 Released!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/2ccebb5e/attachment.html 

From aqsyounas at gmail.com  Fri Mar 27 16:31:17 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 27 Mar 2015 18:31:17 +0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAEJMVkCNDFBmqrh04Z7TGnKQ01jQnPJ5aDOiETEBMZYCqexWfg@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<CAEJMVkCNDFBmqrh04Z7TGnKQ01jQnPJ5aDOiETEBMZYCqexWfg@mail.gmail.com>
Message-ID: <CAGVOjVW0PiaqpGy6ATY3q0WVsHHZTWtH9rhe4jUgavv7-wT71A@mail.gmail.com>

Thanks Brian for your quick reply.

We start getting high memory usage on our server, then we run top command
to see  that is happening. Usually freeswitch is there eating most of our
system ram, then we restart the affected freeswitch to release the memory.

We restart freeswitch to prevent our server for crashing due to memory
starvation.
We are using freeswitch version.

FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git
4909cdb 2015-02-24 20:58:26Z 64bit)


On 27 March 2015 at 18:06, Brian West <brian at freeswitch.org> wrote:

> You sure you have to restart?  Does the process eat up all the ram and
> segfault due to resource starvation? Or stop working at all?
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-CollectionInformationWithValgrind(Linux/Unix)
>
> It may or may not be a leak based on your description, I do know mod_vlc
> does use some memory up, but it may just be memory pool swelling.
>
> Also what version of FreeSWITCH are you running exactly?
>
> On Fri, Mar 27, 2015 at 7:57 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi, users
>>
>> We are using 6 freeswitch instances, 5 freeswitch instances for playing
>> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
>> taking more than 3gb of momory and even though calls are not more than 25.
>>
>> Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
>> Everytime we have to restart the freeswitch to release the captured memory.
>>
>> But is believed, freeswitch must release the momory when there are no
>> calls. But freeswitch still keeps captured memory.
>>
>> I have attactted the top command result along with this email.
>> Willing to perform any test if it helps tackle the problem.
>>
>> Thanks for your help.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/2f84f0a0/attachment-0001.html 

From mishehu at freeswitch.org  Fri Mar 27 16:32:59 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 08:32:59 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
Message-ID: <55155C0B.2070101@freeswitch.org>

Are you using linux?  If so, which memory value are you looking at (VSZ, 
RSS...) ?

If you don't use mod_vlc, do you still see FreeSWITCH consuming this 
much RAM?  Usually FreeSWITCH builds up a memory pool and won't release 
that back to the system until after shutdown, but I don't normally see 
that go above about 1.2GB of RAM in any of the systems I work on (though 
they do not use mod_vlc).

-- 
Yossi Neiman


On 03/27/2015 07:57 AM, Aqs Younas wrote:
> Hi, users
>
> We are using 6 freeswitch instances, 5 freeswitch instances for 
> playing streams with mod_vlc. After 3 to 4 for days,  we see these (5) 
> freeswitch taking more than 3gb of momory and even though calls are 
> not more than 25.
>
> Usually, some calls stay for more than 2 to 3 hours on some 
> freeswitch.  Everytime we have to restart the freeswitch to release 
> the captured memory.
>
> But is believed, freeswitch must release the momory when there are no 
> calls. But freeswitch still keeps captured memory.
>
> I have attactted the top command result along with this email.
> Willing to perform any test if it helps tackle the problem.
>
> Thanks for your help.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/62d72302/attachment.html 

From mishehu at freeswitch.org  Fri Mar 27 16:38:11 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 08:38:11 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVW0PiaqpGy6ATY3q0WVsHHZTWtH9rhe4jUgavv7-wT71A@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>	<CAEJMVkCNDFBmqrh04Z7TGnKQ01jQnPJ5aDOiETEBMZYCqexWfg@mail.gmail.com>
	<CAGVOjVW0PiaqpGy6ATY3q0WVsHHZTWtH9rhe4jUgavv7-wT71A@mail.gmail.com>
Message-ID: <55155D43.6070806@freeswitch.org>

Do you *actually* experience memory starvation on your system, or are 
you pre-emptively restarting freeswitch?

-- 
Yossi Neiman


On 03/27/2015 08:31 AM, Aqs Younas wrote:
> Thanks Brian for your quick reply.
>
> We start getting high memory usage on our server, then we run top 
> command to see  that is happening. Usually freeswitch is there eating 
> most of our system ram, then we restart the affected freeswitch to 
> release the memory.
>
> We restart freeswitch to prevent our server for crashing due to memory 
> starvation.
> We are using freeswitch version.
>
> FreeSWITCH Version 1.5.15b+git~20150224T205826Z~4909cdb7fb~64bit (git 
> 4909cdb 2015-02-24 20:58:26Z 64bit)
>
>
> On 27 March 2015 at 18:06, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     You sure you have to restart?  Does the process eat up all the ram
>     and segfault due to resource starvation? Or stop working at all?
>
>     https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-CollectionInformationWithValgrind(Linux/Unix)
>     <https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-CollectionInformationWithValgrind%28Linux/Unix%29>
>
>     It may or may not be a leak based on your description, I do know
>     mod_vlc does use some memory up, but it may just be memory pool
>     swelling.
>
>     Also what version of FreeSWITCH are you running exactly?
>
>     On Fri, Mar 27, 2015 at 7:57 AM, Aqs Younas <aqsyounas at gmail.com
>     <mailto:aqsyounas at gmail.com>> wrote:
>
>         Hi, users
>
>         We are using 6 freeswitch instances, 5 freeswitch instances
>         for playing streams with mod_vlc. After 3 to 4 for days,  we
>         see these (5) freeswitch taking more than 3gb of momory and
>         even though calls are not more than 25.
>
>         Usually, some calls stay for more than 2 to 3 hours on some
>         freeswitch.  Everytime we have to restart the freeswitch to
>         release the captured memory.
>
>         But is believed, freeswitch must release the momory when there
>         are no calls. But freeswitch still keeps captured memory.
>
>         I have attactted the top command result along with this email.
>         Willing to perform any test if it helps tackle the problem.
>
>         Thanks for your help.
>
>
>
>         _________________________________________________________________________
>         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://confluence.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
>
>
>
>
>     -- 
>
>     */Brian West/*
>     brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
>     */Twitter: @FreeSWITCH , @briankwest/*
>     http://www.freeswitchbook.com
>     http://www.freeswitchcookbook.com
>
>     ClueCon 2015 Call for Speakers
>     <https://www.cluecon.com/call-for-speakers/> | Register
>     <https://freeswitch.com/cart.php?gid=1> TODAY!
>
>     *T:*+19184209001 <tel:%2B19184209001> | *F:*+19184209002
>     <tel:%2B19184209002> | *M:*+1918424WEST (9378)
>     *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>     _________________________________________________________________________
>     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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/9c825b17/attachment.html 

From brian at freeswitch.org  Fri Mar 27 16:39:19 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 08:39:19 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <55155C0B.2070101@freeswitch.org>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<55155C0B.2070101@freeswitch.org>
Message-ID: <CAEJMVkCTT4Gmg6E0O_kQxX9P82+7vmusgbB8ebSbVKO82x-YNg@mail.gmail.com>

If your concern is only that it uses a lot of ram, but still functions for
your needs, then you're probably needlessly worrying about the ram usage.
If it does use up all of the ram in the system and crash as a result then
yes thats a problem.  I can tell you without a doubt that VLC does use a
fair amount of ram.

On Fri, Mar 27, 2015 at 8:32 AM, I put the Who? in Mishehu <
mishehu at freeswitch.org> wrote:

>  Are you using linux?  If so, which memory value are you looking at (VSZ,
> RSS...) ?
>
> If you don't use mod_vlc, do you still see FreeSWITCH consuming this much
> RAM?  Usually FreeSWITCH builds up a memory pool and won't release that
> back to the system until after shutdown, but I don't normally see that go
> above about 1.2GB of RAM in any of the systems I work on (though they do
> not use mod_vlc).
>
> --
> Yossi Neiman
>
>
>
> On 03/27/2015 07:57 AM, Aqs Younas wrote:
>
>    Hi, users
>
>  We are using 6 freeswitch instances, 5 freeswitch instances for playing
> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
> taking more than 3gb of momory and even though calls are not more than 25.
>
>  Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
> Everytime we have to restart the freeswitch to release the captured memory.
>
>  But is believed, freeswitch must release the momory when there are no
> calls. But freeswitch still keeps captured memory.
>
> I have attactted the top command result along with this email.
>  Willing to perform any test if it helps tackle the problem.
>
>  Thanks for your help.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/fc56f0fa/attachment-0001.html 

From aqsyounas at gmail.com  Fri Mar 27 16:51:24 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 27 Mar 2015 18:51:24 +0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <55155C0B.2070101@freeswitch.org>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<55155C0B.2070101@freeswitch.org>
Message-ID: <CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>

Yes, we are using linux and looking at RSS value.

Our dialplan is mostly using mod_vlc for running streams.Just a call comes
in and a radio stream is played with mod_vlc. I am not sure either mod_vlc
or something else in freeswitch is eating up ram.

I just told our dialplan is mostly based on mod_vlc a portion on mod_curl.
I have attached the top command results in above post, you can see
freeswitch consuming more ram than any process.

One more thing, 6th freeswitch doesn't capture ram more than 1 gb, that one
is not using mod_vlc or mod_curl. On other 5 freeswitch we are facing the
issue.

6th one freeswitch dialplan just consist of.

<extension name="garbage">
      <condition>
        <action application="answer"/>
        <action application="playback" data="/opt/garbage/Generic_VM.wav"/>
        <action application="hangup"/>
      </condition>
</extension>


Thanks for your reply.

On 27 March 2015 at 18:32, I put the Who? in Mishehu <mishehu at freeswitch.org
> wrote:

>  Are you using linux?  If so, which memory value are you looking at (VSZ,
> RSS...) ?
>
> If you don't use mod_vlc, do you still see FreeSWITCH consuming this much
> RAM?  Usually FreeSWITCH builds up a memory pool and won't release that
> back to the system until after shutdown, but I don't normally see that go
> above about 1.2GB of RAM in any of the systems I work on (though they do
> not use mod_vlc).
>
> --
> Yossi Neiman
>
>
>
> On 03/27/2015 07:57 AM, Aqs Younas wrote:
>
>    Hi, users
>
>  We are using 6 freeswitch instances, 5 freeswitch instances for playing
> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
> taking more than 3gb of momory and even though calls are not more than 25.
>
>  Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
> Everytime we have to restart the freeswitch to release the captured memory.
>
>  But is believed, freeswitch must release the momory when there are no
> calls. But freeswitch still keeps captured memory.
>
> I have attactted the top command result along with this email.
>  Willing to perform any test if it helps tackle the problem.
>
>  Thanks for your help.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/7963146e/attachment.html 

From krice at freeswitch.org  Fri Mar 27 17:01:43 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 27 Mar 2015 14:01:43 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <551562c785ff3_ab5ce1d3181612c@resque-worker-ip-10-153-136-71.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/096fd1d1/attachment.html 

From aqsyounas at gmail.com  Fri Mar 27 17:06:04 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 27 Mar 2015 19:06:04 +0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<55155C0B.2070101@freeswitch.org>
	<CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>
Message-ID: <CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>

Hi, Brian.

Its ok If vlc takes some ram, but freeswitch must release the ram when
there are no calls.
Is there any command that can make freeswitch release the ram without
restarting the freeswitch.

Like this linux command which makes linux release the cached ram.

free && sync && echo 3 > /proc/sys/vm/drop_caches && free1

Thanks


On 27 March 2015 at 18:51, Aqs Younas <aqsyounas at gmail.com> wrote:

> Yes, we are using linux and looking at RSS value.
>
> Our dialplan is mostly using mod_vlc for running streams.Just a call comes
> in and a radio stream is played with mod_vlc. I am not sure either mod_vlc
> or something else in freeswitch is eating up ram.
>
> I just told our dialplan is mostly based on mod_vlc a portion on mod_curl.
> I have attached the top command results in above post, you can see
> freeswitch consuming more ram than any process.
>
> One more thing, 6th freeswitch doesn't capture ram more than 1 gb, that
> one is not using mod_vlc or mod_curl. On other 5 freeswitch we are facing
> the issue.
>
> 6th one freeswitch dialplan just consist of.
>
> <extension name="garbage">
>       <condition>
>         <action application="answer"/>
>         <action application="playback" data="/opt/garbage/Generic_VM.wav"/>
>         <action application="hangup"/>
>       </condition>
> </extension>
>
>
> Thanks for your reply.
>
> On 27 March 2015 at 18:32, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>>  Are you using linux?  If so, which memory value are you looking at
>> (VSZ, RSS...) ?
>>
>> If you don't use mod_vlc, do you still see FreeSWITCH consuming this much
>> RAM?  Usually FreeSWITCH builds up a memory pool and won't release that
>> back to the system until after shutdown, but I don't normally see that go
>> above about 1.2GB of RAM in any of the systems I work on (though they do
>> not use mod_vlc).
>>
>> --
>> Yossi Neiman
>>
>>
>>
>> On 03/27/2015 07:57 AM, Aqs Younas wrote:
>>
>>    Hi, users
>>
>>  We are using 6 freeswitch instances, 5 freeswitch instances for playing
>> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
>> taking more than 3gb of momory and even though calls are not more than 25.
>>
>>  Usually, some calls stay for more than 2 to 3 hours on some freeswitch.
>> Everytime we have to restart the freeswitch to release the captured memory.
>>
>>  But is believed, freeswitch must release the momory when there are no
>> calls. But freeswitch still keeps captured memory.
>>
>> I have attactted the top command result along with this email.
>>  Willing to perform any test if it helps tackle the problem.
>>
>>  Thanks for your help.
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/d8f0d38a/attachment.html 

From fdelawarde at wirelessmundi.com  Fri Mar 27 17:40:59 2015
From: fdelawarde at wirelessmundi.com (=?windows-1252?Q?Fran=E7ois?=)
Date: Fri, 27 Mar 2015 15:40:59 +0100
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>	<55155C0B.2070101@freeswitch.org>	<CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>
	<CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>
Message-ID: <55156BFB.1030206@wirelessmundi.com>

About the "drop_caches" thing, you should *not* do that unless you 
really understand what you are doing!

It's a common misconception that it might increase performance or "free" 
memory. In reality you would probably see a performance drop and free 
memory that is already available for reuse... It's only useful if you 
are doing kernel benchmarking or debugging.

Fran?ois.


On 03/27/2015 03:06 PM, Aqs Younas wrote:
> Hi, Brian.
>
> Its ok If vlc takes some ram, but freeswitch must release the ram when 
> there are no calls.
> Is there any command that can make freeswitch release the ram without 
> restarting the freeswitch.
>
> Like this linux command which makes linux release the cached ram.
>
> free && sync && echo 3 > /proc/sys/vm/drop_caches && free1
>
> Thanks
>
>
> On 27 March 2015 at 18:51, Aqs Younas <aqsyounas at gmail.com 
> <mailto:aqsyounas at gmail.com>> wrote:
>
>     Yes, we are using linux and looking at RSS value.
>
>     Our dialplan is mostly using mod_vlc for running streams.Just a
>     call comes in and a radio stream is played with mod_vlc. I am not
>     sure either mod_vlc or something else in freeswitch is eating up ram.
>
>     I just told our dialplan is mostly based on mod_vlc a portion on
>     mod_curl. I have attached the top command results in above post,
>     you can see freeswitch consuming more ram than any process.
>
>     One more thing, 6th freeswitch doesn't capture ram more than 1 gb,
>     that one is not using mod_vlc or mod_curl. On other 5 freeswitch
>     we are facing the issue.
>
>     6th one freeswitch dialplan just consist of.
>
>     <extension name="garbage">
>           <condition>
>             <action application="answer"/>
>             <action application="playback"
>     data="/opt/garbage/Generic_VM.wav"/>
>             <action application="hangup"/>
>           </condition>
>     </extension>
>
>
>     Thanks for your reply.
>
>     On 27 March 2015 at 18:32, I put the Who? in Mishehu
>     <mishehu at freeswitch.org <mailto:mishehu at freeswitch.org>> wrote:
>
>         Are you using linux?  If so, which memory value are you
>         looking at (VSZ, RSS...) ?
>
>         If you don't use mod_vlc, do you still see FreeSWITCH
>         consuming this much RAM?  Usually FreeSWITCH builds up a
>         memory pool and won't release that back to the system until
>         after shutdown, but I don't normally see that go above about
>         1.2GB of RAM in any of the systems I work on (though they do
>         not use mod_vlc).
>
>         -- 
>         Yossi Neiman
>
>
>         On 03/27/2015 07:57 AM, Aqs Younas wrote:
>>         Hi, users
>>
>>         We are using 6 freeswitch instances, 5 freeswitch instances
>>         for playing streams with mod_vlc. After 3 to 4 for days,  we
>>         see these (5) freeswitch taking more than 3gb of momory and
>>         even though calls are not more than 25.
>>
>>         Usually, some calls stay for more than 2 to 3 hours on some
>>         freeswitch.  Everytime we have to restart the freeswitch to
>>         release the captured memory.
>>
>>         But is believed, freeswitch must release the momory when
>>         there are no calls. But freeswitch still keeps captured memory.
>>
>>         I have attactted the top command result along with this email.
>>         Willing to perform any test if it helps tackle the problem.
>>
>>         Thanks for your help.
>>
>>
>>
>>
>>         _________________________________________________________________________
>>         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://confluence.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://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/4652fe93/attachment-0001.html 

From brian at freeswitch.org  Fri Mar 27 17:45:06 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 09:45:06 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<55155C0B.2070101@freeswitch.org>
	<CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>
	<CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>
Message-ID: <CAEJMVkC6XygVQe6WgBRf54XwL2Kmjwaevy5u9JFaa2ur9V1+UQ@mail.gmail.com>

FreeSWITCH uses memory pools, so NO in a lot of cases it won't give it
back, it will hit a high water mark based on your use case.  And Drop
caches is dangerous.




On Fri, Mar 27, 2015 at 9:06 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, Brian.
>
> Its ok If vlc takes some ram, but freeswitch must release the ram when
> there are no calls.
> Is there any command that can make freeswitch release the ram without
> restarting the freeswitch.
>
> Like this linux command which makes linux release the cached ram.
>
> free && sync && echo 3 > /proc/sys/vm/drop_caches && free1
>
> Thanks
>
>
> On 27 March 2015 at 18:51, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Yes, we are using linux and looking at RSS value.
>>
>> Our dialplan is mostly using mod_vlc for running streams.Just a call
>> comes in and a radio stream is played with mod_vlc. I am not sure either
>> mod_vlc or something else in freeswitch is eating up ram.
>>
>> I just told our dialplan is mostly based on mod_vlc a portion on
>> mod_curl. I have attached the top command results in above post, you can
>> see freeswitch consuming more ram than any process.
>>
>> One more thing, 6th freeswitch doesn't capture ram more than 1 gb, that
>> one is not using mod_vlc or mod_curl. On other 5 freeswitch we are facing
>> the issue.
>>
>> 6th one freeswitch dialplan just consist of.
>>
>> <extension name="garbage">
>>       <condition>
>>         <action application="answer"/>
>>         <action application="playback"
>> data="/opt/garbage/Generic_VM.wav"/>
>>         <action application="hangup"/>
>>       </condition>
>> </extension>
>>
>>
>> Thanks for your reply.
>>
>> On 27 March 2015 at 18:32, I put the Who? in Mishehu <
>> mishehu at freeswitch.org> wrote:
>>
>>>  Are you using linux?  If so, which memory value are you looking at
>>> (VSZ, RSS...) ?
>>>
>>> If you don't use mod_vlc, do you still see FreeSWITCH consuming this
>>> much RAM?  Usually FreeSWITCH builds up a memory pool and won't release
>>> that back to the system until after shutdown, but I don't normally see that
>>> go above about 1.2GB of RAM in any of the systems I work on (though they do
>>> not use mod_vlc).
>>>
>>> --
>>> Yossi Neiman
>>>
>>>
>>>
>>> On 03/27/2015 07:57 AM, Aqs Younas wrote:
>>>
>>>    Hi, users
>>>
>>>  We are using 6 freeswitch instances, 5 freeswitch instances for playing
>>> streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch
>>> taking more than 3gb of momory and even though calls are not more than 25.
>>>
>>>  Usually, some calls stay for more than 2 to 3 hours on some
>>> freeswitch.  Everytime we have to restart the freeswitch to release the
>>> captured memory.
>>>
>>>  But is believed, freeswitch must release the momory when there are no
>>> calls. But freeswitch still keeps captured memory.
>>>
>>> I have attactted the top command result along with this email.
>>>  Willing to perform any test if it helps tackle the problem.
>>>
>>>  Thanks for your help.
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/cffcf743/attachment.html 

From mike at jerris.com  Fri Mar 27 18:06:40 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Mar 2015 11:06:40 -0400
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <CAEJMVkC6XygVQe6WgBRf54XwL2Kmjwaevy5u9JFaa2ur9V1+UQ@mail.gmail.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>
	<55155C0B.2070101@freeswitch.org>
	<CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>
	<CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>
	<CAEJMVkC6XygVQe6WgBRf54XwL2Kmjwaevy5u9JFaa2ur9V1+UQ@mail.gmail.com>
Message-ID: <F4D38AA1-495E-4C11-86D0-75E228DF783E@jerris.com>

That being said.. there is a known issue in mod_vlc in 1.4 that leaks but I don't think enough to necessarily describe what you are seeing.  We are doing a complete rewrite of that module for 1.6 so that fix may not make it down to 1.4, still to be seen.


> On Mar 27, 2015, at 10:45 AM, Brian West <brian at freeswitch.org> wrote:
> 
> FreeSWITCH uses memory pools, so NO in a lot of cases it won't give it back, it will hit a high water mark based on your use case.  And Drop caches is dangerous.
> 
> 
> 
> 
> On Fri, Mar 27, 2015 at 9:06 AM, Aqs Younas <aqsyounas at gmail.com <mailto:aqsyounas at gmail.com>> wrote:
> Hi, Brian. 
> 
> Its ok If vlc takes some ram, but freeswitch must release the ram when there are no calls.
> Is there any command that can make freeswitch release the ram without restarting the freeswitch. 
> 
> Like this linux command which makes linux release the cached ram.
> 
> free && sync && echo 3 > /proc/sys/vm/drop_caches && free1
> 
> Thanks
> 
> 
> On 27 March 2015 at 18:51, Aqs Younas <aqsyounas at gmail.com <mailto:aqsyounas at gmail.com>> wrote:
> Yes, we are using linux and looking at RSS value. 
> 
> Our dialplan is mostly using mod_vlc for running streams.Just a call comes in and a radio stream is played with mod_vlc. I am not sure either mod_vlc or something else in freeswitch is eating up ram.
> 
> I just told our dialplan is mostly based on mod_vlc a portion on mod_curl. I have attached the top command results in above post, you can see freeswitch consuming more ram than any process. 
> 
> One more thing, 6th freeswitch doesn't capture ram more than 1 gb, that one is not using mod_vlc or mod_curl. On other 5 freeswitch we are facing the issue.
> 
> 6th one freeswitch dialplan just consist of. 
> 
> <extension name="garbage">
>       <condition>
>         <action application="answer"/>
>         <action application="playback" data="/opt/garbage/Generic_VM.wav"/>
>         <action application="hangup"/>
>       </condition>
> </extension>
> 
> 
> Thanks for your reply. 
> 
> On 27 March 2015 at 18:32, I put the Who? in Mishehu <mishehu at freeswitch.org <mailto:mishehu at freeswitch.org>> wrote:
> Are you using linux?  If so, which memory value are you looking at (VSZ, RSS...) ?
> 
> If you don't use mod_vlc, do you still see FreeSWITCH consuming this much RAM?  Usually FreeSWITCH builds up a memory pool and won't release that back to the system until after shutdown, but I don't normally see that go above about 1.2GB of RAM in any of the systems I work on (though they do not use mod_vlc).
> 
> -- 
> Yossi Neiman
> 
> 
> On 03/27/2015 07:57 AM, Aqs Younas wrote:
>> Hi, users
>> 
>> We are using 6 freeswitch instances, 5 freeswitch instances for playing streams with mod_vlc. After 3 to 4 for days,  we see these (5) freeswitch taking more than 3gb of momory and even though calls are not more than 25. 
>> 
>> Usually, some calls stay for more than 2 to 3 hours on some freeswitch.  Everytime we have to restart the freeswitch to release the captured memory.
>> 
>> But is believed, freeswitch must release the momory when there are no calls. But freeswitch still keeps captured memory.
>> 
>> I have attactted the top command result along with this email. 
>> Willing to perform any test if it helps tackle the problem. 
>> 
>> Thanks for your help.
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/81aefefe/attachment-0001.html 

From mishehu at freeswitch.org  Fri Mar 27 19:29:37 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 11:29:37 -0500
Subject: [Freeswitch-users] freeswitch memory leakage.
In-Reply-To: <F4D38AA1-495E-4C11-86D0-75E228DF783E@jerris.com>
References: <CAGVOjVWwN_p9wphYzRD+DU9XGvvJjQtFZ_rc7q7+An-PJxmUwQ@mail.gmail.com>	<55155C0B.2070101@freeswitch.org>	<CAGVOjVXi4Nrv5t_XZhS+pDsY3ux6F5bmcGn5FB1y5MbH9z8b7g@mail.gmail.com>	<CAGVOjVWhEoj9E=rL8YVF1MV4a2mf0ZAkeY2PmOnsKN9fHgfQaA@mail.gmail.com>	<CAEJMVkC6XygVQe6WgBRf54XwL2Kmjwaevy5u9JFaa2ur9V1+UQ@mail.gmail.com>
	<F4D38AA1-495E-4C11-86D0-75E228DF783E@jerris.com>
Message-ID: <55158571.5020909@freeswitch.org>

And even that being said, Aqs Younas, the original poster of the 
question, has never actually described his situation as causing a 
stability problem for any the freeswitch instances or the other running 
programs on those servers.  This may simply be an incorrect assumption 
from the standpoint of how everybody is taught basic memory management: 
free whatever you've allocated as soon as your done with that given object.

In case our original poster (and anybody else who is curious) isn't as 
experienced with memory pools, the general idea behind their use in 
FreeSWITCH is to cut down on memory fragmentation. Every time you 
allocate an object, the operating system has to map out some memory for 
you to use.  If you do this too often for too small of an amount of 
memory, you can negatively impact the performance of the application and 
the system as a whole.  Memory pooling adds another layer onto the 
memory management in that FreeSWITCH in this case requests a larger 
block of memory for allocation, but then it allocates smaller objects 
off of that pool.  This is especially beneficial when the allocations 
happen from within the same places.  It also makes releasing this memory 
a lot simpler because we then simply free the current memory pool, and 
all objects on that given pool are technically no longer valid objects 
("freed").  But it frees the memory pool back to the master memory pool 
that FreeSWITCH uses, and that grows, as Brian stated, up to a given 
"high water mark".  It's only when FreeSWITCH stops running that is 
releases all those blocks of allocated memory back to the operating 
system.  When used properly and in the right situations, as we do in 
FreeSWITCH, memory pools are more efficient.

So 3 GB of RAM used by the application with mod_vlc do not seem to be 
such a stretch.  Heck, I routinely have Mozilla Firefox and Google 
Chrome each grab more RAM than that.  In other words, if there's no 
actual stability issue, this sounds like it's a case of FreeSWITCH 
working in the way it has been engineered to do so.

-- 
Yossi Neiman

On 03/27/2015 10:06 AM, Michael Jerris wrote:

> That being said.. there is a known issue in mod_vlc in 1.4 that leaks 
> but I don't think enough to necessarily describe what you are seeing. 
>  We are doing a complete rewrite of that module for 1.6 so that fix 
> may not make it down to 1.4, still to be seen.
>
>
>> On Mar 27, 2015, at 10:45 AM, Brian West <brian at freeswitch.org 
>> <mailto:brian at freeswitch.org>> wrote:
>>
>> FreeSWITCH uses memory pools, so NO in a lot of cases it won't give 
>> it back, it will hit a high water mark based on your use case.  And 
>> Drop caches is dangerous.
>>
>>
>>
>>
>> On Fri, Mar 27, 2015 at 9:06 AM, Aqs Younas <aqsyounas at gmail.com 
>> <mailto:aqsyounas at gmail.com>> wrote:
>>
>>     Hi, Brian.
>>
>>     Its ok If vlc takes some ram, but freeswitch must release the ram
>>     when there are no calls.
>>     Is there any command that can make freeswitch release the ram
>>     without restarting the freeswitch.
>>
>>     Like this linux command which makes linux release the cached ram.
>>
>>     free && sync && echo 3 > /proc/sys/vm/drop_caches && free1
>>
>>     Thanks
>>
>>
>>     On 27 March 2015 at 18:51, Aqs Younas <aqsyounas at gmail.com
>>     <mailto:aqsyounas at gmail.com>> wrote:
>>
>>         Yes, we are using linux and looking at RSS value.
>>
>>         Our dialplan is mostly using mod_vlc for running streams.Just
>>         a call comes in and a radio stream is played with mod_vlc. I
>>         am not sure either mod_vlc or something else in freeswitch is
>>         eating up ram.
>>
>>         I just told our dialplan is mostly based on mod_vlc a portion
>>         on mod_curl. I have attached the top command results in above
>>         post, you can see freeswitch consuming more ram than any
>>         process.
>>
>>         One more thing, 6th freeswitch doesn't capture ram more than
>>         1 gb, that one is not using mod_vlc or mod_curl. On other 5
>>         freeswitch we are facing the issue.
>>
>>         6th one freeswitch dialplan just consist of.
>>
>>         <extension name="garbage">
>>         <condition>
>>                 <action application="answer"/>
>>                 <action application="playback"
>>         data="/opt/garbage/Generic_VM.wav"/>
>>                 <action application="hangup"/>
>>               </condition>
>>         </extension>
>>
>>
>>         Thanks for your reply.
>>
>>         On 27 March 2015 at 18:32, I put the Who? in Mishehu
>>         <mishehu at freeswitch.org <mailto:mishehu at freeswitch.org>> wrote:
>>
>>             Are you using linux?  If so, which memory value are you
>>             looking at (VSZ, RSS...) ?
>>
>>             If you don't use mod_vlc, do you still see FreeSWITCH
>>             consuming this much RAM? Usually FreeSWITCH builds up a
>>             memory pool and won't release that back to the system
>>             until after shutdown, but I don't normally see that go
>>             above about 1.2GB of RAM in any of the systems I work on
>>             (though they do not use mod_vlc).
>>
>>             -- 
>>             Yossi Neiman
>>
>>
>>             On 03/27/2015 07:57 AM, Aqs Younas wrote:
>>>             Hi, users
>>>
>>>             We are using 6 freeswitch instances, 5 freeswitch
>>>             instances for playing streams with mod_vlc. After 3 to 4
>>>             for days,  we see these (5) freeswitch taking more than
>>>             3gb of momory and even though calls are not more than 25.
>>>
>>>             Usually, some calls stay for more than 2 to 3 hours on
>>>             some freeswitch. Everytime we have to restart the
>>>             freeswitch to release the captured memory.
>>>
>>>             But is believed, freeswitch must release the momory when
>>>             there are no calls. But freeswitch still keeps captured
>>>             memory.
>>>
>>>             I have attactted the top command result along with this
>>>             email.
>>>             Willing to perform any test if it helps tackle the problem.
>>>
>>>             Thanks for your help.
>>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/8081e3b4/attachment-0001.html 

From switcherfree at gmail.com  Fri Mar 27 20:45:00 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Fri, 27 Mar 2015 10:45:00 -0700
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <551558D0.7090803@freeswitch.org>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
Message-ID: <CAM6Un22am59HBoNPCaO+u-tOdMSO-+dhgvHewhQu+YYo6T4nPA@mail.gmail.com>

Hi Yossi,
Thanks for your reply. I got the .repo file from the stash repository.
It is pointing to:
baseurl=http://files.freeswitch.org/yum/$releasever/$basearch

Here is the output of 'rpm -qa |grep -i freeswitch':
freeswitch-1.4.15-1.el6.x86_64

Running 'yum search freeswitch' shows a number of other packages. I
had simply followed the instruction on the wiki that only had: 'yum
install freeswitch'. Looks like that is just the base package and I
need to install modules separately?

Regards,
Andy

On Fri, Mar 27, 2015 at 6:19 AM, I put the Who? in Mishehu
<mishehu at freeswitch.org> wrote:
> Depending on what repo you got those packages from, they could be
> seriously old packages.  Send back the output of:
>
> rpm -qa |grep -i freeswitch
>
> Chances are you only installed the base package.  you can also do `yum
> search freeswitch` to see if other packages are offered. After you
> provide the output from the above command I can comment on the age of
> the packages you installed.
>
> --
> Yossi Neiman
>
> On 03/27/2015 01:12 AM, Free Switcher wrote:
>
>> Hello,
>> I'm trying to create a new installation of freeswitch. I'm running
>> CentOS 6.6 and picked the easy path to install pre-built binaries
>> using yum. Looking at /usr/lib64/freeswitch after the install, I
>> noticed that mod_conference.so wasn't there. Also noticed that no
>> config files were installed in /etc/freeswitch. Is this expected?
>>
>> I downloaded configs from the latest stash repository and I was then
>> able to start freeswitch with vanilla config. I see several startup
>> messages about missing mods. Which modules should I expect to see as
>> part of the pre-built binaries? How/where does one download additional
>> mods? Here is what I have in the /usr/lib64/freeswitch :
>>
>> mod_cdr_csv.so
>> mod_commands.so
>> mod_console.so
>> mod_dialplan_directory.so
>> mod_dialplan_xml.so
>> mod_dptools.so
>> mod_event_socket.so
>> mod_logfile.so
>> mod_loopback.so
>> mod_native_file.so
>> mod_sndfile.so
>> mod_sofia.so
>> mod_spandsp.so
>> mod_syslog.so
>> mod_tone_stream.so
>> mod_xml_rpc.so
>>
>>
>> Any help is appreciated.
>> Thanks,
>> Andy
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From switcherfree at gmail.com  Fri Mar 27 20:49:15 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Fri, 27 Mar 2015 10:49:15 -0700
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
Message-ID: <CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>

Hello Brian,
Thanks for your response. I did not build from source. I just
downloaded/installed pre-built packages using yum. On my CentOS 6.6
machine, it looks like the freeswitch binary is installed under /usr/bin
with configs under /etc/freeswitch.

Thanks,
Andy

On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org> wrote:

> If you downloaded and built the source its all going to be in
> /usr/local/freeswitch unless you configured it with the same configure args
> as the previous package install.
>
> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>> Depending on what repo you got those packages from, they could be
>> seriously old packages.  Send back the output of:
>>
>> rpm -qa |grep -i freeswitch
>>
>> Chances are you only installed the base package.  you can also do `yum
>> search freeswitch` to see if other packages are offered. After you
>> provide the output from the above command I can comment on the age of
>> the packages you installed.
>>
>> --
>> Yossi Neiman
>>
>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>
>> > Hello,
>> > I'm trying to create a new installation of freeswitch. I'm running
>> > CentOS 6.6 and picked the easy path to install pre-built binaries
>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
>> > noticed that mod_conference.so wasn't there. Also noticed that no
>> > config files were installed in /etc/freeswitch. Is this expected?
>> >
>> > I downloaded configs from the latest stash repository and I was then
>> > able to start freeswitch with vanilla config. I see several startup
>> > messages about missing mods. Which modules should I expect to see as
>> > part of the pre-built binaries? How/where does one download additional
>> > mods? Here is what I have in the /usr/lib64/freeswitch :
>> >
>> > mod_cdr_csv.so
>> > mod_commands.so
>> > mod_console.so
>> > mod_dialplan_directory.so
>> > mod_dialplan_xml.so
>> > mod_dptools.so
>> > mod_event_socket.so
>> > mod_logfile.so
>> > mod_loopback.so
>> > mod_native_file.so
>> > mod_sndfile.so
>> > mod_sofia.so
>> > mod_spandsp.so
>> > mod_syslog.so
>> > mod_tone_stream.so
>> > mod_xml_rpc.so
>> >
>> >
>> > Any help is appreciated.
>> > Thanks,
>> > Andy
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/466f3c34/attachment.html 

From brian at freeswitch.org  Fri Mar 27 21:37:28 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 13:37:28 -0500
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
	<CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
Message-ID: <CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>

Can you file a JIRA for us to investigate this?

On Fri, Mar 27, 2015 at 12:49 PM, Free Switcher <switcherfree at gmail.com>
wrote:

> Hello Brian,
> Thanks for your response. I did not build from source. I just
> downloaded/installed pre-built packages using yum. On my CentOS 6.6
> machine, it looks like the freeswitch binary is installed under /usr/bin
> with configs under /etc/freeswitch.
>
> Thanks,
> Andy
>
>
> On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org> wrote:
>
>> If you downloaded and built the source its all going to be in
>> /usr/local/freeswitch unless you configured it with the same configure args
>> as the previous package install.
>>
>> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu <
>> mishehu at freeswitch.org> wrote:
>>
>>> Depending on what repo you got those packages from, they could be
>>> seriously old packages.  Send back the output of:
>>>
>>> rpm -qa |grep -i freeswitch
>>>
>>> Chances are you only installed the base package.  you can also do `yum
>>> search freeswitch` to see if other packages are offered. After you
>>> provide the output from the above command I can comment on the age of
>>> the packages you installed.
>>>
>>> --
>>> Yossi Neiman
>>>
>>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>>
>>> > Hello,
>>> > I'm trying to create a new installation of freeswitch. I'm running
>>> > CentOS 6.6 and picked the easy path to install pre-built binaries
>>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
>>> > noticed that mod_conference.so wasn't there. Also noticed that no
>>> > config files were installed in /etc/freeswitch. Is this expected?
>>> >
>>> > I downloaded configs from the latest stash repository and I was then
>>> > able to start freeswitch with vanilla config. I see several startup
>>> > messages about missing mods. Which modules should I expect to see as
>>> > part of the pre-built binaries? How/where does one download additional
>>> > mods? Here is what I have in the /usr/lib64/freeswitch :
>>> >
>>> > mod_cdr_csv.so
>>> > mod_commands.so
>>> > mod_console.so
>>> > mod_dialplan_directory.so
>>> > mod_dialplan_xml.so
>>> > mod_dptools.so
>>> > mod_event_socket.so
>>> > mod_logfile.so
>>> > mod_loopback.so
>>> > mod_native_file.so
>>> > mod_sndfile.so
>>> > mod_sofia.so
>>> > mod_spandsp.so
>>> > mod_syslog.so
>>> > mod_tone_stream.so
>>> > mod_xml_rpc.so
>>> >
>>> >
>>> > Any help is appreciated.
>>> > Thanks,
>>> > Andy
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/2017e5a2/attachment-0001.html 

From brian at freeswitch.org  Fri Mar 27 22:00:55 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Mar 2015 14:00:55 -0500
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
	<CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
	<CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>
Message-ID: <CAEJMVkCUP3SU9x4_fjxmtGagf5F+zh7J6Z8zX0Fh3w-pe9wxXQ@mail.gmail.com>

Never mind, Its not a bug, Confluence doesn't have the correct process:
 Wiki does
https://wiki.freeswitch.org/wiki/Installation_Guide#YUM_Based_Installation



On Fri, Mar 27, 2015 at 1:37 PM, Brian West <brian at freeswitch.org> wrote:

> Can you file a JIRA for us to investigate this?
>
> On Fri, Mar 27, 2015 at 12:49 PM, Free Switcher <switcherfree at gmail.com>
> wrote:
>
>> Hello Brian,
>> Thanks for your response. I did not build from source. I just
>> downloaded/installed pre-built packages using yum. On my CentOS 6.6
>> machine, it looks like the freeswitch binary is installed under /usr/bin
>> with configs under /etc/freeswitch.
>>
>> Thanks,
>> Andy
>>
>>
>> On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> If you downloaded and built the source its all going to be in
>>> /usr/local/freeswitch unless you configured it with the same configure args
>>> as the previous package install.
>>>
>>> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu <
>>> mishehu at freeswitch.org> wrote:
>>>
>>>> Depending on what repo you got those packages from, they could be
>>>> seriously old packages.  Send back the output of:
>>>>
>>>> rpm -qa |grep -i freeswitch
>>>>
>>>> Chances are you only installed the base package.  you can also do `yum
>>>> search freeswitch` to see if other packages are offered. After you
>>>> provide the output from the above command I can comment on the age of
>>>> the packages you installed.
>>>>
>>>> --
>>>> Yossi Neiman
>>>>
>>>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>>>
>>>> > Hello,
>>>> > I'm trying to create a new installation of freeswitch. I'm running
>>>> > CentOS 6.6 and picked the easy path to install pre-built binaries
>>>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
>>>> > noticed that mod_conference.so wasn't there. Also noticed that no
>>>> > config files were installed in /etc/freeswitch. Is this expected?
>>>> >
>>>> > I downloaded configs from the latest stash repository and I was then
>>>> > able to start freeswitch with vanilla config. I see several startup
>>>> > messages about missing mods. Which modules should I expect to see as
>>>> > part of the pre-built binaries? How/where does one download additional
>>>> > mods? Here is what I have in the /usr/lib64/freeswitch :
>>>> >
>>>> > mod_cdr_csv.so
>>>> > mod_commands.so
>>>> > mod_console.so
>>>> > mod_dialplan_directory.so
>>>> > mod_dialplan_xml.so
>>>> > mod_dptools.so
>>>> > mod_event_socket.so
>>>> > mod_logfile.so
>>>> > mod_loopback.so
>>>> > mod_native_file.so
>>>> > mod_sndfile.so
>>>> > mod_sofia.so
>>>> > mod_spandsp.so
>>>> > mod_syslog.so
>>>> > mod_tone_stream.so
>>>> > mod_xml_rpc.so
>>>> >
>>>> >
>>>> > Any help is appreciated.
>>>> > Thanks,
>>>> > Andy
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY!
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/4c986d21/attachment.html 

From mishehu at freeswitch.org  Fri Mar 27 22:19:07 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 27 Mar 2015 14:19:07 -0500
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAM6Un22am59HBoNPCaO+u-tOdMSO-+dhgvHewhQu+YYo6T4nPA@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>	<551558D0.7090803@freeswitch.org>
	<CAM6Un22am59HBoNPCaO+u-tOdMSO-+dhgvHewhQu+YYo6T4nPA@mail.gmail.com>
Message-ID: <5515AD2B.20802@freeswitch.org>

Yes, this is correct, I believe it still breaks them apart into separate 
rpm's for the modules.  Some people like it this way I suppose (not me, 
but to each his own heh).  :-)

-- 
Yossi Neiman

On 03/27/2015 12:45 PM, Free Switcher wrote:

> Hi Yossi,
> Thanks for your reply. I got the .repo file from the stash repository.
> It is pointing to:
> baseurl=http://files.freeswitch.org/yum/$releasever/$basearch
>
> Here is the output of 'rpm -qa |grep -i freeswitch':
> freeswitch-1.4.15-1.el6.x86_64
>
> Running 'yum search freeswitch' shows a number of other packages. I
> had simply followed the instruction on the wiki that only had: 'yum
> install freeswitch'. Looks like that is just the base package and I
> need to install modules separately?
>
> Regards,
> Andy
>
> On Fri, Mar 27, 2015 at 6:19 AM, I put the Who? in Mishehu
> <mishehu at freeswitch.org> wrote:
>> Depending on what repo you got those packages from, they could be
>> seriously old packages.  Send back the output of:
>>
>> rpm -qa |grep -i freeswitch
>>
>> Chances are you only installed the base package.  you can also do `yum
>> search freeswitch` to see if other packages are offered. After you
>> provide the output from the above command I can comment on the age of
>> the packages you installed.
>>
>> --
>> Yossi Neiman
>>
>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>
>>> Hello,
>>> I'm trying to create a new installation of freeswitch. I'm running
>>> CentOS 6.6 and picked the easy path to install pre-built binaries
>>> using yum. Looking at /usr/lib64/freeswitch after the install, I
>>> noticed that mod_conference.so wasn't there. Also noticed that no
>>> config files were installed in /etc/freeswitch. Is this expected?
>>>
>>> I downloaded configs from the latest stash repository and I was then
>>> able to start freeswitch with vanilla config. I see several startup
>>> messages about missing mods. Which modules should I expect to see as
>>> part of the pre-built binaries? How/where does one download additional
>>> mods? Here is what I have in the /usr/lib64/freeswitch :
>>>
>>> mod_cdr_csv.so
>>> mod_commands.so
>>> mod_console.so
>>> mod_dialplan_directory.so
>>> mod_dialplan_xml.so
>>> mod_dptools.so
>>> mod_event_socket.so
>>> mod_logfile.so
>>> mod_loopback.so
>>> mod_native_file.so
>>> mod_sndfile.so
>>> mod_sofia.so
>>> mod_spandsp.so
>>> mod_syslog.so
>>> mod_tone_stream.so
>>> mod_xml_rpc.so
>>>
>>>
>>> Any help is appreciated.
>>> Thanks,
>>> Andy
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mdalepiane at gmail.com  Fri Mar 27 22:28:02 2015
From: mdalepiane at gmail.com (Mateus Dalepiane)
Date: Fri, 27 Mar 2015 16:28:02 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
Message-ID: <CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>

Hello Brian,

Thank you for the answer. We will consider using Verto in the future.

Right now we will have to stick with WebRTC over SIP, we are using SIP.js
for that.

I ran some more tests and once the Websocket connection drops and is
re-established,
even if we send a re-INVITE, FS identifies it as belonging to the old call,
and
responds to it, after a while FS hangs up the call reporting a
NORMAL_TEMPORARY_FAILURE.

If the Websocket is not disconnected, I can see that FS sends an re-INVITE
to the client after a while,
so I guess that what is happening is that when FS tries to send this
re-INVITE it realizes that the old connection
was closed and hangs up the call.

My question now is: Why FS does not update the connection information for
the call once the re-INVITE from
the new connection is received?

2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:

> Have you taken a look at Verto?
>
> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
> wrote:
>
>> We have the following scenario: The session is established between WebRTC
>> and FreeSWITCH using Websockets.
>>
>> Once the session is established, if the websocket connection drops the
>> media continues to flow util FreeSWITCH tries to send a re-INVITE to the
>> client. At this point it realizes that the connection was closed and hangs
>> up the call.
>>
>> Now, if the websocket connection drops and is re-established, would it be
>> possible to inform FreeSWITCH that the new connection should be used for
>> the previously established session?
>>
>> If the WebRTC client sends an INVITE message with the old session
>> parameters, FreeSWITCH will be able to understand that it belongs to the
>> old session?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/01b6d879/attachment-0001.html 

From jmesquita at freeswitch.org  Fri Mar 27 22:33:30 2015
From: jmesquita at freeswitch.org (=?UTF-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Fri, 27 Mar 2015 16:33:30 -0300
Subject: [Freeswitch-users] FSComm not building in windows
In-Reply-To: <55155995.8070606@freeswitch.org>
References: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>
	<CAA_Nwu474WTRDjAAi-ndtvaPzX-svA-czmKASjpxs-73FtQ+WA@mail.gmail.com>
	<55155995.8070606@freeswitch.org>
Message-ID: <CABH8QX0xjx7wRdd3SRvf87awcXi1B3Hswa0WxsaYrhGqA+zfwg@mail.gmail.com>

What Qt version are you trying to compile agasint? Qt version 5 has changed
considerably and won't ever compile properly...

Jo?o Mesquita

On Fri, Mar 27, 2015 at 10:22 AM, I put the Who? in Mishehu <
mishehu at freeswitch.org> wrote:

>  If you think you've found a bug please file a jira for it.  Though it's
> possible you are simply missing a dependency on your system.  But either
> way, we'll not be able to advise without seeing what those errors actually
> are.
>
> --
> Yossi Neiman
>
> On 03/27/2015 01:31 AM, Ashwin Rath wrote:
>
>  Yes i did
>
>  But even then i get errors like missing files or missing function
> definitions.
>
> On Thu, Mar 26, 2015 at 10:27 PM, Moishe Grunstein <max at nysolutions.com>
> wrote:
>
>>  Did you follow the wiki? https://wiki.freeswitch.org/wiki/FSComm
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Moishe Grunstein
>>
>> Tornado Computer Systems, Inc.
>>
>> 212.400.7650 888.IPPBX.US
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>>
>> Computer Networking * Managed Services * IP Video Surveillance * Network
>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>> Security * Site Surveys * CMS
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashwin Rath
>> *Sent:* Thursday, March 26, 2015 12:40 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] FSComm not building in windows
>>
>>
>>
>> Hi
>>
>> The FSComm app seems to have multiple issues while building on windows.
>>
>> 1) The mod_qsettings ins included in the project but not present on
>> filesystem
>>
>> 2) the ISettings and AccountManager classes are not included in the
>> project
>>
>> 3) Including the above classes causes linker errors for missing QT
>> related methods such as qt_metacast, metaObject etc
>>
>> is FScomm actively maintained or a deprecated project?
>>
>>
>> --
>>
>> Ashwin Rath
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ashwin Kumar Rath
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/78b9092a/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/78b9092a/attachment.jpe 

From mike at jerris.com  Fri Mar 27 22:43:04 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Mar 2015 15:43:04 -0400
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
Message-ID: <7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>

This is not a feature in any of the sip js stacks I know of, and I'm not quite sure how it would be implemented on top of sip.  As Brian said, this is a feature in verto.

> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
> 
> Hello Brian,
> 
> Thank you for the answer. We will consider using Verto in the future.
> 
> Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.
> 
> I ran some more tests and once the Websocket connection drops and is re-established,
> even if we send a re-INVITE, FS identifies it as belonging to the old call, and
> responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.
> 
> If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,
> so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection
> was closed and hangs up the call.
> 
> My question now is: Why FS does not update the connection information for the call once the re-INVITE from
> the new connection is received?
> 
> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
> Have you taken a look at Verto?
> 
> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
> We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.
> 
> Once the session is established, if the websocket connection drops the media continues to flow util FreeSWITCH tries to send a re-INVITE to the client. At this point it realizes that the connection was closed and hangs up the call.
> 
> Now, if the websocket connection drops and is re-established, would it be possible to inform FreeSWITCH that the new connection should be used for the previously established session?
> 
> If the WebRTC client sends an INVITE message with the old session parameters, FreeSWITCH will be able to understand that it belongs to the old session?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/22d15402/attachment-0001.html 

From mdalepiane at gmail.com  Fri Mar 27 22:56:15 2015
From: mdalepiane at gmail.com (Mateus Dalepiane)
Date: Fri, 27 Mar 2015 16:56:15 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
Message-ID: <CAJmiN55zuE6otNJQ8czWHmhmV8S2ghB5bFJYysRbNPXYFHbPZw@mail.gmail.com>

Hello Michael,

The SIP.sj part is working as I expect, the problem as far as I understand
is that FS does not realize that the connection related to the call changed.

I must admit that I don't know how FS handles SIP over TCP, but it seems to
be storing the connection that start the call. I believe it would make
sense to store the connection where the last re-INVITE was received.

2015-03-27 16:43 GMT-03:00 Michael Jerris <mike at jerris.com>:

> This is not a feature in any of the sip js stacks I know of, and I'm not
> quite sure how it would be implemented on top of sip.  As Brian said, this
> is a feature in verto.
>
> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com>
> wrote:
>
> Hello Brian,
>
> Thank you for the answer. We will consider using Verto in the future.
>
> Right now we will have to stick with WebRTC over SIP, we are using SIP.js
> for that.
>
> I ran some more tests and once the Websocket connection drops and is
> re-established,
> even if we send a re-INVITE, FS identifies it as belonging to the old
> call, and
> responds to it, after a while FS hangs up the call reporting a
> NORMAL_TEMPORARY_FAILURE.
>
> If the Websocket is not disconnected, I can see that FS sends an re-INVITE
> to the client after a while,
> so I guess that what is happening is that when FS tries to send this
> re-INVITE it realizes that the old connection
> was closed and hangs up the call.
>
> My question now is: Why FS does not update the connection information for
> the call once the re-INVITE from
> the new connection is received?
>
> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
>
>> Have you taken a look at Verto?
>>
>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>> wrote:
>>
>>> We have the following scenario: The session is established between
>>> WebRTC and FreeSWITCH using Websockets.
>>>
>>> Once the session is established, if the websocket connection drops the
>>> media continues to flow util FreeSWITCH tries to send a re-INVITE to the
>>> client. At this point it realizes that the connection was closed and hangs
>>> up the call.
>>>
>>> Now, if the websocket connection drops and is re-established, would it
>>> be possible to inform FreeSWITCH that the new connection should be used for
>>> the previously established session?
>>>
>>> If the WebRTC client sends an INVITE message with the old session
>>> parameters, FreeSWITCH will be able to understand that it belongs to the
>>> old session?
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/d72e44d3/attachment.html 

From tfred31 at yahoo.com  Fri Mar 27 23:02:30 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Fri, 27 Mar 2015 13:02:30 -0700
Subject: [Freeswitch-users] transfer_fallback_extension
In-Reply-To: <CFF69A91-8CB3-4542-880C-0507559DEC30@gmail.com>
Message-ID: <1427486550.89589.YahooMailBasic@web160202.mail.bf1.yahoo.com>

I very much appreciate your reply.
And, I even more greatly appreciate the thoroughness of your reply.

It sounds like exactly what I am looking for.
I will try it and see how it goes.

Thanks again.



--------------------------------------------
On Fri, 3/27/15, Raphael Lechner <raphael.lechner at gmail.com> wrote:

 Subject: Re: [Freeswitch-users] transfer_fallback_extension
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Friday, March 27, 2015, 3:21 AM
 
 Hi,
 
 I have created the following dial plan for
 that.
 
 If the call is not
 answered after 30 seconds then the call is going back to the
 initial caller and on the display is showing
 (NO_ANSWER)>>orig_caller_id_name>>orig_caller_id_number.
 
 <extension
 name="Local_Extension">
 ?
 <condition field="destination_number"
 expression="^([0-9][0-9])$">
 ?
 ? <action application="export"
 data="dialed_extension=$1" />
 ? ? <action
 application="bind_meta_app" data="1 b s
 execute_extension::dx XML features" />
 ? ? <action
 application="bind_meta_app"
 ? ?
 ? data="2 b s
 record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
 />
 ? ? <action
 application="bind_meta_app" data="3 b s
 execute_extension::cf XML features" />
 ? ? <action
 application="bind_meta_app" data="4 b s
 execute_extension::att_xfer XML features" />
 ? ? <action application="set"
 data="transfer_ringback=$${hold_music}" />
 ? ? <action application="set"
 data="call_timeout=30" />
 ? ?
 <action application="set"
 data="hangup_after_bridge=true" />
 ? ? <action application="set"
 data="continue_on_fail=NO_ANSWER" />
 ? ? <action application="hash"
 data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"
 />
 ? ? <action
 application="set"
 data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
 var callgroup)}" />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/global/${uuid}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"
 />
 ? ? <action
 application="set"
 ? ? ?
 data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? <action application="set"
 data="effective_callee_id_number=${dialed_extension}"
 />
 ? ? <action
 application="log" data="INFO
 effective_callee_id_number=${dialed_extension}"
 />
 ? ? <action
 application="log"
 ? ? ?
 data="INFO
 effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? <action application="export"
 data="alert_info=http://localhost/;info=internal"
 />
 ? ? <action
 application="log" data="INFO
 alert_info=${alert_info}" />
 
 ? ? <condition
 field="${last_bridge_hangup_cause}"
 expression="^NO_ANSWER$">
 ? ?
 ? <action application="log"
 ?
 ? ? ? data="INFO NO_ANSWER --
 ${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})"
 />
 ? ? ? <action
 application="export"
 ? ? ? ?
 data="origination_caller_id_name=${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})"
 />
 ? ? ? <anti-action
 application="set"
 ? ? ? ?
 data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? </condition>
 
 ? ? <action
 application="bridge"
 ? ? ?
 data="{effective_callee_id_name=${effective_callee_id_name},effective_callee_id_number=${effective_callee_id_number},origination_caller_id_name=${origination_caller_id_name}}user/${dialed_extension}@${domain_name}"
 />
 ? ? <action
 application="transfer"
 data="${last_sent_callee_id_number}" />
 ? </condition>
 </extension>
 
 Raphael
 
 >
 On 26 Mar 2015, at 18:55, T Fred Farmington <tfred31 at yahoo.com>
 wrote:
 > 
 > On a
 number of websites and in my conf/directory/default.xml file
 I see:???
 >?
 ???<variable name =
 "transfer_fallback_extension" value =
 "operator"/>
 > 
 > And in my? ?
 conf/dialplan/default.xml???file I see?
 ???<extension name =
 "operator">? ? where the operator
 'extension' is defined as a single extension
 number.
 > 
 > But I
 don't want the failed transfer attempt to go to a single
 pre-defined extension 
 > I want the
 failed transfer attempt to go back to the extension which
 initially launched the transfer attempt
 >
 
 > How/where would I modify the
 variable???transfer_fallback_extension???to
 get this to work as needed?
 > 
 > Or should this be handled in a different
 manner somewhere else within FreeSWITCH?
 > If so, where/how?
 > 
 > Thanks
 > 
 > 
 > 
 > 
 > 
 > 
 >
 _________________________________________________________________________
 > Professional FreeSWITCH Consulting
 Services: 
 > consulting at freeswitch.org
 > http://www.freeswitchsolutions.com
 > 
 > Official FreeSWITCH
 Sites
 > http://www.freeswitch.org
 > http://confluence.freeswitch.org
 > http://www.cluecon.com
 > 
 > FreeSWITCH-users
 mailing list
 > FreeSWITCH-users at lists.freeswitch.org
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 


From tfred31 at yahoo.com  Fri Mar 27 23:02:30 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Fri, 27 Mar 2015 13:02:30 -0700
Subject: [Freeswitch-users] transfer_fallback_extension
In-Reply-To: <CFF69A91-8CB3-4542-880C-0507559DEC30@gmail.com>
Message-ID: <1427486550.89589.YahooMailBasic@web160202.mail.bf1.yahoo.com>

I very much appreciate your reply.
And, I even more greatly appreciate the thoroughness of your reply.

It sounds like exactly what I am looking for.
I will try it and see how it goes.

Thanks again.



--------------------------------------------
On Fri, 3/27/15, Raphael Lechner <raphael.lechner at gmail.com> wrote:

 Subject: Re: [Freeswitch-users] transfer_fallback_extension
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Friday, March 27, 2015, 3:21 AM
 
 Hi,
 
 I have created the following dial plan for
 that.
 
 If the call is not
 answered after 30 seconds then the call is going back to the
 initial caller and on the display is showing
 (NO_ANSWER)>>orig_caller_id_name>>orig_caller_id_number.
 
 <extension
 name="Local_Extension">
 ?
 <condition field="destination_number"
 expression="^([0-9][0-9])$">
 ?
 ? <action application="export"
 data="dialed_extension=$1" />
 ? ? <action
 application="bind_meta_app" data="1 b s
 execute_extension::dx XML features" />
 ? ? <action
 application="bind_meta_app"
 ? ?
 ? data="2 b s
 record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"
 />
 ? ? <action
 application="bind_meta_app" data="3 b s
 execute_extension::cf XML features" />
 ? ? <action
 application="bind_meta_app" data="4 b s
 execute_extension::att_xfer XML features" />
 ? ? <action application="set"
 data="transfer_ringback=$${hold_music}" />
 ? ? <action application="set"
 data="call_timeout=30" />
 ? ?
 <action application="set"
 data="hangup_after_bridge=true" />
 ? ? <action application="set"
 data="continue_on_fail=NO_ANSWER" />
 ? ? <action application="hash"
 data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"
 />
 ? ? <action
 application="set"
 data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
 var callgroup)}" />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial_ext/global/${uuid}"
 />
 ? ? <action
 application="hash"
 data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"
 />
 ? ? <action
 application="set"
 ? ? ?
 data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? <action application="set"
 data="effective_callee_id_number=${dialed_extension}"
 />
 ? ? <action
 application="log" data="INFO
 effective_callee_id_number=${dialed_extension}"
 />
 ? ? <action
 application="log"
 ? ? ?
 data="INFO
 effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? <action application="export"
 data="alert_info=http://localhost/;info=internal"
 />
 ? ? <action
 application="log" data="INFO
 alert_info=${alert_info}" />
 
 ? ? <condition
 field="${last_bridge_hangup_cause}"
 expression="^NO_ANSWER$">
 ? ?
 ? <action application="log"
 ?
 ? ? ? data="INFO NO_ANSWER --
 ${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})"
 />
 ? ? ? <action
 application="export"
 ? ? ? ?
 data="origination_caller_id_name=${dialed_extension}(NO_ANSWER)>>${orig_caller_id_name}(${orig_caller_id_number})"
 />
 ? ? ? <anti-action
 application="set"
 ? ? ? ?
 data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name}
 var effective_caller_id_name)}" />
 ? ? </condition>
 
 ? ? <action
 application="bridge"
 ? ? ?
 data="{effective_callee_id_name=${effective_callee_id_name},effective_callee_id_number=${effective_callee_id_number},origination_caller_id_name=${origination_caller_id_name}}user/${dialed_extension}@${domain_name}"
 />
 ? ? <action
 application="transfer"
 data="${last_sent_callee_id_number}" />
 ? </condition>
 </extension>
 
 Raphael
 
 >
 On 26 Mar 2015, at 18:55, T Fred Farmington <tfred31 at yahoo.com>
 wrote:
 > 
 > On a
 number of websites and in my conf/directory/default.xml file
 I see:???
 >?
 ???<variable name =
 "transfer_fallback_extension" value =
 "operator"/>
 > 
 > And in my? ?
 conf/dialplan/default.xml???file I see?
 ???<extension name =
 "operator">? ? where the operator
 'extension' is defined as a single extension
 number.
 > 
 > But I
 don't want the failed transfer attempt to go to a single
 pre-defined extension 
 > I want the
 failed transfer attempt to go back to the extension which
 initially launched the transfer attempt
 >
 
 > How/where would I modify the
 variable???transfer_fallback_extension???to
 get this to work as needed?
 > 
 > Or should this be handled in a different
 manner somewhere else within FreeSWITCH?
 > If so, where/how?
 > 
 > Thanks
 > 
 > 
 > 
 > 
 > 
 > 
 >
 _________________________________________________________________________
 > Professional FreeSWITCH Consulting
 Services: 
 > consulting at freeswitch.org
 > http://www.freeswitchsolutions.com
 > 
 > Official FreeSWITCH
 Sites
 > http://www.freeswitch.org
 > http://confluence.freeswitch.org
 > http://www.cluecon.com
 > 
 > FreeSWITCH-users
 mailing list
 > FreeSWITCH-users at lists.freeswitch.org
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 


From alhakeem at gmail.com  Fri Mar 27 23:05:46 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Fri, 27 Mar 2015 20:05:46 -0000
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>

Hi Guys,
What's the best recommended client to connect to Verto ?
Cheers,
Abdul Hakeem
 
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Friday, March 27, 2015 7:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-establish connection within a SIP session
 
This is not a feature in any of the sip js stacks I know of, and I'm not quite
sure how it would be implemented on top of sip.  As Brian said, this is a
feature in verto.
 
On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
 
Hello Brian,
Thank you for the answer. We will consider using Verto in the future.

Right now we will have to stick with WebRTC over SIP, we are using SIP.js for
that.

I ran some more tests and once the Websocket connection drops and is
re-established,
even if we send a re-INVITE, FS identifies it as belonging to the old call, and
responds to it, after a while FS hangs up the call reporting a
NORMAL_TEMPORARY_FAILURE.
If the Websocket is not disconnected, I can see that FS sends an re-INVITE to
the client after a while,
so I guess that what is happening is that when FS tries to send this re-INVITE
it realizes that the old connection
was closed and hangs up the call.
My question now is: Why FS does not update the connection information for the
call once the re-INVITE from
the new connection is received?
 
2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
Have you taken a look at Verto?
 
On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
We have the following scenario: The session is established between WebRTC and
FreeSWITCH using Websockets.
 
Once the session is established, if the websocket connection drops the media
continues to flow util FreeSWITCH tries to send a re-INVITE to the client. At
this point it realizes that the connection was closed and hangs up the call.
 
Now, if the websocket connection drops and is re-established, would it be
possible to inform FreeSWITCH that the new connection should be used for the
previously established session?
 
If the WebRTC client sends an INVITE message with the old session parameters,
FreeSWITCH will be able to understand that it belongs to the old session?
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/bbf8eef6/attachment-0001.html 

From mike at jerris.com  Fri Mar 27 23:15:01 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Mar 2015 16:15:01 -0400
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
Message-ID: <90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>

verto has its own JS client in tree.

> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hi Guys,
> What?s the best recommended client to connect to Verto ?
> Cheers,
> Abdul Hakeem
> ? <>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Friday, March 27, 2015 7:43 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Re-establish connection within a SIP session
>  
> This is not a feature in any of the sip js stacks I know of, and I'm not quite sure how it would be implemented on top of sip.  As Brian said, this is a feature in verto.
>  
>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>  
>> Hello Brian,
>> 
>> Thank you for the answer. We will consider using Verto in the future.
>> 
>> Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.
>> 
>> I ran some more tests and once the Websocket connection drops and is re-established,
>> even if we send a re-INVITE, FS identifies it as belonging to the old call, and
>> responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.
>> 
>> If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,
>> so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection
>> was closed and hangs up the call.
>> 
>> My question now is: Why FS does not update the connection information for the call once the re-INVITE from
>> the new connection is received?
>>  
>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
>> Have you taken a look at Verto?
>>  
>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>> We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.
>>>  
>>> Once the session is established, if the websocket connection drops the media continues to flow utilFreeSWITCH tries to send a re-INVITE to the client. At this point it realizes that the connection was closed and hangs up the call.
>>>  
>>> Now, if the websocket connection drops and is re-established, would it be possible to inform FreeSWITCH that the new connection should be used for the previously established session?
>>>  
>>> If the WebRTC client sends an INVITE message with the old session parameters, FreeSWITCH will be able to understand that it belongs to the old session?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/5593c21f/attachment.html 

From mdalepiane at gmail.com  Sat Mar 28 00:31:08 2015
From: mdalepiane at gmail.com (Mateus Dalepiane)
Date: Fri, 27 Mar 2015 18:31:08 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
Message-ID: <CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>

Hey guys,

Thank you all for the attention and patience to respond my questions.

I understand that the ideal solution would be to use Verto, but that's not
practicable in our project right now.

So, about the reconnected session, I am not sure if I made myself clear
about what is happening, and what I am trying to do.

    WebRTC client  . . . nginx  . . . . FreeSWITCH
      (SIP.js)           proxy               |
         |                 |                 |
         |     CONNECT     |                 |
         |---------------->|                 |
         |              INVITE               |
         |---------------------------------->|
         |                OK                 |
         |<----------------------------------|
         |                ACK                |
         |---------------------------------->|
         |           Media Session           |
         |<=================================>|
         |                 .                 |
         |                 .                 |
         |                 .                 |
         | CONNECTION FAIL |                 |
         |<-----XXXX------>|                 |
         |       Media continues to flow     |
         |<=================================>|
         |     CONNECT     |                 |
         |---------------->|                 |
         |              re-INVITE            |
         |---------------------------------->|
         |                OK                 |
         |<----------------------------------|
         |                ACK                |
         |---------------------------------->|
         |                 .                 |
         |                 .                 |
         |                 .                 |
         |                 |     INVITE      |
         |                 |<------XXX-------|
         |                 |                 | FS hang up call
         |         Media stop flowing        |
         |<==============XXXXX==============>|

So, based on this scenario, when the Websocket connection to nginx fails,
we reconnect it, but since the media is going through other connections,
RTP over UDP, it is not affected.

Now, with the new websocket connection in place the client is able to send
re-INVITEs and BYE to FS, and it is recognized as requests for the session
established using the first connection.

The problem is that when FS tries to send a message to the client it fails
(NORMAL_TEMPORARY_FAILURE) and hangs up the call.

Right now my question is:
 - How does FS know which connection it should use to send SIP messages to
the client?

Thank you!

2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com>:

> verto has its own JS client in tree.
>
> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>
> Hi Guys,
> What?s the best recommended client to connect to Verto ?
> Cheers,
> Abdul Hakeem
>
> *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 *Michael
> Jerris
> *Sent:* Friday, March 27, 2015 7:43 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Re-establish connection within a SIP
> session
>
> This is not a feature in any of the sip js stacks I know of, and I'm not
> quite sure how it would be implemented on top of sip.  As Brian said, this
> is a feature in verto.
>
>
> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com>
> wrote:
>
>
> Hello Brian,
> Thank you for the answer. We will consider using Verto in the future.
>
> Right now we will have to stick with WebRTC over SIP, we are using SIP.js
> for that.
>
> I ran some more tests and once the Websocket connection drops and is
> re-established,
> even if we send a re-INVITE, FS identifies it as belonging to the old
> call, and
>
> responds to it, after a while FS hangs up the call reporting a
> NORMAL_TEMPORARY_FAILURE.
> If the Websocket is not disconnected, I can see that FS sends an
> re-INVITE to the client after a while,
> so I guess that what is happening is that when FS tries to send this
> re-INVITE it realizes that the old connection
>
> was closed and hangs up the call.
> My question now is: Why FS does not update the connection information for
> the call once the re-INVITE from
> the new connection is received?
>
> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
> Have you taken a look at Verto?
>
> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
> wrote:
>
> We have the following scenario: The session is established between WebRTC
> and FreeSWITCH using Websockets.
>
> Once the session is established, if the websocket connection drops the
> media continues to flow utilFreeSWITCH tries to send a re-INVITE to the
> client. At this point it realizes that the connection was closed and hangs
> up the call.
>
> Now, if the websocket connection drops and is re-established, would it be
> possible to inform FreeSWITCH that the new connection should be used for
> the previously established session?
>
> If the WebRTC client sends an INVITE message with the old session
> parameters, FreeSWITCH will be able to understand that it belongs to the
> old session?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/2f737d98/attachment-0001.html 

From mike at jerris.com  Sat Mar 28 01:31:00 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Mar 2015 18:31:00 -0400
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
	<CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
Message-ID: <525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>

I understand what you are trying to do.  The current sofia code has no way to handle this currently, and we don't have plans to add this functionality, because we can already do so with mod_verto.  If you want this functionality in freeswitch, your options are using mod_verto, or a huge amount of c code in mod_sofia that will be very error prone as it will have to turn the state engine in that module on its head.

> On Mar 27, 2015, at 5:31 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
> 
> Hey guys,
> 
> Thank you all for the attention and patience to respond my questions.
> 
> I understand that the ideal solution would be to use Verto, but that's not practicable in our project right now.
> 
> So, about the reconnected session, I am not sure if I made myself clear about what is happening, and what I am trying to do.
> 
>     WebRTC client  . . . nginx  . . . . FreeSWITCH
>       (SIP.js)           proxy               |
>          |                 |                 |
>          |     CONNECT     |                 |
>          |---------------->|                 |
>          |              INVITE               |
>          |---------------------------------->|
>          |                OK                 |
>          |<----------------------------------|
>          |                ACK                |
>          |---------------------------------->|
>          |           Media Session           |
>          |<=================================>|
>          |                 .                 |
>          |                 .                 |
>          |                 .                 |
>          | CONNECTION FAIL |                 |
>          |<-----XXXX------>|                 |
>          |       Media continues to flow     |
>          |<=================================>|
>          |     CONNECT     |                 |
>          |---------------->|                 |
>          |              re-INVITE            |
>          |---------------------------------->|
>          |                OK                 |
>          |<----------------------------------|
>          |                ACK                |
>          |---------------------------------->|
>          |                 .                 |
>          |                 .                 |
>          |                 .                 |
>          |                 |     INVITE      |
>          |                 |<------XXX-------|
>          |                 |                 | FS hang up call
>          |         Media stop flowing        |
>          |<==============XXXXX==============>|
> 
> So, based on this scenario, when the Websocket connection to nginx fails, we reconnect it, but since the media is going through other connections, RTP over UDP, it is not affected.
> 
> Now, with the new websocket connection in place the client is able to send re-INVITEs and BYE to FS, and it is recognized as requests for the session established using the first connection.
> 
> The problem is that when FS tries to send a message to the client it fails (NORMAL_TEMPORARY_FAILURE) and hangs up the call.
> 
> Right now my question is:
>  - How does FS know which connection it should use to send SIP messages to the client?
> 
> Thank you!
> 
> 2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> verto has its own JS client in tree.
> 
>> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com <mailto:alhakeem at gmail.com>> wrote:
>> 
>> Hi Guys,
>> What?s the best recommended client to connect to Verto ?
>> Cheers,
>> Abdul Hakeem
>> ? <>
>> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
>> Sent: Friday, March 27, 2015 7:43 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Re-establish connection within a SIP session
>>  
>> This is not a feature in any of the sip js stacks I know of, and I'm not quite sure how it would be implemented on top of sip.  As Brian said, this is a feature in verto.
>>  
>>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>>  
>>> Hello Brian,
>>> 
>>> Thank you for the answer. We will consider using Verto in the future.
>>> 
>>> Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.
>>> 
>>> I ran some more tests and once the Websocket connection drops and is re-established,
>>> even if we send a re-INVITE, FS identifies it as belonging to the old call, and
>>> responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.
>>> 
>>> If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,
>>> so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection
>>> was closed and hangs up the call.
>>> 
>>> My question now is: Why FS does not update the connection information for the call once the re-INVITE from
>>> the new connection is received?
>>>  
>>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
>>> Have you taken a look at Verto?
>>>  
>>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>>> We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.
>>>>  
>>>> Once the session is established, if the websocket connection drops the media continues to flow utilFreeSWITCH tries to send a re-INVITE to the client. At this point it realizes that the connection was closed and hangs up the call.
>>>>  
>>>> Now, if the websocket connection drops and is re-established, would it be possible to inform FreeSWITCH that the new connection should be used for the previously established session?
>>>>  
>>>> If the WebRTC client sends an INVITE message with the old session parameters, FreeSWITCH will be able to understand that it belongs to the old session?
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/096aff9e/attachment-0001.html 

From mike at jerris.com  Sat Mar 28 01:35:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Mar 2015 18:35:27 -0400
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
	<CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
	<525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
Message-ID: <59E74FB1-AAE4-4719-AD2B-BF473DC66816@jerris.com>

Out of curiosity, what about using verto makes it not practical in your project right now?

> On Mar 27, 2015, at 6:31 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> I understand what you are trying to do.  The current sofia code has no way to handle this currently, and we don't have plans to add this functionality, because we can already do so with mod_verto.  If you want this functionality in freeswitch, your options are using mod_verto, or a huge amount of c code in mod_sofia that will be very error prone as it will have to turn the state engine in that module on its head.
> 
>> On Mar 27, 2015, at 5:31 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>> 
>> Hey guys,
>> 
>> Thank you all for the attention and patience to respond my questions.
>> 
>> I understand that the ideal solution would be to use Verto, but that's not practicable in our project right now.
>> 
>> So, about the reconnected session, I am not sure if I made myself clear about what is happening, and what I am trying to do.
>> 
>>     WebRTC client  . . . nginx  . . . . FreeSWITCH
>>       (SIP.js)           proxy               |
>>          |                 |                 |
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              INVITE               |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |           Media Session           |
>>          |<=================================>|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          | CONNECTION FAIL |                 |
>>          |<-----XXXX------>|                 |
>>          |       Media continues to flow     |
>>          |<=================================>|
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              re-INVITE            |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          |                 |     INVITE      |
>>          |                 |<------XXX-------|
>>          |                 |                 | FS hang up call
>>          |         Media stop flowing        |
>>          |<==============XXXXX==============>|
>> 
>> So, based on this scenario, when the Websocket connection to nginx fails, we reconnect it, but since the media is going through other connections, RTP over UDP, it is not affected.
>> 
>> Now, with the new websocket connection in place the client is able to send re-INVITEs and BYE to FS, and it is recognized as requests for the session established using the first connection.
>> 
>> The problem is that when FS tries to send a message to the client it fails (NORMAL_TEMPORARY_FAILURE) and hangs up the call.
>> 
>> Right now my question is:
>>  - How does FS know which connection it should use to send SIP messages to the client?
>> 
>> Thank you!
>> 
>> 2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
>> verto has its own JS client in tree.
>> 
>>> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com <mailto:alhakeem at gmail.com>> wrote:
>>> 
>>> Hi Guys,
>>> What?s the best recommended client to connect to Verto ?
>>> Cheers,
>>> Abdul Hakeem
>>> ? <>
>>> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
>>> Sent: Friday, March 27, 2015 7:43 PM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Re-establish connection within a SIP session
>>>  
>>> This is not a feature in any of the sip js stacks I know of, and I'm not quite sure how it would be implemented on top of sip.  As Brian said, this is a feature in verto.
>>>  
>>>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>>>  
>>>> Hello Brian,
>>>> 
>>>> Thank you for the answer. We will consider using Verto in the future.
>>>> 
>>>> Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.
>>>> 
>>>> I ran some more tests and once the Websocket connection drops and is re-established,
>>>> even if we send a re-INVITE, FS identifies it as belonging to the old call, and
>>>> responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.
>>>> 
>>>> If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,
>>>> so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection
>>>> was closed and hangs up the call.
>>>> 
>>>> My question now is: Why FS does not update the connection information for the call once the re-INVITE from
>>>> the new connection is received?
>>>>  
>>>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
>>>> Have you taken a look at Verto?
>>>>  
>>>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com <mailto:mdalepiane at gmail.com>> wrote:
>>>>> We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.
>>>>>  
>>>>> Once the session is established, if the websocket connection drops the media continues to flow utilFreeSWITCH tries to send a re-INVITE to the client. At this point it realizes that the connection was closed and hangs up the call.
>>>>>  
>>>>> Now, if the websocket connection drops and is re-established, would it be possible to inform FreeSWITCH that the new connection should be used for the previously established session?
>>>>>  
>>>>> If the WebRTC client sends an INVITE message with the old session parameters, FreeSWITCH will be able to understand that it belongs to the old session?
>> 
>> _
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/caf4fa66/attachment-0001.html 

From anthony.minessale at gmail.com  Sat Mar 28 01:38:41 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 27 Mar 2015 17:38:41 -0500
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
	<CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
	<525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
Message-ID: <CAKbxfG95zTRF65ohsTmjJErdXBszkoSUux_qHii==cBeyd6T=A@mail.gmail.com>

You could probably use a proxy like kamailio or opensips to translate the
websockets to UDP or TCP and pass it on to FS but FS itself cannot be
modified to do what you want.


On Fri, Mar 27, 2015 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:

> I understand what you are trying to do.  The current sofia code has no way
> to handle this currently, and we don't have plans to add this
> functionality, because we can already do so with mod_verto.  If you want
> this functionality in freeswitch, your options are using mod_verto, or a
> huge amount of c code in mod_sofia that will be very error prone as it will
> have to turn the state engine in that module on its head.
>
> On Mar 27, 2015, at 5:31 PM, Mateus Dalepiane <mdalepiane at gmail.com>
> wrote:
>
> Hey guys,
>
> Thank you all for the attention and patience to respond my questions.
>
> I understand that the ideal solution would be to use Verto, but that's not
> practicable in our project right now.
>
> So, about the reconnected session, I am not sure if I made myself clear
> about what is happening, and what I am trying to do.
>
>     WebRTC client  . . . nginx  . . . . FreeSWITCH
>       (SIP.js)           proxy               |
>          |                 |                 |
>          |     CONNECT     |                 |
>          |---------------->|                 |
>          |              INVITE               |
>          |---------------------------------->|
>          |                OK                 |
>          |<----------------------------------|
>          |                ACK                |
>          |---------------------------------->|
>          |           Media Session           |
>          |<=================================>|
>          |                 .                 |
>          |                 .                 |
>          |                 .                 |
>          | CONNECTION FAIL |                 |
>          |<-----XXXX------>|                 |
>          |       Media continues to flow     |
>          |<=================================>|
>          |     CONNECT     |                 |
>          |---------------->|                 |
>          |              re-INVITE            |
>          |---------------------------------->|
>          |                OK                 |
>          |<----------------------------------|
>          |                ACK                |
>          |---------------------------------->|
>          |                 .                 |
>          |                 .                 |
>          |                 .                 |
>          |                 |     INVITE      |
>          |                 |<------XXX-------|
>          |                 |                 | FS hang up call
>          |         Media stop flowing        |
>          |<==============XXXXX==============>|
>
> So, based on this scenario, when the Websocket connection to nginx fails,
> we reconnect it, but since the media is going through other connections,
> RTP over UDP, it is not affected.
>
> Now, with the new websocket connection in place the client is able to send
> re-INVITEs and BYE to FS, and it is recognized as requests for the session
> established using the first connection.
>
> The problem is that when FS tries to send a message to the client it fails
> (NORMAL_TEMPORARY_FAILURE) and hangs up the call.
>
> Right now my question is:
>  - How does FS know which connection it should use to send SIP messages to
> the client?
>
> Thank you!
>
> 2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com>:
>
>> verto has its own JS client in tree.
>>
>> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>>
>> Hi Guys,
>> What?s the best recommended client to connect to Verto ?
>> Cheers,
>> Abdul Hakeem
>>
>> *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 *Michael
>> Jerris
>> *Sent:* Friday, March 27, 2015 7:43 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Re-establish connection within a SIP
>> session
>>
>> This is not a feature in any of the sip js stacks I know of, and I'm not
>> quite sure how it would be implemented on top of sip.  As Brian said, this
>> is a feature in verto.
>>
>>
>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>> wrote:
>>
>>
>> Hello Brian,
>> Thank you for the answer. We will consider using Verto in the future.
>>
>> Right now we will have to stick with WebRTC over SIP, we are using SIP.js
>> for that.
>>
>> I ran some more tests and once the Websocket connection drops and is
>> re-established,
>> even if we send a re-INVITE, FS identifies it as belonging to the old
>> call, and
>>
>> responds to it, after a while FS hangs up the call reporting a
>> NORMAL_TEMPORARY_FAILURE.
>> If the Websocket is not disconnected, I can see that FS sends an
>> re-INVITE to the client after a while,
>> so I guess that what is happening is that when FS tries to send this
>> re-INVITE it realizes that the old connection
>>
>> was closed and hangs up the call.
>> My question now is: Why FS does not update the connection information for
>> the call once the re-INVITE from
>> the new connection is received?
>>
>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
>> Have you taken a look at Verto?
>>
>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>> wrote:
>>
>> We have the following scenario: The session is established between WebRTC
>> and FreeSWITCH using Websockets.
>>
>> Once the session is established, if the websocket connection drops the
>> media continues to flow utilFreeSWITCH tries to send a re-INVITE to the
>> client. At this point it realizes that the connection was closed and hangs
>> up the call.
>>
>> Now, if the websocket connection drops and is re-established, would it
>> be possible to inform FreeSWITCH that the new connection should be used for
>> the previously established session?
>>
>> If the WebRTC client sends an INVITE message with the old session
>> parameters, FreeSWITCH will be able to understand that it belongs to the
>> old session?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/d48d936e/attachment-0001.html 

From anthony.minessale at gmail.com  Sat Mar 28 01:39:55 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 27 Mar 2015 17:39:55 -0500
Subject: [Freeswitch-users] unexpected segfault with latest debian and
	libmyodbc.so
In-Reply-To: <CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>
References: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>
	<CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>
Message-ID: <CAKbxfG8+=HVT_D7FqUwdynRa8-8KEvkOVzuPnjF6tnmZipuvQA@mail.gmail.com>

We actually recommend Threading=0 for postgres and mysql for the most part.


On Fri, Mar 27, 2015 at 7:48 AM, ?talo Rossi <italorossib at gmail.com> wrote:

> I have seen a lot of threading issues with mysql + odbc, make sure you're
> using unixodbc >= 2.3.
>
> If you're using an older version you can set Threading = 2 in your
> /etc/odbcinst.ini as a workaround, but this is *not* recommended for
> production/high volume, upgrade as soon as possible.
>
> On Fri, Mar 27, 2015 at 9:41 AM, Yuriy Nasida <nasida at live.ru> wrote:
>
>> Hi guys,
>>
>> I just got unexpected segfault I try to understand if anybody had similar
>> problems.
>>
>> *Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at
>> 500 ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in
>> libmyodbc.so[7fd3a8340000+3c000]*
>>
>>
>> # lsb_release -a
>> Distributor ID:    Debian
>> Description:    Debian GNU/Linux 7.8 (wheezy)
>> Release:    7.8
>> Codename:    wheezy
>>
>> freeswitch at internal> version
>> FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git
>> 507a0f2 2014-12-29 18:59:51Z 64bit)
>>
>> # apt-cache show libmyodbc
>> Package: libmyodbc
>> Source: myodbc
>> Version: 5.1.10-2+deb7u1
>>
>> Please advice,
>> Thanks.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/e3d28867/attachment.html 

From mdalepiane at gmail.com  Sat Mar 28 02:12:09 2015
From: mdalepiane at gmail.com (Mateus Dalepiane)
Date: Fri, 27 Mar 2015 20:12:09 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAKbxfG95zTRF65ohsTmjJErdXBszkoSUux_qHii==cBeyd6T=A@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
	<CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
	<525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
	<CAKbxfG95zTRF65ohsTmjJErdXBszkoSUux_qHii==cBeyd6T=A@mail.gmail.com>
Message-ID: <CAJmiN57AXZ6T=P2TD4Q3Osv6BnEipgjLhG+9FqVRqFnOZJqk=w@mail.gmail.com>

Alright, now I am convinced that this is not a viable approach.

> Out of curiosity, what about using verto makes it not practical in your
project right now?
Michael, I am working on Mconf, a project based on BigBlueButton. We do
plan on moving the WebRTC support from SIP.js to Verto, but right now we
were looking into this as a temporary solution to mitigate connection
problems that we are having with some clients.

Thank you very much for your help!

2015-03-27 19:38 GMT-03:00 Anthony Minessale <anthony.minessale at gmail.com>:

> You could probably use a proxy like kamailio or opensips to translate the
> websockets to UDP or TCP and pass it on to FS but FS itself cannot be
> modified to do what you want.
>
>
> On Fri, Mar 27, 2015 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I understand what you are trying to do.  The current sofia code has no
>> way to handle this currently, and we don't have plans to add this
>> functionality, because we can already do so with mod_verto.  If you want
>> this functionality in freeswitch, your options are using mod_verto, or a
>> huge amount of c code in mod_sofia that will be very error prone as it will
>> have to turn the state engine in that module on its head.
>>
>> On Mar 27, 2015, at 5:31 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>> wrote:
>>
>> Hey guys,
>>
>> Thank you all for the attention and patience to respond my questions.
>>
>> I understand that the ideal solution would be to use Verto, but that's
>> not practicable in our project right now.
>>
>> So, about the reconnected session, I am not sure if I made myself clear
>> about what is happening, and what I am trying to do.
>>
>>     WebRTC client  . . . nginx  . . . . FreeSWITCH
>>       (SIP.js)           proxy               |
>>          |                 |                 |
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              INVITE               |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |           Media Session           |
>>          |<=================================>|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          | CONNECTION FAIL |                 |
>>          |<-----XXXX------>|                 |
>>          |       Media continues to flow     |
>>          |<=================================>|
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              re-INVITE            |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          |                 |     INVITE      |
>>          |                 |<------XXX-------|
>>          |                 |                 | FS hang up call
>>          |         Media stop flowing        |
>>          |<==============XXXXX==============>|
>>
>> So, based on this scenario, when the Websocket connection to nginx fails,
>> we reconnect it, but since the media is going through other connections,
>> RTP over UDP, it is not affected.
>>
>> Now, with the new websocket connection in place the client is able to
>> send re-INVITEs and BYE to FS, and it is recognized as requests for the
>> session established using the first connection.
>>
>> The problem is that when FS tries to send a message to the client it
>> fails (NORMAL_TEMPORARY_FAILURE) and hangs up the call.
>>
>> Right now my question is:
>>  - How does FS know which connection it should use to send SIP messages
>> to the client?
>>
>> Thank you!
>>
>> 2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com>:
>>
>>> verto has its own JS client in tree.
>>>
>>> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>>>
>>> Hi Guys,
>>> What?s the best recommended client to connect to Verto ?
>>> Cheers,
>>> Abdul Hakeem
>>>
>>> *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 *Michael
>>> Jerris
>>> *Sent:* Friday, March 27, 2015 7:43 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Re-establish connection within a SIP
>>> session
>>>
>>> This is not a feature in any of the sip js stacks I know of, and I'm
>>> not quite sure how it would be implemented on top of sip.  As Brian said,
>>> this is a feature in verto.
>>>
>>>
>>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>>> wrote:
>>>
>>>
>>> Hello Brian,
>>> Thank you for the answer. We will consider using Verto in the future.
>>>
>>> Right now we will have to stick with WebRTC over SIP, we are using
>>> SIP.js for that.
>>>
>>> I ran some more tests and once the Websocket connection drops and is
>>> re-established,
>>> even if we send a re-INVITE, FS identifies it as belonging to the old
>>> call, and
>>>
>>> responds to it, after a while FS hangs up the call reporting a
>>> NORMAL_TEMPORARY_FAILURE.
>>> If the Websocket is not disconnected, I can see that FS sends an
>>> re-INVITE to the client after a while,
>>> so I guess that what is happening is that when FS tries to send this
>>> re-INVITE it realizes that the old connection
>>>
>>> was closed and hangs up the call.
>>> My question now is: Why FS does not update the connection information
>>> for the call once the re-INVITE from
>>> the new connection is received?
>>>
>>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
>>> Have you taken a look at Verto?
>>>
>>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>>> wrote:
>>>
>>> We have the following scenario: The session is established between
>>> WebRTC and FreeSWITCH using Websockets.
>>>
>>> Once the session is established, if the websocket connection drops the
>>> media continues to flow utilFreeSWITCH tries to send a re-INVITE to the
>>> client. At this point it realizes that the connection was closed and hangs
>>> up the call.
>>>
>>> Now, if the websocket connection drops and is re-established, would it
>>> be possible to inform FreeSWITCH that the new connection should be used for
>>> the previously established session?
>>>
>>> If the WebRTC client sends an INVITE message with the old session
>>> parameters, FreeSWITCH will be able to understand that it belongs to the
>>> old session?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/8aad39d7/attachment-0001.html 

From fernando at softov.com.br  Sat Mar 28 02:17:43 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Fri, 27 Mar 2015 19:17:43 -0400
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice calls
Message-ID: <CAKy26iPnx21d_8SuTpDa4yZJM+5iSgy6opBStoWsa7qVCk-6Mg@mail.gmail.com>

I thy to open pastebin

https://pastebin.freeswitch.org/23657, but it's without content.

I have a lot of problem with modem/dongle

Lets say i use

/dev/ttyU3.2 to audio device
/dev/ttyU3.3 to control device

Some times the modem power off and crash my usb controller

I tried using different power supplies

40A, 60A, but no sollution

I use FreeBSD, and when some dongle reset usbconfig stop reply, and i need
to restart the machine

When its crash, i can see this ttys remaining

/dev/ttyU3.3
/dev/ttyU3.3.init
/dev/ttyU3.3.lock

The others tty close normally, i think this crash because freewitch still
attached in ttyU3.3

Someone can help me?

-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150327/7357aced/attachment.html 

From ssinyagin at gmail.com  Sat Mar 28 03:13:41 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 28 Mar 2015 01:13:41 +0100
Subject: [Freeswitch-users] Strange effect with delay_echo against VAD
Message-ID: <CACOYK=iDrw8Qw3qFkz2OY8vb684=pTEUN-qRVgV-7b+nXApqDw@mail.gmail.com>

I got a strange effect, and help in understanding it would be appreciated.

My FreeSWITCH server registers at an ITSP, and forwards incoming calls
to "echo" or "delay_echo" applications, immediately after answering.

The ITSP uses VAD and comfort noise when it sends audio to my server.

If the call is sent to "echo" application, the calling party hears the
echo, and also it hears comfort noise.

If the call is sent to "delay_echo", with any delay (I tried 200 to
1000ms), the calling party hears only a loud noise.
Here's a recording on the calling side (my voice in the left ear)
http://www.k-open.com/s/0202eb6bfa655e47974325574ac47c7b.wav

Also if the call is sent to an IVR, and delay_echo is executed as an
IVR action, the caller hears the echo as expected.

The call is happening in PCMA.

I've got also packet captures, will share them on request, as they
contain some confidential information.


From ssinyagin at gmail.com  Sat Mar 28 03:58:42 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 28 Mar 2015 01:58:42 +0100
Subject: [Freeswitch-users] Strange effect with delay_echo against VAD
In-Reply-To: <CACOYK=iDrw8Qw3qFkz2OY8vb684=pTEUN-qRVgV-7b+nXApqDw@mail.gmail.com>
References: <CACOYK=iDrw8Qw3qFkz2OY8vb684=pTEUN-qRVgV-7b+nXApqDw@mail.gmail.com>
Message-ID: <CACOYK=iRTZAEnyF+LkAp+M7BgkWxteVE1i30_3V+i9-Ntj5Z1g@mail.gmail.com>

the problem is solved with
<param name="suppress-cng" value="true"/>
in the SIP profile, so that FreeSWITCH stops offering CN in SDP.
Still it will be interesting to know what caused this problem.




On Sat, Mar 28, 2015 at 1:13 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> I got a strange effect, and help in understanding it would be appreciated.
>
> My FreeSWITCH server registers at an ITSP, and forwards incoming calls
> to "echo" or "delay_echo" applications, immediately after answering.
>
> The ITSP uses VAD and comfort noise when it sends audio to my server.
>
> If the call is sent to "echo" application, the calling party hears the
> echo, and also it hears comfort noise.
>
> If the call is sent to "delay_echo", with any delay (I tried 200 to
> 1000ms), the calling party hears only a loud noise.
> Here's a recording on the calling side (my voice in the left ear)
> http://www.k-open.com/s/0202eb6bfa655e47974325574ac47c7b.wav
>
> Also if the call is sent to an IVR, and delay_echo is executed as an
> IVR action, the caller hears the echo as expected.
>
> The call is happening in PCMA.
>
> I've got also packet captures, will share them on request, as they
> contain some confidential information.


From s.safarov at gmail.com  Sat Mar 28 08:06:59 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 28 Mar 2015 08:06:59 +0300
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice
	calls
In-Reply-To: <CAKy26iPnx21d_8SuTpDa4yZJM+5iSgy6opBStoWsa7qVCk-6Mg@mail.gmail.com>
References: <CAKy26iPnx21d_8SuTpDa4yZJM+5iSgy6opBStoWsa7qVCk-6Mg@mail.gmail.com>
Message-ID: <CAHtxdDcj9BgwemLnoCybLg62sKnUqnnMeYz9M7h0UMKYSRKyXg@mail.gmail.com>

- Enable coredump http://blog.urdada.net/2007/12/31/71/
- Find it after next crash
- Analyze coredump

On Sat, Mar 28, 2015 at 2:17 AM, Luiz Fernando Softov <
fernando at softov.com.br> wrote:

>
> I thy to open pastebin
>
> https://pastebin.freeswitch.org/23657, but it's without content.
>
> I have a lot of problem with modem/dongle
>
> Lets say i use
>
> /dev/ttyU3.2 to audio device
> /dev/ttyU3.3 to control device
>
> Some times the modem power off and crash my usb controller
>
> I tried using different power supplies
>
> 40A, 60A, but no sollution
>
> I use FreeBSD, and when some dongle reset usbconfig stop reply, and i need
> to restart the machine
>
> When its crash, i can see this ttys remaining
>
> /dev/ttyU3.3
> /dev/ttyU3.3.init
> /dev/ttyU3.3.lock
>
> The others tty close normally, i think this crash because freewitch still
> attached in ttyU3.3
>
> Someone can help me?
>
> --
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150328/4346d8e2/attachment.html 

From s.safarov at gmail.com  Sat Mar 28 08:14:10 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 28 Mar 2015 08:14:10 +0300
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice
	calls
In-Reply-To: <CAHtxdDcj9BgwemLnoCybLg62sKnUqnnMeYz9M7h0UMKYSRKyXg@mail.gmail.com>
References: <CAKy26iPnx21d_8SuTpDa4yZJM+5iSgy6opBStoWsa7qVCk-6Mg@mail.gmail.com>
	<CAHtxdDcj9BgwemLnoCybLg62sKnUqnnMeYz9M7h0UMKYSRKyXg@mail.gmail.com>
Message-ID: <CAHtxdDdyuXXNRWTG7EgNC00hn15_8HzuNPZ9BMQ5aS8rirpi+Q@mail.gmail.com>

Find what is calling USB modem power off and disable it.
No poweroff, no error.

On Sat, Mar 28, 2015 at 8:06 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> - Enable coredump http://blog.urdada.net/2007/12/31/71/
> - Find it after next crash
> - Analyze coredump
>
> On Sat, Mar 28, 2015 at 2:17 AM, Luiz Fernando Softov <
> fernando at softov.com.br> wrote:
>
>>
>> I thy to open pastebin
>>
>> https://pastebin.freeswitch.org/23657, but it's without content.
>>
>> I have a lot of problem with modem/dongle
>>
>> Lets say i use
>>
>> /dev/ttyU3.2 to audio device
>> /dev/ttyU3.3 to control device
>>
>> Some times the modem power off and crash my usb controller
>>
>> I tried using different power supplies
>>
>> 40A, 60A, but no sollution
>>
>> I use FreeBSD, and when some dongle reset usbconfig stop reply, and i
>> need to restart the machine
>>
>> When its crash, i can see this ttys remaining
>>
>> /dev/ttyU3.3
>> /dev/ttyU3.3.init
>> /dev/ttyU3.3.lock
>>
>> The others tty close normally, i think this crash because freewitch still
>> attached in ttyU3.3
>>
>> Someone can help me?
>>
>> --
>> Luiz Fernando Softov
>> http://www.softov.com.br
>> fernando at softov.com.br
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150328/ca21ed0a/attachment.html 

From ben at langfeld.co.uk  Sat Mar 28 20:39:25 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sat, 28 Mar 2015 14:39:25 -0300
Subject: [Freeswitch-users] Re-establish connection within a SIP session
In-Reply-To: <CAKbxfG95zTRF65ohsTmjJErdXBszkoSUux_qHii==cBeyd6T=A@mail.gmail.com>
References: <CAJmiN57fDizJzaHO-v4qSU8sKdam4QipcA08=JOMfQxBpHQ++Q@mail.gmail.com>
	<CAEJMVkCvWY6qDfv6ZQ+zovbMejeyNE0Yz3LRjraLbHyGghW1zQ@mail.gmail.com>
	<CAJmiN54G1VM1Ws7qJYNL1_WKMmPS=aw2_KgQQvLhG0wK5N=5-w@mail.gmail.com>
	<7D25D9B0-368F-4C0B-95ED-091E281A31C5@jerris.com>
	<!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAACPj1jhcpwlJgmHq4L3eoxoBAAAAAA==@gmail.com>
	<90EE19E9-2B2F-45FF-8523-7CB8F9797872@jerris.com>
	<CAJmiN56Exo=dmsMT8pUF7OsnEtoCuaPR20nKwn_WgwGEONjdbg@mail.gmail.com>
	<525F35D0-E7C5-4891-B817-365F523D32A5@jerris.com>
	<CAKbxfG95zTRF65ohsTmjJErdXBszkoSUux_qHii==cBeyd6T=A@mail.gmail.com>
Message-ID: <CAAyX+KFiiDqC9YgHoAkkAT3QL_Xd1X+AHugizxOLG+6CCWCpLA@mail.gmail.com>

Anthony, there was a question earlier in the thread about how Websockets
and TCP differ in this scenario. Is it not the case that they are
conceptually equivalent here? Would this issue also manifest with TCP, or
are the code paths significantly different between Websocket and TCP?

On 27 March 2015 at 19:38, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

> You could probably use a proxy like kamailio or opensips to translate the
> websockets to UDP or TCP and pass it on to FS but FS itself cannot be
> modified to do what you want.
>
>
> On Fri, Mar 27, 2015 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I understand what you are trying to do.  The current sofia code has no
>> way to handle this currently, and we don't have plans to add this
>> functionality, because we can already do so with mod_verto.  If you want
>> this functionality in freeswitch, your options are using mod_verto, or a
>> huge amount of c code in mod_sofia that will be very error prone as it will
>> have to turn the state engine in that module on its head.
>>
>> On Mar 27, 2015, at 5:31 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>> wrote:
>>
>> Hey guys,
>>
>> Thank you all for the attention and patience to respond my questions.
>>
>> I understand that the ideal solution would be to use Verto, but that's
>> not practicable in our project right now.
>>
>> So, about the reconnected session, I am not sure if I made myself clear
>> about what is happening, and what I am trying to do.
>>
>>     WebRTC client  . . . nginx  . . . . FreeSWITCH
>>       (SIP.js)           proxy               |
>>          |                 |                 |
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              INVITE               |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |           Media Session           |
>>          |<=================================>|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          | CONNECTION FAIL |                 |
>>          |<-----XXXX------>|                 |
>>          |       Media continues to flow     |
>>          |<=================================>|
>>          |     CONNECT     |                 |
>>          |---------------->|                 |
>>          |              re-INVITE            |
>>          |---------------------------------->|
>>          |                OK                 |
>>          |<----------------------------------|
>>          |                ACK                |
>>          |---------------------------------->|
>>          |                 .                 |
>>          |                 .                 |
>>          |                 .                 |
>>          |                 |     INVITE      |
>>          |                 |<------XXX-------|
>>          |                 |                 | FS hang up call
>>          |         Media stop flowing        |
>>          |<==============XXXXX==============>|
>>
>> So, based on this scenario, when the Websocket connection to nginx fails,
>> we reconnect it, but since the media is going through other connections,
>> RTP over UDP, it is not affected.
>>
>> Now, with the new websocket connection in place the client is able to
>> send re-INVITEs and BYE to FS, and it is recognized as requests for the
>> session established using the first connection.
>>
>> The problem is that when FS tries to send a message to the client it
>> fails (NORMAL_TEMPORARY_FAILURE) and hangs up the call.
>>
>> Right now my question is:
>>  - How does FS know which connection it should use to send SIP messages
>> to the client?
>>
>> Thank you!
>>
>> 2015-03-27 17:15 GMT-03:00 Michael Jerris <mike at jerris.com>:
>>
>>> verto has its own JS client in tree.
>>>
>>> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>>>
>>> Hi Guys,
>>> What?s the best recommended client to connect to Verto ?
>>> Cheers,
>>> Abdul Hakeem
>>>
>>> *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 *Michael
>>> Jerris
>>> *Sent:* Friday, March 27, 2015 7:43 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Re-establish connection within a SIP
>>> session
>>>
>>> This is not a feature in any of the sip js stacks I know of, and I'm
>>> not quite sure how it would be implemented on top of sip.  As Brian said,
>>> this is a feature in verto.
>>>
>>>
>>> On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>>> wrote:
>>>
>>>
>>> Hello Brian,
>>> Thank you for the answer. We will consider using Verto in the future.
>>>
>>> Right now we will have to stick with WebRTC over SIP, we are using
>>> SIP.js for that.
>>>
>>> I ran some more tests and once the Websocket connection drops and is
>>> re-established,
>>> even if we send a re-INVITE, FS identifies it as belonging to the old
>>> call, and
>>>
>>> responds to it, after a while FS hangs up the call reporting a
>>> NORMAL_TEMPORARY_FAILURE.
>>> If the Websocket is not disconnected, I can see that FS sends an
>>> re-INVITE to the client after a while,
>>> so I guess that what is happening is that when FS tries to send this
>>> re-INVITE it realizes that the old connection
>>>
>>> was closed and hangs up the call.
>>> My question now is: Why FS does not update the connection information
>>> for the call once the re-INVITE from
>>> the new connection is received?
>>>
>>> 2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
>>> Have you taken a look at Verto?
>>>
>>> On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com>
>>> wrote:
>>>
>>> We have the following scenario: The session is established between
>>> WebRTC and FreeSWITCH using Websockets.
>>>
>>> Once the session is established, if the websocket connection drops the
>>> media continues to flow utilFreeSWITCH tries to send a re-INVITE to the
>>> client. At this point it realizes that the connection was closed and hangs
>>> up the call.
>>>
>>> Now, if the websocket connection drops and is re-established, would it
>>> be possible to inform FreeSWITCH that the new connection should be used for
>>> the previously established session?
>>>
>>> If the WebRTC client sends an INVITE message with the old session
>>> parameters, FreeSWITCH will be able to understand that it belongs to the
>>> old session?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150328/80d7aa42/attachment-0001.html 

From fernando at softov.com.br  Sun Mar 29 03:05:43 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Sat, 28 Mar 2015 19:05:43 -0400
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice calls
Message-ID: <CAKy26iMZLhqo5PCSYP63y+XdfF9XgTm+H5bTjGGzx9_YMdxuHA@mail.gmail.com>

Hi


***** - Enable coredump http://blog.urdada.net/2007/12/31/71/
***** - Find it after next crash
***** - Analyze coredump

First, when i say crash, it's a panic in USB, not a crach of
daemon/executable. This is because Freeswitch/GsmOpen is attached in TTY
and lock the TTY, blocking other processes to manage USB (tty).

***** Find what is calling USB modem power off and disable it.***** No
poweroff, no error.

Some modems power off and power on again, i try to find what is
causing this error, without success. Maybe some errors in power suply
or modem just reset, because i try it with 4, 10, 20 and 30 modems,
and the problem persists.
In my search, google and other forums, i found someone talking about
modem resetting for own.

Sometimes the user wan't to change de SIM in modem, and remove it and
put it again, without make a dettach in my interface/system.

Or if he not remove the modem and just change de SIM (o.O), its a
request of many users of my system using E303 or other modems.

I identify this, because i'm attached in ttyU0.0, ang get event im my
source or a CME-ERROR 10, if modem just changed SIM, then i reset the
modem with AT command ^RESET.


I'm using mod_xml_curl to get interfaces info, and ESL events to
manage this... Then i make a "gsm remove xxxx" and when modem arrive i
make a "gsm reload".

This way not affect the calls in progress.


I talk with Shlomi Agiv, and apply some functions of his patch, now
i'm just making tests....


Now i'm having some problems, just when i call a "gsm remove" in ESL,
its stop to reply, even when i send other command, like "sofia
status". It's the same when i send command with fs_cli...


Before ask, i make some debugs, change source, put log and other things..

I ask, because some times other people have the same problem!

Thanks for the reply...



-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150328/5d501256/attachment.html 

From fernando at softov.com.br  Sun Mar 29 07:02:04 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Sat, 28 Mar 2015 23:02:04 -0400
Subject: [Freeswitch-users] Transfering Attended Call
Message-ID: <CAKy26iOqxXcXGNa6YxPnwgA4pG0b2oTyA2+ZWA4P-VvKB-nx7A@mail.gmail.com>

Hi, i'm executing "att_xfer"


https://freeswitch.org/confluence/display/FREESWITCH/Attended+Transfer

So...

A Bridge to B, and i make a att_xfer using B uuid, to C...

When B in call with C.

A gets silence, i try to play a music with playback, and its work, but i
can't execute anything until music is played.

And when B digits *, to start with transfer, i try to play a music to A
uuid, but i can't receive other DTMF events from B...

Is there a way to play a music to A, when i make a att_xfer?

Then when att_xfer is done, de music stop to play....

I try to use "fifo" too, but is the same result, i can't get B leg DTMF
events after...

i'm not using lock in this Events...

I'm just want to play some music or sound to A, until he is waiting for B
or C to take the call

-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150328/bb68032a/attachment.html 

From s.safarov at gmail.com  Sun Mar 29 10:07:45 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 29 Mar 2015 09:07:45 +0300
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice
	calls
In-Reply-To: <CAKy26iMZLhqo5PCSYP63y+XdfF9XgTm+H5bTjGGzx9_YMdxuHA@mail.gmail.com>
References: <CAKy26iMZLhqo5PCSYP63y+XdfF9XgTm+H5bTjGGzx9_YMdxuHA@mail.gmail.com>
Message-ID: <CAHtxdDdJm-Wyhb9BA+AbSMPWRVxzwF44s=DR2ASx8Y9tpGajig@mail.gmail.com>

I thinking following device cannot be removed
/dev/ttyU3.3
/dev/ttyU3.3.init
/dev/ttyU3.3.lock
It is exist because GSM device exist (/dev/ttyU3.3) in the system and must
be controlled (restarted, reinited, disabled, /dev/ttyU3.3.init) and device
is opened (/dev/ttyU3.3.lock)

On Sun, Mar 29, 2015 at 2:05 AM, Luiz Fernando Softov <
fernando at softov.com.br> wrote:

> Hi
>
>
> ***** - Enable coredump http://blog.urdada.net/2007/12/31/71/
> ***** - Find it after next crash
> ***** - Analyze coredump
>
> First, when i say crash, it's a panic in USB, not a crach of
> daemon/executable. This is because Freeswitch/GsmOpen is attached in TTY
> and lock the TTY, blocking other processes to manage USB (tty).
>
> ***** Find what is calling USB modem power off and disable it.***** No poweroff, no error.
>
> Some modems power off and power on again, i try to find what is causing this error, without success. Maybe some errors in power suply or modem just reset, because i try it with 4, 10, 20 and 30 modems, and the problem persists.
> In my search, google and other forums, i found someone talking about modem resetting for own.
>
> Sometimes the user wan't to change de SIM in modem, and remove it and put it again, without make a dettach in my interface/system.
>
> Or if he not remove the modem and just change de SIM (o.O), its a request of many users of my system using E303 or other modems.
>
> I identify this, because i'm attached in ttyU0.0, ang get event im my source or a CME-ERROR 10, if modem just changed SIM, then i reset the modem with AT command ^RESET.
>
>
> I'm using mod_xml_curl to get interfaces info, and ESL events to manage this... Then i make a "gsm remove xxxx" and when modem arrive i make a "gsm reload".
>
> This way not affect the calls in progress.
>
>
> I talk with Shlomi Agiv, and apply some functions of his patch, now i'm just making tests....
>
>
> Now i'm having some problems, just when i call a "gsm remove" in ESL, its stop to reply, even when i send other command, like "sofia status". It's the same when i send command with fs_cli...
>
>
> Before ask, i make some debugs, change source, put log and other things..
>
> I ask, because some times other people have the same problem!
>
> Thanks for the reply...
>
>
>
> --
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150329/f923af80/attachment.html 

From shabbirabbasi92 at gmail.com  Sun Mar 29 10:52:59 2015
From: shabbirabbasi92 at gmail.com (Shabbir abbasi)
Date: Sun, 29 Mar 2015 11:52:59 +0500
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice
	calls
In-Reply-To: <CAHtxdDdJm-Wyhb9BA+AbSMPWRVxzwF44s=DR2ASx8Y9tpGajig@mail.gmail.com>
References: <CAKy26iMZLhqo5PCSYP63y+XdfF9XgTm+H5bTjGGzx9_YMdxuHA@mail.gmail.com>
	<CAHtxdDdJm-Wyhb9BA+AbSMPWRVxzwF44s=DR2ASx8Y9tpGajig@mail.gmail.com>
Message-ID: <CAGpk5BS-dW50qqNAxjmhtSe1VMV2otZvxE9Xn40pege3_jMrFA@mail.gmail.com>

today i tried to test gsmopen it has many problums, gsmopen has lack of
reconnect to modem and modem is marked as dead instead of disconnect and
try to reconnect,
 i have also tested asterisk  module chan_dongle which works great(it has
function to test modem every 10 secon with AT command andif no reply then
disconnect it and reconnect again ) and i have no disconnection with that
module same cpu same hardware same OS, but i want to test  module gsmopen
for voice quality,  gsmopen needs some patches to work fine
 , if anyone make patches it will be great

On Sun, Mar 29, 2015 at 11:07 AM, Sergey Safarov <s.safarov at gmail.com>
wrote:

> I thinking following device cannot be removed
> /dev/ttyU3.3
> /dev/ttyU3.3.init
> /dev/ttyU3.3.lock
> It is exist because GSM device exist (/dev/ttyU3.3) in the system and must
> be controlled (restarted, reinited, disabled, /dev/ttyU3.3.init) and device
> is opened (/dev/ttyU3.3.lock)
>
> On Sun, Mar 29, 2015 at 2:05 AM, Luiz Fernando Softov <
> fernando at softov.com.br> wrote:
>
>> Hi
>>
>>
>> ***** - Enable coredump http://blog.urdada.net/2007/12/31/71/
>> ***** - Find it after next crash
>> ***** - Analyze coredump
>>
>> First, when i say crash, it's a panic in USB, not a crach of
>> daemon/executable. This is because Freeswitch/GsmOpen is attached in TTY
>> and lock the TTY, blocking other processes to manage USB (tty).
>>
>> ***** Find what is calling USB modem power off and disable it.***** No poweroff, no error.
>>
>> Some modems power off and power on again, i try to find what is causing this error, without success. Maybe some errors in power suply or modem just reset, because i try it with 4, 10, 20 and 30 modems, and the problem persists.
>> In my search, google and other forums, i found someone talking about modem resetting for own.
>>
>> Sometimes the user wan't to change de SIM in modem, and remove it and put it again, without make a dettach in my interface/system.
>>
>> Or if he not remove the modem and just change de SIM (o.O), its a request of many users of my system using E303 or other modems.
>>
>> I identify this, because i'm attached in ttyU0.0, ang get event im my source or a CME-ERROR 10, if modem just changed SIM, then i reset the modem with AT command ^RESET.
>>
>>
>> I'm using mod_xml_curl to get interfaces info, and ESL events to manage this... Then i make a "gsm remove xxxx" and when modem arrive i make a "gsm reload".
>>
>> This way not affect the calls in progress.
>>
>>
>> I talk with Shlomi Agiv, and apply some functions of his patch, now i'm just making tests....
>>
>>
>> Now i'm having some problems, just when i call a "gsm remove" in ESL, its stop to reply, even when i send other command, like "sofia status". It's the same when i send command with fs_cli...
>>
>>
>> Before ask, i make some debugs, change source, put log and other things..
>>
>> I ask, because some times other people have the same problem!
>>
>> Thanks for the reply...
>>
>>
>>
>> --
>> Luiz Fernando Softov
>> http://www.softov.com.br
>> fernando at softov.com.br
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150329/39e84440/attachment-0001.html 

From alhakeem at gmail.com  Sun Mar 29 16:03:03 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Sun, 29 Mar 2015 13:03:03 +0100
Subject: [Freeswitch-users] Sofia.h limits ?
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAIn3dvZm/1ZKpcs80qJj93IBAAAAAA==@gmail.com>

Hello Guys,
Could someone shed lights if  these lines are hard codec limits and suggestions
for alternative parameters ?
#define SOFIA_MAX_AL 100
#define MAX_RTP 50
#define SOFIA_MAX_MSG_QUEUE 64

Many thanks,
Abdul Hakeem





From mike at jerris.com  Sun Mar 29 21:28:09 2015
From: mike at jerris.com (Michael Jerris)
Date: Sun, 29 Mar 2015 13:28:09 -0400
Subject: [Freeswitch-users] Sofia.h limits ?
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAIn3dvZm/1ZKpcs80qJj93IBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAIn3dvZm/1ZKpcs80qJj93IBAAAAAA==@gmail.com>
Message-ID: <79D8FB71-A127-4FF5-902B-FEB589B6C5F5@jerris.com>

These are hardcoded constants, they should not have to be changed.  Is there any reason you are thinking they should be?


> On Mar 29, 2015, at 8:03 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:
> 
> Hello Guys,
> Could someone shed lights if  these lines are hard codec limits and suggestions
> for alternative parameters ?
> #define SOFIA_MAX_AL 100
> #define MAX_RTP 50
> #define SOFIA_MAX_MSG_QUEUE 64
> 
> Many thanks,
> Abdul Hakeem



From switcherfree at gmail.com  Sun Mar 29 23:29:24 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Sun, 29 Mar 2015 12:29:24 -0700
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAEJMVkCUP3SU9x4_fjxmtGagf5F+zh7J6Z8zX0Fh3w-pe9wxXQ@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
	<CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
	<CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>
	<CAEJMVkCUP3SU9x4_fjxmtGagf5F+zh7J6Z8zX0Fh3w-pe9wxXQ@mail.gmail.com>
Message-ID: <CAM6Un21JH0z=3D_cvRYpV2dgYXq7H-Z_mFC1gZQ9M_S_FyN3kg@mail.gmail.com>

Hi Brian,
Thank you for pointing to better instructions. Still running into
problems though. Running 'yum install --nogpgcheck
freeswitch-config-vanilla' gives the following errors:

--> Finished Dependency Resolution
Error: Package: freeswitch-format-portaudio-stream-1.4.15-1.el6.x86_64
(freeswitch)
           Requires: libportaudio.so.2()(64bit)
Error: Package: freeswitch-application-enum-1.4.15-1.el6.x86_64 (freeswitch)
           Requires: libldns.so.1()(64bit)
 You could try using --skip-broken to work around the problem
 You could try running: rpm -Va --nofiles --nodigest

-----
Appreciate any further suggestions.
Thanks,
Andy


On Fri, Mar 27, 2015 at 12:00 PM, Brian West <brian at freeswitch.org> wrote:
> Never mind, Its not a bug, Confluence doesn't have the correct process:
> Wiki does
> https://wiki.freeswitch.org/wiki/Installation_Guide#YUM_Based_Installation
>
>
>
> On Fri, Mar 27, 2015 at 1:37 PM, Brian West <brian at freeswitch.org> wrote:
>>
>> Can you file a JIRA for us to investigate this?
>>
>> On Fri, Mar 27, 2015 at 12:49 PM, Free Switcher <switcherfree at gmail.com>
>> wrote:
>>>
>>> Hello Brian,
>>> Thanks for your response. I did not build from source. I just
>>> downloaded/installed pre-built packages using yum. On my CentOS 6.6 machine,
>>> it looks like the freeswitch binary is installed under /usr/bin with configs
>>> under /etc/freeswitch.
>>>
>>> Thanks,
>>> Andy
>>>
>>>
>>> On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> If you downloaded and built the source its all going to be in
>>>> /usr/local/freeswitch unless you configured it with the same configure args
>>>> as the previous package install.
>>>>
>>>> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu
>>>> <mishehu at freeswitch.org> wrote:
>>>>>
>>>>> Depending on what repo you got those packages from, they could be
>>>>> seriously old packages.  Send back the output of:
>>>>>
>>>>> rpm -qa |grep -i freeswitch
>>>>>
>>>>> Chances are you only installed the base package.  you can also do `yum
>>>>> search freeswitch` to see if other packages are offered. After you
>>>>> provide the output from the above command I can comment on the age of
>>>>> the packages you installed.
>>>>>
>>>>> --
>>>>> Yossi Neiman
>>>>>
>>>>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>>>>
>>>>> > Hello,
>>>>> > I'm trying to create a new installation of freeswitch. I'm running
>>>>> > CentOS 6.6 and picked the easy path to install pre-built binaries
>>>>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
>>>>> > noticed that mod_conference.so wasn't there. Also noticed that no
>>>>> > config files were installed in /etc/freeswitch. Is this expected?
>>>>> >
>>>>> > I downloaded configs from the latest stash repository and I was then
>>>>> > able to start freeswitch with vanilla config. I see several startup
>>>>> > messages about missing mods. Which modules should I expect to see as
>>>>> > part of the pre-built binaries? How/where does one download
>>>>> > additional
>>>>> > mods? Here is what I have in the /usr/lib64/freeswitch :
>>>>> >
>>>>> > mod_cdr_csv.so
>>>>> > mod_commands.so
>>>>> > mod_console.so
>>>>> > mod_dialplan_directory.so
>>>>> > mod_dialplan_xml.so
>>>>> > mod_dptools.so
>>>>> > mod_event_socket.so
>>>>> > mod_logfile.so
>>>>> > mod_loopback.so
>>>>> > mod_native_file.so
>>>>> > mod_sndfile.so
>>>>> > mod_sofia.so
>>>>> > mod_spandsp.so
>>>>> > mod_syslog.so
>>>>> > mod_tone_stream.so
>>>>> > mod_xml_rpc.so
>>>>> >
>>>>> >
>>>>> > Any help is appreciated.
>>>>> > Thanks,
>>>>> > Andy
>>>>> >
>>>>> >
>>>>> > _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://confluence.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Brian West
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> Twitter: @FreeSWITCH , @briankwest
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> ClueCon 2015 Call for Speakers | Register TODAY!
>>>>
>>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>>>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>> Brian West
>> brian at freeswitch.org
>>
>>
>> Twitter: @FreeSWITCH , @briankwest
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers | Register TODAY!
>>
>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>
>
>
>
> --
>
> Brian West
> brian at freeswitch.org
>
>
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers | Register TODAY!
>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From s.safarov at gmail.com  Sun Mar 29 23:51:52 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 29 Mar 2015 22:51:52 +0300
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAM6Un21JH0z=3D_cvRYpV2dgYXq7H-Z_mFC1gZQ9M_S_FyN3kg@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
	<CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
	<CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>
	<CAEJMVkCUP3SU9x4_fjxmtGagf5F+zh7J6Z8zX0Fh3w-pe9wxXQ@mail.gmail.com>
	<CAM6Un21JH0z=3D_cvRYpV2dgYXq7H-Z_mFC1gZQ9M_S_FyN3kg@mail.gmail.com>
Message-ID: <CAHtxdDfO0am-9ZnMXHOt0SoUf+0E-n8+-JUvqKnO6b=EG95jgQ@mail.gmail.com>

[root at localhost ~]# find /usr | grep libldns
/usr/lib64/libldns.so.1
/usr/lib64/libldns.so.1.6.16
/usr/lib64/libldns.so
[root at localhost ~]# rpm -qf /usr/lib64/libldns.so
ldns-devel-1.6.16-7.el7.x86_64
[root at localhost ~]#


On Sun, Mar 29, 2015 at 10:29 PM, Free Switcher <switcherfree at gmail.com>
wrote:

> Hi Brian,
> Thank you for pointing to better instructions. Still running into
> problems though. Running 'yum install --nogpgcheck
> freeswitch-config-vanilla' gives the following errors:
>
> --> Finished Dependency Resolution
> Error: Package: freeswitch-format-portaudio-stream-1.4.15-1.el6.x86_64
> (freeswitch)
>            Requires: libportaudio.so.2()(64bit)
> Error: Package: freeswitch-application-enum-1.4.15-1.el6.x86_64
> (freeswitch)
>            Requires: libldns.so.1()(64bit)
>  You could try using --skip-broken to work around the problem
>  You could try running: rpm -Va --nofiles --nodigest
>
> -----
> Appreciate any further suggestions.
> Thanks,
> Andy
>
>
> On Fri, Mar 27, 2015 at 12:00 PM, Brian West <brian at freeswitch.org> wrote:
> > Never mind, Its not a bug, Confluence doesn't have the correct process:
> > Wiki does
> >
> https://wiki.freeswitch.org/wiki/Installation_Guide#YUM_Based_Installation
> >
> >
> >
> > On Fri, Mar 27, 2015 at 1:37 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>
> >> Can you file a JIRA for us to investigate this?
> >>
> >> On Fri, Mar 27, 2015 at 12:49 PM, Free Switcher <switcherfree at gmail.com
> >
> >> wrote:
> >>>
> >>> Hello Brian,
> >>> Thanks for your response. I did not build from source. I just
> >>> downloaded/installed pre-built packages using yum. On my CentOS 6.6
> machine,
> >>> it looks like the freeswitch binary is installed under /usr/bin with
> configs
> >>> under /etc/freeswitch.
> >>>
> >>> Thanks,
> >>> Andy
> >>>
> >>>
> >>> On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org>
> wrote:
> >>>>
> >>>> If you downloaded and built the source its all going to be in
> >>>> /usr/local/freeswitch unless you configured it with the same
> configure args
> >>>> as the previous package install.
> >>>>
> >>>> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu
> >>>> <mishehu at freeswitch.org> wrote:
> >>>>>
> >>>>> Depending on what repo you got those packages from, they could be
> >>>>> seriously old packages.  Send back the output of:
> >>>>>
> >>>>> rpm -qa |grep -i freeswitch
> >>>>>
> >>>>> Chances are you only installed the base package.  you can also do
> `yum
> >>>>> search freeswitch` to see if other packages are offered. After you
> >>>>> provide the output from the above command I can comment on the age of
> >>>>> the packages you installed.
> >>>>>
> >>>>> --
> >>>>> Yossi Neiman
> >>>>>
> >>>>> On 03/27/2015 01:12 AM, Free Switcher wrote:
> >>>>>
> >>>>> > Hello,
> >>>>> > I'm trying to create a new installation of freeswitch. I'm running
> >>>>> > CentOS 6.6 and picked the easy path to install pre-built binaries
> >>>>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
> >>>>> > noticed that mod_conference.so wasn't there. Also noticed that no
> >>>>> > config files were installed in /etc/freeswitch. Is this expected?
> >>>>> >
> >>>>> > I downloaded configs from the latest stash repository and I was
> then
> >>>>> > able to start freeswitch with vanilla config. I see several startup
> >>>>> > messages about missing mods. Which modules should I expect to see
> as
> >>>>> > part of the pre-built binaries? How/where does one download
> >>>>> > additional
> >>>>> > mods? Here is what I have in the /usr/lib64/freeswitch :
> >>>>> >
> >>>>> > mod_cdr_csv.so
> >>>>> > mod_commands.so
> >>>>> > mod_console.so
> >>>>> > mod_dialplan_directory.so
> >>>>> > mod_dialplan_xml.so
> >>>>> > mod_dptools.so
> >>>>> > mod_event_socket.so
> >>>>> > mod_logfile.so
> >>>>> > mod_loopback.so
> >>>>> > mod_native_file.so
> >>>>> > mod_sndfile.so
> >>>>> > mod_sofia.so
> >>>>> > mod_spandsp.so
> >>>>> > mod_syslog.so
> >>>>> > mod_tone_stream.so
> >>>>> > mod_xml_rpc.so
> >>>>> >
> >>>>> >
> >>>>> > Any help is appreciated.
> >>>>> > Thanks,
> >>>>> > Andy
> >>>>> >
> >>>>> >
> >>>>> >
> _________________________________________________________________________
> >>>>> > Professional FreeSWITCH Consulting Services:
> >>>>> > consulting at freeswitch.org
> >>>>> > http://www.freeswitchsolutions.com
> >>>>> >
> >>>>> > Official FreeSWITCH Sites
> >>>>> > http://www.freeswitch.org
> >>>>> > http://confluence.freeswitch.org
> >>>>> > http://www.cluecon.com
> >>>>> >
> >>>>> > FreeSWITCH-users mailing list
> >>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> >
> >>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> > http://www.freeswitch.org
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>>
> >>>> Brian West
> >>>> brian at freeswitch.org
> >>>>
> >>>>
> >>>> Twitter: @FreeSWITCH , @briankwest
> >>>> http://www.freeswitchbook.com
> >>>> http://www.freeswitchcookbook.com
> >>>>
> >>>> ClueCon 2015 Call for Speakers | Register TODAY!
> >>>>
> >>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> >>>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> >>>>
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >>
> >> Brian West
> >> brian at freeswitch.org
> >>
> >>
> >> Twitter: @FreeSWITCH , @briankwest
> >> http://www.freeswitchbook.com
> >> http://www.freeswitchcookbook.com
> >>
> >> ClueCon 2015 Call for Speakers | Register TODAY!
> >>
> >> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> >> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> >
> >
> >
> >
> > --
> >
> > Brian West
> > brian at freeswitch.org
> >
> >
> > Twitter: @FreeSWITCH , @briankwest
> > http://www.freeswitchbook.com
> > http://www.freeswitchcookbook.com
> >
> > ClueCon 2015 Call for Speakers | Register TODAY!
> >
> > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150329/b56719e4/attachment-0001.html 

From switcherfree at gmail.com  Mon Mar 30 10:32:24 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Sun, 29 Mar 2015 23:32:24 -0700
Subject: [Freeswitch-users] Missing modules and configs in fresh install
In-Reply-To: <CAM6Un21JH0z=3D_cvRYpV2dgYXq7H-Z_mFC1gZQ9M_S_FyN3kg@mail.gmail.com>
References: <CAM6Un22CwUJ4VvT0BKq2KydRVdRNtzodtceiZfWDKWBsxi6-Fg@mail.gmail.com>
	<551558D0.7090803@freeswitch.org>
	<CAEJMVkB6TUq992qA4631Pk5-pefiGy3mEQ-HCqAk2pKPxSQ2ow@mail.gmail.com>
	<CAM6Un22DsXKwcGY-1o47vVXDSo399e96-NC_Wnd=8Ks6-maLoA@mail.gmail.com>
	<CAEJMVkD+XpVh7pMMLuWQ1MkrM35nTS1m=4r_fE2ybajjk9myjA@mail.gmail.com>
	<CAEJMVkCUP3SU9x4_fjxmtGagf5F+zh7J6Z8zX0Fh3w-pe9wxXQ@mail.gmail.com>
	<CAM6Un21JH0z=3D_cvRYpV2dgYXq7H-Z_mFC1gZQ9M_S_FyN3kg@mail.gmail.com>
Message-ID: <CAM6Un20v7BzzHX=onLpWw9g=grC=hS_OTp_8dL5bLcAEoVH5dQ@mail.gmail.com>

Never mind. Adding epel repo resolved all dependencies. But portaudio
looks fairly old (portaudio-19-9.el6.x86_64). Not sure if a more
recent version is required & where to find one.

On Sun, Mar 29, 2015 at 12:29 PM, Free Switcher <switcherfree at gmail.com> wrote:
> Hi Brian,
> Thank you for pointing to better instructions. Still running into
> problems though. Running 'yum install --nogpgcheck
> freeswitch-config-vanilla' gives the following errors:
>
> --> Finished Dependency Resolution
> Error: Package: freeswitch-format-portaudio-stream-1.4.15-1.el6.x86_64
> (freeswitch)
>            Requires: libportaudio.so.2()(64bit)
> Error: Package: freeswitch-application-enum-1.4.15-1.el6.x86_64 (freeswitch)
>            Requires: libldns.so.1()(64bit)
>  You could try using --skip-broken to work around the problem
>  You could try running: rpm -Va --nofiles --nodigest
>
> -----
> Appreciate any further suggestions.
> Thanks,
> Andy
>
>
> On Fri, Mar 27, 2015 at 12:00 PM, Brian West <brian at freeswitch.org> wrote:
>> Never mind, Its not a bug, Confluence doesn't have the correct process:
>> Wiki does
>> https://wiki.freeswitch.org/wiki/Installation_Guide#YUM_Based_Installation
>>
>>
>>
>> On Fri, Mar 27, 2015 at 1:37 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> Can you file a JIRA for us to investigate this?
>>>
>>> On Fri, Mar 27, 2015 at 12:49 PM, Free Switcher <switcherfree at gmail.com>
>>> wrote:
>>>>
>>>> Hello Brian,
>>>> Thanks for your response. I did not build from source. I just
>>>> downloaded/installed pre-built packages using yum. On my CentOS 6.6 machine,
>>>> it looks like the freeswitch binary is installed under /usr/bin with configs
>>>> under /etc/freeswitch.
>>>>
>>>> Thanks,
>>>> Andy
>>>>
>>>>
>>>> On Fri, Mar 27, 2015 at 6:23 AM, Brian West <brian at freeswitch.org> wrote:
>>>>>
>>>>> If you downloaded and built the source its all going to be in
>>>>> /usr/local/freeswitch unless you configured it with the same configure args
>>>>> as the previous package install.
>>>>>
>>>>> On Fri, Mar 27, 2015 at 8:19 AM, I put the Who? in Mishehu
>>>>> <mishehu at freeswitch.org> wrote:
>>>>>>
>>>>>> Depending on what repo you got those packages from, they could be
>>>>>> seriously old packages.  Send back the output of:
>>>>>>
>>>>>> rpm -qa |grep -i freeswitch
>>>>>>
>>>>>> Chances are you only installed the base package.  you can also do `yum
>>>>>> search freeswitch` to see if other packages are offered. After you
>>>>>> provide the output from the above command I can comment on the age of
>>>>>> the packages you installed.
>>>>>>
>>>>>> --
>>>>>> Yossi Neiman
>>>>>>
>>>>>> On 03/27/2015 01:12 AM, Free Switcher wrote:
>>>>>>
>>>>>> > Hello,
>>>>>> > I'm trying to create a new installation of freeswitch. I'm running
>>>>>> > CentOS 6.6 and picked the easy path to install pre-built binaries
>>>>>> > using yum. Looking at /usr/lib64/freeswitch after the install, I
>>>>>> > noticed that mod_conference.so wasn't there. Also noticed that no
>>>>>> > config files were installed in /etc/freeswitch. Is this expected?
>>>>>> >
>>>>>> > I downloaded configs from the latest stash repository and I was then
>>>>>> > able to start freeswitch with vanilla config. I see several startup
>>>>>> > messages about missing mods. Which modules should I expect to see as
>>>>>> > part of the pre-built binaries? How/where does one download
>>>>>> > additional
>>>>>> > mods? Here is what I have in the /usr/lib64/freeswitch :
>>>>>> >
>>>>>> > mod_cdr_csv.so
>>>>>> > mod_commands.so
>>>>>> > mod_console.so
>>>>>> > mod_dialplan_directory.so
>>>>>> > mod_dialplan_xml.so
>>>>>> > mod_dptools.so
>>>>>> > mod_event_socket.so
>>>>>> > mod_logfile.so
>>>>>> > mod_loopback.so
>>>>>> > mod_native_file.so
>>>>>> > mod_sndfile.so
>>>>>> > mod_sofia.so
>>>>>> > mod_spandsp.so
>>>>>> > mod_syslog.so
>>>>>> > mod_tone_stream.so
>>>>>> > mod_xml_rpc.so
>>>>>> >
>>>>>> >
>>>>>> > Any help is appreciated.
>>>>>> > Thanks,
>>>>>> > Andy
>>>>>> >
>>>>>> >
>>>>>> > _________________________________________________________________________
>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>> > consulting at freeswitch.org
>>>>>> > http://www.freeswitchsolutions.com
>>>>>> >
>>>>>> > Official FreeSWITCH Sites
>>>>>> > http://www.freeswitch.org
>>>>>> > http://confluence.freeswitch.org
>>>>>> > http://www.cluecon.com
>>>>>> >
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >
>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> Brian West
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> Twitter: @FreeSWITCH , @briankwest
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> ClueCon 2015 Call for Speakers | Register TODAY!
>>>>>
>>>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>>>>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Brian West
>>> brian at freeswitch.org
>>>
>>>
>>> Twitter: @FreeSWITCH , @briankwest
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers | Register TODAY!
>>>
>>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>>
>>
>>
>>
>> --
>>
>> Brian West
>> brian at freeswitch.org
>>
>>
>> Twitter: @FreeSWITCH , @briankwest
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers | Register TODAY!
>>
>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From denis at ringme.ru  Mon Mar 30 13:17:29 2015
From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=)
Date: Mon, 30 Mar 2015 12:17:29 +0300
Subject: [Freeswitch-users] bind_digit_action or bind_meta_app?
Message-ID: <551914A9.6030202@ringme.ru>

Hello.

We need a bind to buttons with * and #, and both legs.
bind_digit_action simply binds to * and #, but on leg A only, on B it's 
perversion...

bind_meta_app - very strange behavior when i bind * AND #
         <action application="bind_meta_app" data="2 b b 
execute_extension::att_xfer XML features"/>
         <action application="set" data="bind_meta_key=#"/>
         <action application="bind_meta_app" data="1 b b 
execute_extension::dx XML features"/>

does not always work


From ashwinrath at gmail.com  Mon Mar 30 13:34:51 2015
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Mon, 30 Mar 2015 15:04:51 +0530
Subject: [Freeswitch-users] FSComm not building in windows
In-Reply-To: <CABH8QX0xjx7wRdd3SRvf87awcXi1B3Hswa0WxsaYrhGqA+zfwg@mail.gmail.com>
References: <CAA_Nwu5D2NeA4TQsP7GJyXNJDApKh6ZkLF7qotS8h6huru8sMg@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315D46B25@ex2.connex.local>
	<CAA_Nwu474WTRDjAAi-ndtvaPzX-svA-czmKASjpxs-73FtQ+WA@mail.gmail.com>
	<55155995.8070606@freeswitch.org>
	<CABH8QX0xjx7wRdd3SRvf87awcXi1B3Hswa0WxsaYrhGqA+zfwg@mail.gmail.com>
Message-ID: <CAA_Nwu5vA2Xr92uJVin2hGhCmD4He9U4NyjfxDTPAZ-p_XX_1g@mail.gmail.com>

I am using 4.6.0. I tried with 5.x but it didnt work as you mentioned.

On Sat, Mar 28, 2015 at 1:03 AM, Jo?o Mesquita <jmesquita at freeswitch.org>
wrote:

> What Qt version are you trying to compile agasint? Qt version 5 has
> changed considerably and won't ever compile properly...
>
> Jo?o Mesquita
>
> On Fri, Mar 27, 2015 at 10:22 AM, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>>  If you think you've found a bug please file a jira for it.  Though it's
>> possible you are simply missing a dependency on your system.  But either
>> way, we'll not be able to advise without seeing what those errors actually
>> are.
>>
>> --
>> Yossi Neiman
>>
>> On 03/27/2015 01:31 AM, Ashwin Rath wrote:
>>
>>  Yes i did
>>
>>  But even then i get errors like missing files or missing function
>> definitions.
>>
>> On Thu, Mar 26, 2015 at 10:27 PM, Moishe Grunstein <max at nysolutions.com>
>> wrote:
>>
>>>  Did you follow the wiki? https://wiki.freeswitch.org/wiki/FSComm
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Moishe Grunstein
>>>
>>> Tornado Computer Systems, Inc.
>>>
>>> 212.400.7650 888.IPPBX.US
>>> *Service Request Email: support at nysolutions.com
>>> <support at nysolutions.com> *
>>>
>>> [image: cid:image001.jpg at 01C72F94.9EE45D60]
>>> <http://www.nysolutions.com/>
>>>
>>> Computer Networking * Managed Services * IP Video Surveillance * Network
>>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>>> Security * Site Surveys * CMS
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashwin
>>> Rath
>>> *Sent:* Thursday, March 26, 2015 12:40 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] FSComm not building in windows
>>>
>>>
>>>
>>> Hi
>>>
>>> The FSComm app seems to have multiple issues while building on windows.
>>>
>>> 1) The mod_qsettings ins included in the project but not present on
>>> filesystem
>>>
>>> 2) the ISettings and AccountManager classes are not included in the
>>> project
>>>
>>> 3) Including the above classes causes linker errors for missing QT
>>> related methods such as qt_metacast, metaObject etc
>>>
>>> is FScomm actively maintained or a deprecated project?
>>>
>>>
>>> --
>>>
>>> Ashwin Rath
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Ashwin Kumar Rath
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ashwin Kumar Rath
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/63d3d20c/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/jpeg
Size: 2424 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/63d3d20c/attachment-0001.jpe 

From garyf9656 at gmail.com  Mon Mar 30 12:33:17 2015
From: garyf9656 at gmail.com (Gary F)
Date: Mon, 30 Mar 2015 09:33:17 +0100
Subject: [Freeswitch-users] Fwd: Files recorded using record_session do not
	include metadata
In-Reply-To: <CAMbyZx9MPVdqzOJwJTe5LKNZ+e8DnaTJinvwQCo8QLJGz464pA@mail.gmail.com>
References: <CAMbyZx9MPVdqzOJwJTe5LKNZ+e8DnaTJinvwQCo8QLJGz464pA@mail.gmail.com>
Message-ID: <CAMbyZx8QWxoJ+iQtqD5k+CaLYu+_KVnSvcciUtMEzO4w=homYA@mail.gmail.com>

Hi,

I'm using the following configuration to record internal calls however the
files that are produced don't seem to include any of the metadata
information. Do I need to do anything else to trigger the metadata to be
saved?

<action application="set" data="RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
<action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory Records,
Inc."/>
<action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
<action application="set" data="RECORD_ARTIST=Ian Curtis"/>
<action application="set" data="RECORD_COMMENT=Love will tear us apart"/>
<action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
<action application="set" data="RECORD_STEREO=true"/>
<action application="record_session"
data="$${base_dir}/recordings/meta_${uuid}.wav"/>

I'm using version 1.14.18.

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/96f72aa5/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar 30 17:46:45 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Mon, 30 Mar 2015 15:46:45 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
Message-ID: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>

Hi,

does someone know how to change the BLF lamp persistently? I found the
channel variable
presence id but this one will only last as long as the channel is active.
Is there a way to change
it permerently?

Thanks for helping.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/0fb9a2d8/attachment.html 

From mike at jerris.com  Mon Mar 30 18:08:31 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Mar 2015 10:08:31 -0400
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
Message-ID: <CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>

Can you describe a bit more exactly what you are trying to accomplish?
Presence changes in reaction to events that happen in calls.

On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
paul.atreides83 at googlemail.com> wrote:

> Hi,
>
> does someone know how to change the BLF lamp persistently? I found the
> channel variable
> presence id but this one will only last as long as the channel is active.
> Is there a way to change
> it permerently?
>
> Thanks for helping.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/2a55193b/attachment.html 

From paul.atreides83 at googlemail.com  Mon Mar 30 18:31:38 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Mon, 30 Mar 2015 16:31:38 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
Message-ID: <CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>

I want to have an indicator if the company voice mail is active or not.

At the moment I am calling a number and setting a global variable to
activate the central company voice mail.
But I would like to use the BLF from a dummy account so that the user can
see at the phone whether the voice mail is
active or not.


On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com> wrote:

> Can you describe a bit more exactly what you are trying to accomplish?
> Presence changes in reaction to events that happen in calls.
>
> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
> paul.atreides83 at googlemail.com> wrote:
>
>> Hi,
>>
>> does someone know how to change the BLF lamp persistently? I found the
>> channel variable
>> presence id but this one will only last as long as the channel is active.
>> Is there a way to change
>> it permerently?
>>
>> Thanks for helping.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/12858cbe/attachment.html 

From bpriddy at bryantschools.org  Mon Mar 30 18:43:42 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 30 Mar 2015 09:43:42 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
Message-ID: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>

Is anyone out there having success with sending/receiving faxes with
Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
have referred myself to this document
https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was
just wanting to see what everyone is doing out there with flowroute for the
people who still must use faxing..

-- 


*Blakelund Priddy*
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/c6c8b373/attachment.html 

From gmaruzz at gmail.com  Mon Mar 30 18:45:27 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 30 Mar 2015 16:45:27 +0200
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
Message-ID: <CALXCt0rXfjwR8ZMPixvRbnS7W81kSmkjGbGoGgUB=Feoq0OQUg@mail.gmail.com>

yep, never had a problem with flowroute and T38

-giovanni

On Mon, Mar 30, 2015 at 4:43 PM, Blake Priddy <bpriddy at bryantschools.org>
wrote:

> Is anyone out there having success with sending/receiving faxes with
> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
> have referred myself to this document
> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was
> just wanting to see what everyone is doing out there with flowroute for the
> people who still must use faxing..
>
> --
>
>
> *Blakelund Priddy*
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
> p 501-653-5038
> f 501-847-5656
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/2aadb1ad/attachment-0001.html 

From vladget at gmail.com  Mon Mar 30 18:48:56 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Mon, 30 Mar 2015 17:48:56 +0300
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
Message-ID: <CAMxcU5KDn3b6+i3vbFOP5NAmLj_py9bkT6y9uCF0mzgo_ASdHQ@mail.gmail.com>

Hi!

Did you guys change something with escaping hash ASCII char at SIP URI?
I guess you did.. Is there any way to except some character from URL
encoding list?

Thank you!

On Thu, Mar 26, 2015 at 6:39 PM, Ken Rice <krice at freeswitch.org> wrote:

> New Post on freeswitch.org from krice387
> check it out at http://ift.tt/1Izjq3R
> FreeSWITCH 1.4.18 Released!
>
> FreeSWITCH 1.4.18 has been released!
>
> This is routine maintenance release.
>
> Source Tarball available at http://ift.tt/1Izjssi
>
> Debian and Yum Repos have been updated as well.
>
> See the release notes below for a list of notable changes.
>
> For additional information about the FreeSWITCH release process, please
> see http://ift.tt/1E59SyO .
>
> FreeSWITCH 1.4.18 Release Notes
>
> FreeSWITCH 1.4.18 is a routine maintenance release.
>
> New features that were added:
>
>    - FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for
>    ftmod_libpri
>    - FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted
>    correctly by libpri, fallback to calling TON/type.
>    - FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
>    - FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
>    - FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line
>    arguments to allow for using -base instead of specifying each directory
>    when using alternate configs.
>    - FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one
>    leg
>    - FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in
>    the grammar tag for mod_rayo. This is useful for MRCP engines to look up
>    their grammars directly.
>    - FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for
>    mod_xml_cdr
>    - FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for
>    path to conf_dir -k for path to log_dir -m for path to mod_dir
>    - FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header
>    from the sip message
>    - FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML
>    dialplan date and time conditions
>    - FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy additional
>    variables
>    - FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in
>    mod_http_cache using {refresh=true} parameter that can be prefixed to a URL
>    to force refresh when using http:// https:// file formats or the
>    http_get API. And added http_remove_cache API call to manually expire a
>    cached URL.
>    - FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from
>    mod_event_socket to mod_erlang_event
>
>
>
> Improvements to the documentation:
>
>    - FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for
>    configuring FreeSWITCH from scratch
>
>
>
> Improvements in build system, cross platform support, and packaging:
>
>    - FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use
>    flite-2.0.0-release
>    - FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
>    - FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5
>    and other distributions with older autotools
>    - FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC
>    2013
>    - FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean
>    targets
>    - FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for
>    libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
>    - FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a
>    pre-bootstrapped orig file
>
>
>    - FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for
>    flite-2.0.0
>    - FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in
>    mod_conference
>    - FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on
>    Windows building ws.c
>    - FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with
>    clang compiler
>    - FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
>    - FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a
>    broken openssl 1.0.2 includes
>    - FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5
>    and other distributions with older autotools
>    - FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor of
>    our own json code
>    - FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer
>    configure flag to enable clang address sanitizer
>    - FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link
>    error that can occur when using clang
>
>
>
> The following bugs were fixed:
>
>    - FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded
>    which makes it impossible to call a registered user
>    - FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection string
>    to remove unnecessary option that may cause a failure on some systems
>    - FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is
>    in the uri
>    - FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable
>    to locate registered user
>    - FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident
>    will make it be an empty string which is the same you were doing on your
>    local builds in mod_spandsp
>    - FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command
>    argument bounds checking for flush and delete in mod_memcache
>    - FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
>    - FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully
>    stopped and ftdm is not running, return success from the span stop function.
>    - FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting
>    recorded files in append mode if appended data is shorter than
>    RECORD_MIN_SEC
>    - FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race
>    condition and segfault in mod_conference
>
>
>    - FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error
>    caused by 27c8622
>    - 0dc48df Fix for a bug from original implementation, cannot send call
>    state about state destroy, this is an internal state and the session is
>    already destroyed.
>    - FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load mod_java
>    - FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from
>    commit 525f1ac back in 2008
>    - FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk
>    negotiation issue
>    - FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from
>    commit 7c63670
>
>
>    - FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler
>    for SIG_CHLD because it isn?t necessary anymore and it causes dependent
>    libraries that tried to start a child process to hang waiting on a signal
>    that FreeSWITCH core intercepted.
>    - FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load
>    averages on older kernels with related bugs FS-7253 and FS-7231
>    - FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter
>    member cancels the call
>    - FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with
>    a pass through tone
>    - FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs
>    responding to an auth challenge in mod_sofia
>    - FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when
>    debugging is enabled in mod_sofia
>    - FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp
>    sleeping too much
>    - FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in mod_conference
>
>
>    - FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when recording
>    calls in native format
>    - FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the
>    translate application
>    - FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing
>    invalid params to the vm_fsdb_msg_email api in mod_voicemail
>
>
>    - FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql
>    statements for basic_calls and detailed_calls to happen after the creation
>    of the tables so the creation works and won?t have to be run a second time.
>    - FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in
>    mod_conference caused by FS-7230
>    - FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of
>    FS startup and shutdown faster, like in the case of tone2wav.
>    - FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to
>    group api command. When the ?multi? parameter is present, the group command
>    will return a list of group members delimited by :_: which allows for
>    multiply-registered endpoints to participate in a group.
>
>
>    - FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in
>    SPEECH-COMPLETE event and validate load input grammar URLs
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/f76c47bf/attachment.html 

From mike at jerris.com  Mon Mar 30 18:54:56 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Mar 2015 10:54:56 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <CAMxcU5KDn3b6+i3vbFOP5NAmLj_py9bkT6y9uCF0mzgo_ASdHQ@mail.gmail.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<CAMxcU5KDn3b6+i3vbFOP5NAmLj_py9bkT6y9uCF0mzgo_ASdHQ@mail.gmail.com>
Message-ID: <9D55D321-5ADC-47A6-A1FC-D0377CB36ED5@jerris.com>

new var "sofia_suppress_url_encoding"  that can tweak this behavior.  Out of curiosity, what are you getting that the defaults are not working for you?


> On Mar 30, 2015, at 10:48 AM, Vladimir Getmanshchuk <vladget at gmail.com> wrote:
> 
> Hi!
> 
> Did you guys change something with escaping hash ASCII char at SIP URI?
> I guess you did.. Is there any way to except some character from URL encoding list?
> 
> Thank you!
> 
> On Thu, Mar 26, 2015 at 6:39 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> New Post on freeswitch.org <http://freeswitch.org/> from krice387
> check it out at http://ift.tt/1Izjq3R <http://ift.tt/1Izjq3R>
> FreeSWITCH 1.4.18 Released!
> FreeSWITCH 1.4.18 has been released!
> 
> This is routine maintenance release.
> 
> Source Tarball available at http://ift.tt/1Izjssi <http://ift.tt/1Izjssi>
> Debian and Yum Repos have been updated as well.
> 
> See the release notes below for a list of notable changes.
> 
> For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO <http://ift.tt/1E59SyO> .
> 
> 
> FreeSWITCH 1.4.18 Release Notes
> 
> FreeSWITCH 1.4.18 is a routine maintenance release.
> 
> New features that were added:
> 
> FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for ftmod_libpri
> FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type.
> FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
> FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
> FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs.
> FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one leg
> FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly.
> FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for mod_xml_cdr
> FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir
> FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header from the sip message
> FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML dialplan date and time conditions
> FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy additional variables
> FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.
> FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from mod_event_socket to mod_erlang_event
>  
> Improvements to the documentation:
> 
> FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for configuring FreeSWITCH from scratch
>  
> Improvements in build system, cross platform support, and packaging:
> 
> FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use flite-2.0.0-release
> FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
> FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5 and other distributions with older autotools
> FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC 2013
> FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean targets
> FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
> FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a pre-bootstrapped orig file
> FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for flite-2.0.0
> FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in mod_conference
> FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on Windows building ws.c
> FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with clang compiler
> FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
> FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a broken openssl 1.0.2 includes
> FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5 and other distributions with older autotools
> FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor of our own json code
> FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer configure flag to enable clang address sanitizer
> FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link error that can occur when using clang
>  
> The following bugs were fixed:
> 
> FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded which makes it impossible to call a registered user
> FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems
> FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is in the uri
> FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable to locate registered user
> FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp
> FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache
> FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
> FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully stopped and ftdm is not running, return success from the span stop function.
> FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC
> FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race condition and segfault in mod_conference
> FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error caused by 27c8622
> 0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed.
> FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load mod_java
> FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from commit 525f1ac back in 2008
> FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk negotiation issue
> FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from commit 7c63670
> FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler for SIG_CHLD because it isn?t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted.
> FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231
> FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter member cancels the call
> FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with a pass through tone
> FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia
> FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when debugging is enabled in mod_sofia
> FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp sleeping too much
> FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in mod_conference
> FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when recording calls in native format
> FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the translate application
> FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail
> FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time.
> FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in mod_conference caused by FS-7230
> FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav.
> FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.
> FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/ecb99f41/attachment-0001.html 

From sdevoy at bizfocused.com  Mon Mar 30 19:17:58 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 30 Mar 2015 15:17:58 +0000
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
Message-ID: <BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>

I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.

Sean

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Atreides
Sent: Monday, March 30, 2015 10:32 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Changing BLF lamp persistently

I want to have an indicator if the company voice mail is active or not.

At the moment I am calling a number and setting a global variable to activate the central company voice mail.
But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
active or not.

On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.

On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com<mailto:paul.atreides83 at googlemail.com>> wrote:
Hi,

does someone know how to change the BLF lamp persistently? I found the channel variable
presence id but this one will only last as long as the channel is active. Is there a way to change
it permerently?

Thanks for helping.


_________________________________________________________________________
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://confluence.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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/743211e0/attachment.html 

From mike at jerris.com  Mon Mar 30 19:22:04 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Mar 2015 11:22:04 -0400
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <A0EC266D-6F08-4362-9E6F-E32258EB49BF@jerris.com>

parking has built in presence support, you can just subscribe to park+extennum

> On Mar 30, 2015, at 11:17 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.
>  
> Sean
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Atreides
> Sent: Monday, March 30, 2015 10:32 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Changing BLF lamp persistently
>  
> I want to have an indicator if the company voice mail is active or not.
> 
> At the moment I am calling a number and setting a global variable to activate the central company voice mail.
> But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
> active or not.
> 
>  
> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.
>  
> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com <mailto:paul.atreides83 at googlemail.com>> wrote:
> Hi,
> 
> does someone know how to change the BLF lamp persistently? I found the channel variable
> presence id but this one will only last as long as the channel is active. Is there a way to change
> it permerently?
> 
> Thanks for helping.
>  
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/ba6418f0/attachment.html 

From brian at freeswitch.org  Mon Mar 30 19:23:47 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 30 Mar 2015 10:23:47 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CALXCt0rXfjwR8ZMPixvRbnS7W81kSmkjGbGoGgUB=Feoq0OQUg@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CALXCt0rXfjwR8ZMPixvRbnS7W81kSmkjGbGoGgUB=Feoq0OQUg@mail.gmail.com>
Message-ID: <CAEJMVkDh1t1ok6Xm1tzPR0Q1JLnjEu0OEeXC_PzNU1XSCbHsAw@mail.gmail.com>

What firmware do you have on that SPA?

On Mon, Mar 30, 2015 at 9:45 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> yep, never had a problem with flowroute and T38
>
> -giovanni
>
> On Mon, Mar 30, 2015 at 4:43 PM, Blake Priddy <bpriddy at bryantschools.org>
> wrote:
>
>> Is anyone out there having success with sending/receiving faxes with
>> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
>> have referred myself to this document
>> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I
>> was just wanting to see what everyone is doing out there with flowroute for
>> the people who still must use faxing..
>>
>> --
>>
>>
>> *Blakelund Priddy*
>> Network & Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org
>> p 501-653-5038
>> f 501-847-5656
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/225fc18d/attachment-0001.html 

From raphael.lechner at gmail.com  Mon Mar 30 19:25:37 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Mon, 30 Mar 2015 17:25:37 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
Message-ID: <5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>

We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they had also a lamp for indicating if the ?today closed? playback or call forward to voicemail is enabled or not.

Raphael

> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.
>  
> Sean
>  
> 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 Paul Atreides
> Sent: Monday, March 30, 2015 10:32 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Changing BLF lamp persistently
>  
> I want to have an indicator if the company voice mail is active or not.
> 
> At the moment I am calling a number and setting a global variable to activate the central company voice mail.
> But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
> active or not.
> 
>  
> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.
>  
> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com <mailto:paul.atreides83 at googlemail.com>> wrote:
> Hi,
> 
> does someone know how to change the BLF lamp persistently? I found the channel variable
> presence id but this one will only last as long as the channel is active. Is there a way to change
> it permerently?
> 
> Thanks for helping.
>  
> 
>  
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/bc51ae19/attachment-0001.html 

From andrew at cassidywebservices.co.uk  Mon Mar 30 19:37:08 2015
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 30 Mar 2015 16:37:08 +0100
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
Message-ID: <CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>

There's always the SEND_PRESENCE esl event...

On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
wrote:

> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they
> had also a lamp for indicating if the ?today closed? playback or call
> forward to voicemail is enabled or not.
>
> Raphael
>
>
> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
> I have a customer who had this same request as well as one for a lamp
> indicating a call parked using ?valet parking?.  They are worried a parked
> call could get forgotten.  I would love to hear a solution to either/both.
>
> Sean
>
> *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 *Paul
> Atreides
> *Sent:* Monday, March 30, 2015 10:32 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>
> I want to have an indicator if the company voice mail is active or not.
>
> At the moment I am calling a number and setting a global variable to
> activate the central company voice mail.
> But I would like to use the BLF from a dummy account so that the user can
> see at the phone whether the voice mail is
>
> active or not.
>
> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Can you describe a bit more exactly what you are trying to accomplish?
> Presence changes in reaction to events that happen in calls.
>
> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
> paul.atreides83 at googlemail.com> wrote:
>
> Hi,
>
> does someone know how to change the BLF lamp persistently? I found the
> channel variable
> presence id but this one will only last as long as the channel is active.
> Is there a way to change
> it permerently?
>
> Thanks for helping.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/82532812/attachment.html 

From msc at freeswitch.org  Mon Mar 30 20:59:26 2015
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 30 Mar 2015 09:59:26 -0700
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
Message-ID: <CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>

Well said! Faxing is evil, especially in a VoIP environment.

To add to what Brian said, I think we've seen some success with version
1.3.1 of the SPA firmware for that device.

-MC


On Mon, Mar 30, 2015 at 7:43 AM, Blake Priddy <bpriddy at bryantschools.org>
wrote:

> Is anyone out there having success with sending/receiving faxes with
> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
> have referred myself to this document
> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was
> just wanting to see what everyone is doing out there with flowroute for the
> people who still must use faxing..
>
> --
>
>
> *Blakelund Priddy*
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
> p 501-653-5038
> f 501-847-5656
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/1e6aa2c6/attachment-0001.html 

From nneul at mst.edu  Mon Mar 30 20:59:32 2015
From: nneul at mst.edu (Nathan Neulinger)
Date: Mon, 30 Mar 2015 11:59:32 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
Message-ID: <551980F4.4090101@mst.edu>

I have not done anything with T38, but I do know that with plain analog I had no end of problems with the SPA 112. The 
2102 worked great, but the 112 didn't work worth anything. We abandoned efforts to get it to work and just switched to 
Grandstream HT701.

-- Nathan

On 03/30/2015 09:43 AM, Blake Priddy wrote:
> Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just
> the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with
> no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use
> faxing..
>
> --
> *
> *
> *Blakelund Priddy*
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org <http://www.bryantschools.org/>
> p 501-653-5038
> f 501-847-5656
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bpriddy at bryantschools.org  Mon Mar 30 21:01:54 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 30 Mar 2015 12:01:54 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CAEJMVkDh1t1ok6Xm1tzPR0Q1JLnjEu0OEeXC_PzNU1XSCbHsAw@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CALXCt0rXfjwR8ZMPixvRbnS7W81kSmkjGbGoGgUB=Feoq0OQUg@mail.gmail.com>
	<CAEJMVkDh1t1ok6Xm1tzPR0Q1JLnjEu0OEeXC_PzNU1XSCbHsAw@mail.gmail.com>
Message-ID: <CADK-BjuWS1S0_RxVv1PBw4Lc0Cp8cra=MQinj0cAKzZnbqTcrw@mail.gmail.com>

1.3.3 (015) Dec 13 2013
Looks a little old ;)

On Mon, Mar 30, 2015 at 10:23 AM, Brian West <brian at freeswitch.org> wrote:

> What firmware do you have on that SPA?
>
> On Mon, Mar 30, 2015 at 9:45 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> yep, never had a problem with flowroute and T38
>>
>> -giovanni
>>
>> On Mon, Mar 30, 2015 at 4:43 PM, Blake Priddy <bpriddy at bryantschools.org>
>> wrote:
>>
>>> Is anyone out there having success with sending/receiving faxes with
>>> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
>>> have referred myself to this document
>>> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I
>>> was just wanting to see what everyone is doing out there with flowroute for
>>> the people who still must use faxing..
>>>
>>> --
>>>
>>>
>>> *Blakelund Priddy*
>>> Network & Systems Engineer
>>> Bryant Public School District
>>> Bryant, Arkansas 72022
>>> http://www.bryantschools.org
>>> p 501-653-5038
>>> f 501-847-5656
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY!
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


*Blakelund Priddy*
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/4c1c5096/attachment.html 

From max at nysolutions.com  Mon Mar 30 21:08:00 2015
From: max at nysolutions.com (Moishe Grunstein)
Date: Mon, 30 Mar 2015 17:08:00 +0000
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-BjuWS1S0_RxVv1PBw4Lc0Cp8cra=MQinj0cAKzZnbqTcrw@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CALXCt0rXfjwR8ZMPixvRbnS7W81kSmkjGbGoGgUB=Feoq0OQUg@mail.gmail.com>
	<CAEJMVkDh1t1ok6Xm1tzPR0Q1JLnjEu0OEeXC_PzNU1XSCbHsAw@mail.gmail.com>
	<CADK-BjuWS1S0_RxVv1PBw4Lc0Cp8cra=MQinj0cAKzZnbqTcrw@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315D56249@ex2.connex.local>

Yeah Back in the 90?s we had no problems faxing, nowadays I find email or IP faxing much more reliable and secure. T38 works if your device and providers and their upstreams support it.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy
Sent: Monday, March 30, 2015 1:02 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Fax....How I hate you.

1.3.3 (015) Dec 13 2013
Looks a little old ;)

On Mon, Mar 30, 2015 at 10:23 AM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
What firmware do you have on that SPA?

On Mon, Mar 30, 2015 at 9:45 AM, Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>> wrote:
yep, never had a problem with flowroute and T38
-giovanni

On Mon, Mar 30, 2015 at 4:43 PM, Blake Priddy <bpriddy at bryantschools.org<mailto:bpriddy at bryantschools.org>> wrote:
Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing..

--


Blakelund Priddy
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org<http://www.bryantschools.org/>
p 501-653-5038<tel:501-653-5038>
f 501-847-5656<tel:501-847-5656>

_________________________________________________________________________
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://confluence.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<tel:%2B39-347-2665618>

_________________________________________________________________________
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://confluence.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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY!

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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://confluence.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



--
[https://dl.dropboxusercontent.com/u/6313391/BryantSchoolDist_Seal_final.png] [http://www.aerohive.com/sites/default/files/acwa.png]

Blakelund Priddy
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org<http://www.bryantschools.org/>
p 501-653-5038
f 501-847-5656
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/e7b19f6a/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 2424 bytes
Desc: image001.jpg
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/e7b19f6a/attachment-0001.jpg 

From bpriddy at bryantschools.org  Mon Mar 30 21:12:43 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 30 Mar 2015 12:12:43 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
Message-ID: <CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>

 condition  context  public  10
<https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>

<https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
  condition  destination_number  ^(15012449620)$  20
<https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>

<https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
  action  set  fax_enable_t38=true  26
<https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>

<https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
  action  set  fax_enable_t38_request=true  27
<https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>

<https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
  action  transfer  109 XML default

On Mon, Mar 30, 2015 at 11:59 AM, Michael Collins <msc at freeswitch.org>
wrote:

> Well said! Faxing is evil, especially in a VoIP environment.
>
> To add to what Brian said, I think we've seen some success with version
> 1.3.1 of the SPA firmware for that device.
>
> -MC
>
>
> On Mon, Mar 30, 2015 at 7:43 AM, Blake Priddy <bpriddy at bryantschools.org>
> wrote:
>
>> Is anyone out there having success with sending/receiving faxes with
>> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
>> have referred myself to this document
>> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I
>> was just wanting to see what everyone is doing out there with flowroute for
>> the people who still must use faxing..
>>
>> --
>>
>>
>> *Blakelund Priddy*
>> Network & Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org
>> p 501-653-5038
>> f 501-847-5656
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


*Blakelund Priddy*
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/d1e3e13f/attachment.html 

From bpriddy at bryantschools.org  Mon Mar 30 21:13:06 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 30 Mar 2015 12:13:06 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
	<CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>
Message-ID: <CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>

Didnt mean to send that just yet....

On Mon, Mar 30, 2015 at 12:12 PM, Blake Priddy <bpriddy at bryantschools.org>
wrote:

>  condition  context  public  10
> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>
> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>   condition  destination_number  ^(15012449620)$  20
> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>
> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>   action  set  fax_enable_t38=true  26
> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>
> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>   action  set  fax_enable_t38_request=true  27
> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>
> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>   action  transfer  109 XML default
>
> On Mon, Mar 30, 2015 at 11:59 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>
>> Well said! Faxing is evil, especially in a VoIP environment.
>>
>> To add to what Brian said, I think we've seen some success with version
>> 1.3.1 of the SPA firmware for that device.
>>
>> -MC
>>
>>
>> On Mon, Mar 30, 2015 at 7:43 AM, Blake Priddy <bpriddy at bryantschools.org>
>> wrote:
>>
>>> Is anyone out there having success with sending/receiving faxes with
>>> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
>>> have referred myself to this document
>>> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I
>>> was just wanting to see what everyone is doing out there with flowroute for
>>> the people who still must use faxing..
>>>
>>> --
>>>
>>>
>>> *Blakelund Priddy*
>>> Network & Systems Engineer
>>> Bryant Public School District
>>> Bryant, Arkansas 72022
>>> http://www.bryantschools.org
>>> p 501-653-5038
>>> f 501-847-5656
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
> *Blakelund Priddy*
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
> p 501-653-5038
> f 501-847-5656
>



-- 


*Blakelund Priddy*
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/53cd6568/attachment-0001.html 

From vladget at gmail.com  Mon Mar 30 21:57:03 2015
From: vladget at gmail.com (Vladimir Getmanshchuk)
Date: Mon, 30 Mar 2015 20:57:03 +0300
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <9D55D321-5ADC-47A6-A1FC-D0377CB36ED5@jerris.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<CAMxcU5KDn3b6+i3vbFOP5NAmLj_py9bkT6y9uCF0mzgo_ASdHQ@mail.gmail.com>
	<9D55D321-5ADC-47A6-A1FC-D0377CB36ED5@jerris.com>
Message-ID: <CAMxcU5JA1USG-dMsTwVcVUwOaP=FqqSs6FPBwzTgQQy=umd4Nw@mail.gmail.com>

Michael, Thanks!

Is this variable present at last 1.4.18?
May I user it per channel in [ ] before bridge application?

Relating your question.
I have some SIP operators configured on switch with no supporting of
escaped hash char in SIP URL

On Mon, Mar 30, 2015 at 5:54 PM, Michael Jerris <mike at jerris.com> wrote:

> new var "sofia_suppress_url_encoding"  that can tweak this behavior.  Out
> of curiosity, what are you getting that the defaults are not working for
> you?
>
>
> On Mar 30, 2015, at 10:48 AM, Vladimir Getmanshchuk <vladget at gmail.com>
> wrote:
>
> Hi!
>
> Did you guys change something with escaping hash ASCII char at SIP URI?
> I guess you did.. Is there any way to except some character from URL
> encoding list?
>
> Thank you!
>
> On Thu, Mar 26, 2015 at 6:39 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> New Post on freeswitch.org from krice387
>> check it out at http://ift.tt/1Izjq3R
>> FreeSWITCH 1.4.18 Released!
>>
>> FreeSWITCH 1.4.18 has been released!
>>
>> This is routine maintenance release.
>>
>> Source Tarball available at http://ift.tt/1Izjssi
>>
>> Debian and Yum Repos have been updated as well.
>>
>> See the release notes below for a list of notable changes.
>>
>> For additional information about the FreeSWITCH release process, please
>> see http://ift.tt/1E59SyO .
>>
>> FreeSWITCH 1.4.18 Release Notes
>>
>> FreeSWITCH 1.4.18 is a routine maintenance release.
>>
>> New features that were added:
>>
>>    - FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for
>>    ftmod_libpri
>>    - FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted
>>    correctly by libpri, fallback to calling TON/type.
>>    - FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
>>    - FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
>>    - FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line
>>    arguments to allow for using -base instead of specifying each directory
>>    when using alternate configs.
>>    - FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one
>>    leg
>>    - FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in
>>    the grammar tag for mod_rayo. This is useful for MRCP engines to look up
>>    their grammars directly.
>>    - FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for
>>    mod_xml_cdr
>>    - FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for
>>    path to conf_dir -k for path to log_dir -m for path to mod_dir
>>    - FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header
>>    from the sip message
>>    - FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for
>>    XML dialplan date and time conditions
>>    - FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy
>>    additional variables
>>    - FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in
>>    mod_http_cache using {refresh=true} parameter that can be prefixed to a URL
>>    to force refresh when using http:// https:// file formats or the
>>    http_get API. And added http_remove_cache API call to manually expire a
>>    cached URL.
>>    - FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from
>>    mod_event_socket to mod_erlang_event
>>
>>
>>
>> Improvements to the documentation:
>>
>>    - FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for
>>    configuring FreeSWITCH from scratch
>>
>>
>>
>> Improvements in build system, cross platform support, and packaging:
>>
>>    - FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use
>>    flite-2.0.0-release
>>    - FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
>>    - FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5
>>    and other distributions with older autotools
>>    - FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using
>>    MSVC 2013
>>    - FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean
>>    targets
>>    - FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for
>>    libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
>>    - FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a
>>    pre-bootstrapped orig file
>>
>>
>>    - FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for
>>    flite-2.0.0
>>    - FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in
>>    mod_conference
>>    - FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on
>>    Windows building ws.c
>>    - FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with
>>    clang compiler
>>    - FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
>>    - FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a
>>    broken openssl 1.0.2 includes
>>    - FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5
>>    and other distributions with older autotools
>>    - FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor
>>    of our own json code
>>    - FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer
>>    configure flag to enable clang address sanitizer
>>    - FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link
>>    error that can occur when using clang
>>
>>
>>
>> The following bugs were fixed:
>>
>>    - FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded
>>    which makes it impossible to call a registered user
>>    - FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection
>>    string to remove unnecessary option that may cause a failure on some systems
>>    - FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is
>>    in the uri
>>    - FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns
>>    unable to locate registered user
>>    - FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident
>>    will make it be an empty string which is the same you were doing on your
>>    local builds in mod_spandsp
>>    - FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command
>>    argument bounds checking for flush and delete in mod_memcache
>>    - FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
>>    - FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully
>>    stopped and ftdm is not running, return success from the span stop function.
>>    - FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting
>>    recorded files in append mode if appended data is shorter than
>>    RECORD_MIN_SEC
>>    - FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race
>>    condition and segfault in mod_conference
>>
>>
>>    - FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error
>>    caused by 27c8622
>>    - 0dc48df Fix for a bug from original implementation, cannot send
>>    call state about state destroy, this is an internal state and the session
>>    is already destroyed.
>>    - FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load
>>    mod_java
>>    - FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from
>>    commit 525f1ac back in 2008
>>    - FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk
>>    negotiation issue
>>    - FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from
>>    commit 7c63670
>>
>>
>>    - FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler
>>    for SIG_CHLD because it isn?t necessary anymore and it causes dependent
>>    libraries that tried to start a child process to hang waiting on a signal
>>    that FreeSWITCH core intercepted.
>>    - FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load
>>    averages on older kernels with related bugs FS-7253 and FS-7231
>>    - FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter
>>    member cancels the call
>>    - FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers
>>    with a pass through tone
>>    - FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs
>>    responding to an auth challenge in mod_sofia
>>    - FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when
>>    debugging is enabled in mod_sofia
>>    - FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp
>>    sleeping too much
>>    - FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in
>>    mod_conference
>>
>>
>>    - FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when
>>    recording calls in native format
>>    - FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the
>>    translate application
>>    - FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing
>>    invalid params to the vm_fsdb_msg_email api in mod_voicemail
>>
>>
>>    - FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql
>>    statements for basic_calls and detailed_calls to happen after the creation
>>    of the tables so the creation works and won?t have to be run a second time.
>>    - FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in
>>    mod_conference caused by FS-7230
>>    - FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of
>>    FS startup and shutdown faster, like in the case of tone2wav.
>>    - FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to
>>    group api command. When the ?multi? parameter is present, the group command
>>    will return a list of group members delimited by :_: which allows for
>>    multiply-registered endpoints to participate in a group.
>>
>>
>>    - FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes
>>    in SPEECH-COMPLETE event and validate load input grammar URLs
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Yours sincerely,
Vladimir Getmanshchuk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/76470b65/attachment.html 

From mike at jerris.com  Mon Mar 30 22:11:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 30 Mar 2015 14:11:11 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.4.18 Released!
In-Reply-To: <CAMxcU5JA1USG-dMsTwVcVUwOaP=FqqSs6FPBwzTgQQy=umd4Nw@mail.gmail.com>
References: <55143638ae25f_a1bc51b328192d2@resque-worker-ip-10-13-198-102.mail>
	<CAMxcU5KDn3b6+i3vbFOP5NAmLj_py9bkT6y9uCF0mzgo_ASdHQ@mail.gmail.com>
	<9D55D321-5ADC-47A6-A1FC-D0377CB36ED5@jerris.com>
	<CAMxcU5JA1USG-dMsTwVcVUwOaP=FqqSs6FPBwzTgQQy=umd4Nw@mail.gmail.com>
Message-ID: <428E61E9-0D1F-4771-B6CA-BAF1334B8011@jerris.com>

it is in 1.4.18.  I believe it will work fine from [] or { } but I have not tested it myself, so you should confirm.  Its strange, what are you putting in there that is getting encoded?  I believe its actually a required part of the spec, but should also be very rare that it happens.

Mike

> On Mar 30, 2015, at 1:57 PM, Vladimir Getmanshchuk <vladget at gmail.com> wrote:
> 
> Michael, Thanks! 
> 
> Is this variable present at last 1.4.18?
> May I user it per channel in [ ] before bridge application?
> 
> Relating your question. 
> I have some SIP operators configured on switch with no supporting of escaped hash char in SIP URL
> 
> On Mon, Mar 30, 2015 at 5:54 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> new var "sofia_suppress_url_encoding"  that can tweak this behavior.  Out of curiosity, what are you getting that the defaults are not working for you?
> 
> 
>> On Mar 30, 2015, at 10:48 AM, Vladimir Getmanshchuk <vladget at gmail.com <mailto:vladget at gmail.com>> wrote:
>> 
>> Hi!
>> 
>> Did you guys change something with escaping hash ASCII char at SIP URI?
>> I guess you did.. Is there any way to except some character from URL encoding list?
>> 
>> Thank you!
>> 
>> On Thu, Mar 26, 2015 at 6:39 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>> New Post on freeswitch.org <http://freeswitch.org/> from krice387
>> check it out at http://ift.tt/1Izjq3R <http://ift.tt/1Izjq3R>
>> FreeSWITCH 1.4.18 Released!
>> FreeSWITCH 1.4.18 has been released!
>> 
>> This is routine maintenance release.
>> 
>> Source Tarball available at http://ift.tt/1Izjssi <http://ift.tt/1Izjssi>
>> Debian and Yum Repos have been updated as well.
>> 
>> See the release notes below for a list of notable changes.
>> 
>> For additional information about the FreeSWITCH release process, please see http://ift.tt/1E59SyO <http://ift.tt/1E59SyO> .
>> 
>> 
>> FreeSWITCH 1.4.18 Release Notes
>> 
>> FreeSWITCH 1.4.18 is a routine maintenance release.
>> 
>> New features that were added:
>> 
>> FS-7201 <http://ift.tt/1Izjssm> Set ANI plan and ANI type for ftmod_libpri
>> FS-7209 <http://ift.tt/1E59QHb> If ANI TON is not interpreted correctly by libpri, fallback to calling TON/type.
>> FS-7265 <http://ift.tt/1CBB8Bj> Add mongo_find_n API
>> FS-7269 <http://ift.tt/1DZDaxd> Add error logs in mod_java
>> FS-7284 <http://ift.tt/1DLZjjG> A simplification of command line arguments to allow for using -base instead of specifying each directory when using alternate configs.
>> FS-7285 <http://ift.tt/1Flzggf> Allow eavesdrop to bridge only one leg
>> FS-7164 <http://ift.tt/1DLZhbq> Added support for URL attribute in the grammar tag for mod_rayo. This is useful for MRCP engines to look up their grammars directly.
>> FS-7299 <http://ift.tt/1Flzgwx> Implement cookie-file option for mod_xml_cdr
>> FS-7302 <http://ift.tt/1DLZjjI> Added params to fs_encode.c: -c for path to conf_dir -k for path to log_dir -m for path to mod_dir
>> FS-7309 <http://ift.tt/1FlzgwB> Allow removal of User-Agent header from the sip message
>> FS-7304 <http://ift.tt/1DLZjzZ> Multiple and reversed ranges for XML dialplan date and time conditions
>> FS-7312 <http://ift.tt/1vUSFoy> Update mod_verto to proxy additional variables
>> FS-7323 <http://ift.tt/1GJ3SJN> Add ability to force URL refresh in mod_http_cache using {refresh=true} parameter that can be prefixed to a URL to force refresh when using http:// https:// file formats or the http_get API. And added http_remove_cache API call to manually expire a cached URL.
>> FS-7354 <http://ift.tt/1GeoZmD> Filter feature ported from mod_event_socket to mod_erlang_event
>>  
>> Improvements to the documentation:
>> 
>> FS-7362 <http://ift.tt/1O0xNC0> Add minimal configuration for configuring FreeSWITCH from scratch
>>  
>> Improvements in build system, cross platform support, and packaging:
>> 
>> FS-7149 <http://ift.tt/1Gl1nRp> Update Windows build to use flite-2.0.0-release
>> FS-7346 <http://ift.tt/18vTmJJ> Update mod_mongo driver to 1.1.0
>> FS-7122 <http://ift.tt/1FlzgwF> Fixed issues building on CentOS 5 and other distributions with older autotools
>> FS-6520 <http://ift.tt/1GJ3SJQ> Fix for libv8 build issue using MSVC 2013
>> FS-7245 <http://ift.tt/1zPRy84> Don?t rebuild core on mod_foo-clean targets
>> FS-7270 <http://ift.tt/1CBB8Bo> Set the makefile to look for libtool-bin first and update libjpeg-dev to libjpeg8-dev in Debian makefile
>> FS-7318 <http://ift.tt/1DLZjA3> Debian rules update to handle a pre-bootstrapped orig file
>> FS-7149 <http://ift.tt/1Gl1nRp> Update freeswitch.spec for flite-2.0.0
>> FS-7236 <http://ift.tt/1AbGMuF> Fix code before declaration in mod_conference
>> FS-7264 <http://ift.tt/1CBB8RI> Fix signed/unsigned warnings on Windows building ws.c
>> FS-7294 <http://ift.tt/1Flzio8> Enable -Werror when building with clang compiler
>> FS-7296 <http://ift.tt/1DLZjA5> Fix build error on newest gcc
>> FS-7314 <http://ift.tt/1vUSEkv> Fix for configure error caused by a broken openssl 1.0.2 includes
>> FS-7322 <http://ift.tt/1vUSFER> Fix for issues building on CentOS 5 and other distributions with older autotools
>> FS-7340 <http://ift.tt/1BjWYKt> Remove json-c dependency in favor of our own json code
>> FS-7350 <http://ift.tt/18vTmJL> Add ?enable-address-sanitizer configure flag to enable clang address sanitizer
>> FS-7355 <http://ift.tt/1O0xMhx> Fix rpl_realloc symbol missing link error that can occur when using clang
>>  
>> The following bugs were fixed:
>> 
>> FS-7193 <http://ift.tt/1Akme4d> Fix for sofia contact being encoded which makes it impossible to call a registered user
>> FS-7191 <http://ift.tt/1yw2rsQ> Edit pgsql example connection string to remove unnecessary option that may cause a failure on some systems
>> FS-7205 <http://ift.tt/1Akmccp> Do not url encode unless an ?@? is in the uri
>> FS-7211 <http://ift.tt/1yw2rsW> Fix for sofia_contact returns unable to locate registered user
>> FS-7208 <http://ift.tt/1Akmekt> _undef_ as the header and/or ident will make it be an empty string which is the same you were doing on your local builds in mod_spandsp
>> FS-7214 <http://ift.tt/1yw2u7Y> Fix segfault caused by bad command argument bounds checking for flush and delete in mod_memcache
>> FS-7217 <http://ift.tt/1yw2u84> Use upper case when you query
>> FS-7197 <http://ift.tt/1Izjqkp> If the span has been already fully stopped and ftdm is not running, return success from the span stop function.
>> FS-7235 <http://ift.tt/1AbGLqx> Fix for call recording deleting recorded files in append mode if appended data is shorter than RECORD_MIN_SEC
>> FS-7236 <http://ift.tt/1AbGMuF> Added lock to prevent a race condition and segfault in mod_conference
>> FS-7236 <http://ift.tt/1AbGMuF> Fix mutex use before init error caused by 27c8622
>> 0dc48df Fix for a bug from original implementation, cannot send call state about state destroy, this is an internal state and the session is already destroyed.
>> FS-7256 <http://ift.tt/1CBB5FN> Fix for being unable to load mod_java
>> FS-7252 <http://ift.tt/1DZDaNr> Fix for 6-year-old regression from commit 525f1ac back in 2008
>> FS-7260 <http://ift.tt/1DZDcoK> Fix for L16 at 16000h with Asterisk negotiation issue
>> FS-7236 <http://ift.tt/1AbGMuF> Re-factor to fix audio problem from commit 7c63670
>> FS-7250 <http://ift.tt/1Flzio8> Removed the FreeSWITCH core handler for SIG_CHLD because it isn?t necessary anymore and it causes dependent libraries that tried to start a child process to hang waiting on a signal that FreeSWITCH core intercepted.
>> FS-7066 <http://ift.tt/1DLZjA5> Fixed a bug causing higher cpu load averages on older kernels with related bugs FS-7253 and FS-7231
>> FS-7298 <http://ift.tt/1DLZjA5> Fix race condition when callcenter member cancels the call
>> FS-7301 <http://ift.tt/1FlzgwJ> Fix for issue faxing to numbers with a pass through tone
>> FS-7192 <http://ift.tt/1DLZjAb> Exclude Expires header in INVITEs responding to an auth challenge in mod_sofia
>> FS-7308 <http://ift.tt/1Flziok> Only log SLA SQL query SQL when debugging is enabled in mod_sofia
>> FS-7306 <http://ift.tt/1DLZhrT> Fix for fs_encode in mod_spandsp sleeping too much
>> FS-7230 <http://ift.tt/1FlzgwN> Fixed a memory leak in mod_conference
>> FS-7307 <http://ift.tt/1vUSEkp> Fixed buffering issue when recording calls in native format
>> FS-7126 <http://ift.tt/1GJ3R8O> Fixed coredump when calling the translate application
>> FS-7313 <http://ift.tt/1vUSFoD> Fix for coredump when passing invalid params to the vm_fsdb_msg_email api in mod_voicemail
>> FS-7339 <http://ift.tt/1BjWWCs> Move the creation of view sql statements for basic_calls and detailed_calls to happen after the creation of the tables so the creation works and won?t have to be run a second time.
>> FS-7342 <http://ift.tt/18vTol4> Fixed a crash regression in mod_conference caused by FS-7230
>> FS-7305 <http://ift.tt/18vTmJN> Fix for making embedded versions of FS startup and shutdown faster, like in the case of tone2wav.
>> FS-5570 <http://ift.tt/1BjWYKx> Patch to add ?multi? parameter to group api command. When the ?multi? parameter is present, the group command will return a list of group members delimited by :_: which allows for multiply-registered endpoints to participate in a group.
>> FS-7300 <http://ift.tt/1Gep0Hk> Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs
>> 
> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Yours sincerely,
> Vladimir Getmanshchuk
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/15f8247e/attachment-0001.html 

From schoch+freeswitch.org at xwin32.com  Mon Mar 30 22:15:01 2015
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Mon, 30 Mar 2015 11:15:01 -0700
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
	<CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>
	<CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>
Message-ID: <CAOkuKdetQcUMr6AiWNDto6qXannTtJP2yCymFhHrUJmJP24v1Q@mail.gmail.com>

We're using a Grandstream HT502 to send from a FAX machine. It works
sometimes. Fortunately, we don't send many FAXes, but instead scan to a PDF
and email it.

We still have a POTS line that we use to receive FAXes (using HylaFax).
That works well, but I'd like to get rid of the POTS line eventually. I
have to be careful, however, because some customers still insist on FAXing
purchase orders.

-- 
Steve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/a9406716/attachment.html 

From blasterjr at gmail.com  Mon Mar 30 22:43:48 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Mon, 30 Mar 2015 12:43:48 -0600
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
Message-ID: <CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>

This can be done with lua, what i've done is created a small lua set of
functions that make presence easy to handle.

here's the presence function file:
http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq

then you setup a lua file (call it scripts/set_presence.lua) that is
similar to the following

dofile "presence.lua"

state = argv[2]
user = argv[1]
domain = session:getVariable('domain')

local p = Presence:new()
p:init{user = user, domain = domain, uuid = 'custom_blf_key'};

p:set(state,false);

you can execute this by adding the following line into your extension that
controls it.

<action application="lua" data="set_presence.lua vm_blf_extension
confirmed"/>

you can swap out the "vm_blf_extension" with anything you want, and as long
as the phone "Subscribes" to this, it will work.

You can use early, confirmed, terminated for the statuses.


On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> There's always the SEND_PRESENCE esl event...
>
> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
> wrote:
>
>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they
>> had also a lamp for indicating if the ?today closed? playback or call
>> forward to voicemail is enabled or not.
>>
>> Raphael
>>
>>
>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>
>> I have a customer who had this same request as well as one for a lamp
>> indicating a call parked using ?valet parking?.  They are worried a parked
>> call could get forgotten.  I would love to hear a solution to either/both.
>>
>> Sean
>>
>> *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 *Paul
>> Atreides
>> *Sent:* Monday, March 30, 2015 10:32 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>>
>> I want to have an indicator if the company voice mail is active or not.
>>
>> At the moment I am calling a number and setting a global variable to
>> activate the central company voice mail.
>> But I would like to use the BLF from a dummy account so that the user can
>> see at the phone whether the voice mail is
>>
>> active or not.
>>
>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> Can you describe a bit more exactly what you are trying to accomplish?
>> Presence changes in reaction to events that happen in calls.
>>
>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
>> paul.atreides83 at googlemail.com> wrote:
>>
>> Hi,
>>
>> does someone know how to change the BLF lamp persistently? I found the
>> channel variable
>> presence id but this one will only last as long as the channel is active.
>> Is there a way to change
>> it permerently?
>>
>> Thanks for helping.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/152f1970/attachment-0001.html 

From bpriddy at bryantschools.org  Mon Mar 30 22:52:03 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Mon, 30 Mar 2015 13:52:03 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CAOkuKdetQcUMr6AiWNDto6qXannTtJP2yCymFhHrUJmJP24v1Q@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
	<CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>
	<CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>
	<CAOkuKdetQcUMr6AiWNDto6qXannTtJP2yCymFhHrUJmJP24v1Q@mail.gmail.com>
Message-ID: <CADK-BjtY_9HZjtpQT80sj3AsGh2OAFR1eaOYC=tpHdCf74vm9w@mail.gmail.com>

Yes I have nurses here that say fax is more secure. That's why its HIPPA
compliant. I just roll my eyes...
On Mar 30, 2015 1:18 PM, "Steven Schoch" <schoch+freeswitch.org at xwin32.com>
wrote:

> We're using a Grandstream HT502 to send from a FAX machine. It works
> sometimes. Fortunately, we don't send many FAXes, but instead scan to a PDF
> and email it.
>
> We still have a POTS line that we use to receive FAXes (using HylaFax).
> That works well, but I'd like to get rid of the POTS line eventually. I
> have to be careful, however, because some customers still insist on FAXing
> purchase orders.
>
> --
> Steve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/a5082ea2/attachment.html 

From krice at freeswitch.org  Mon Mar 30 22:53:38 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 30 Mar 2015 18:53:38 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March
	21st-27th
Message-ID: <55199bb2b56bc_ad02777338218ba@resque-worker-ip-10-5-152-213.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/19qOGWv
FreeSWITCH Week in Review (Master Branch) March 21st-27th
Hello, again. This passed week in the FreeSWITCH master branch we had 18 commits. The features for this week include: support for mute in mod_verto and support for filtering on file:func to mod_logfile mapping.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7401 Add support for mute in mod_verto

FS-7402 Add support for filtering on file:func to mod_logfile mapping

Improvements in build system, cross platform support, and packaging:

FS-7383 Fixed a memory leak causing building Freeswitch with clang AddressSanitizer enabled to fail

FS-7149 The Windows portion of the update to use flite-2.0.0-release

The following bugs were squashed:

FS-7300 Fixed mod_rayo to properly handle errors in mod_unimrcp

FS-7385 Fixed a segfault caused by an invalid contact URI and instead fail the call with 502 bad gateway if the outbound leg returns a redirect with an invalid URI.

FS-6954 Detect when we have T.38 in nomedia or proxy media mode and apply same fixups as in media mode.

FS-7391 Corrected currency for mod_say_de it?s now ?Ein Euro und Ein Cent? rather than ?Eins Euro und Eins Cent?

FS-7396 Fix for WebRTC call setup hang when Chrome 42 includes IPv6 ICE candidate

FS-7386 Allows auto-adjust to fix media endpoints after receiving a 183 then a 200 from a different media endpoint, like in the case of proxy forking scenarios

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/ea0c881a/attachment.html 

From raphael.lechner at gmail.com  Mon Mar 30 23:20:08 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Mon, 30 Mar 2015 21:20:08 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
Message-ID: <EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>

Thank you Chris for sharing your lua files with us.

I tried but probably I missed something, because I got that error:
EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua vm_blf_extension confirmed)
2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
stack traceback:
	scripts/presence.lua:2: in main chunk
	[C]: in function 'dofile'
	/usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk

Any idea?

I have named the files set_presence.lua and presence.lua(for the functions) and both are located under scripts.

Thanks!
 

> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
> 
> This can be done with lua, what i've done is created a small lua set of functions that make presence easy to handle.
> 
> here's the presence function file: http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq <http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq>
> 
> then you setup a lua file (call it scripts/set_presence.lua) that is similar to the following
> 
> dofile "presence.lua"
> 
> state = argv[2]
> user = argv[1]
> domain = session:getVariable('domain')
> 
> local p = Presence:new()
> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
> 
> p:set(state,false);
> 
> you can execute this by adding the following line into your extension that controls it.
> 
> <action application="lua" data="set_presence.lua vm_blf_extension confirmed"/>
> 
> you can swap out the "vm_blf_extension" with anything you want, and as long as the phone "Subscribes" to this, it will work.
> 
> You can use early, confirmed, terminated for the statuses.
> 
> 
> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>> wrote:
> There's always the SEND_PRESENCE esl event...
> 
> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they had also a lamp for indicating if the ?today closed? playback or call forward to voicemail is enabled or not.
> 
> Raphael
> 
> 
>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com <mailto:sdevoy at bizfocused.com>> wrote:
>> 
>> I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.
>>  
>> Sean
>>  
>> 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 Paul Atreides
>> Sent: Monday, March 30, 2015 10:32 AM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Changing BLF lamp persistently
>>  
>> I want to have an indicator if the company voice mail is active or not.
>> 
>> At the moment I am calling a number and setting a global variable to activate the central company voice mail.
>> But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
>> active or not.
>> 
>>  
>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.
>>  
>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com <mailto:paul.atreides83 at googlemail.com>> wrote:
>> Hi,
>> 
>> does someone know how to change the BLF lamp persistently? I found the channel variable
>> presence id but this one will only last as long as the channel is active. Is there a way to change
>> it permerently?
>> 
>> Thanks for helping.
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/3beb523a/attachment-0001.html 

From magnus.kelly at gmail.com  Mon Mar 30 23:43:23 2015
From: magnus.kelly at gmail.com (Magnus Kelly)
Date: Mon, 30 Mar 2015 20:43:23 +0100
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
 March 21st-27th
In-Reply-To: <55199bb2b56bc_ad02777338218ba@resque-worker-ip-10-5-152-213.mail>
References: <55199bb2b56bc_ad02777338218ba@resque-worker-ip-10-5-152-213.mail>
Message-ID: <D13F6580.E96CC%magnus.kelly@gmail.com>

Hello Ken,

Looks good ? it might be silly question but do you now push out so one can
use likes of yum or apt-get ?

Regards
Magnus


From:  Ken Rice <krice at freeswitch.org>
Reply-To:  FreeSWITCH Help <freeswitch-users at lists.freeswitch.org>
Date:  Monday, 30 March 2015 19:53
To:  FreeSWITCH Help <freeswitch-users at lists.freeswitch.org>
Subject:  [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March
21st-27th

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/19qOGWv
FreeSWITCH Week in Review (Master Branch) March 21st-27th

Hello, again. This passed week in the FreeSWITCH master branch we had 18
commits. The features for this week include: support for mute in mod_verto
and support for filtering on file:func to mod_logfile mapping.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head
over to freeswitch.com <https://freeswitch.com/>  to learn more about
FreeSWITCH support.

New features that were added:
* FS-7401 <http://ift.tt/19qOGWz>  Add support for mute in mod_verto
* FS-7402 <http://ift.tt/1DlAwDQ>  Add support for filtering on file:func to
mod_logfile mapping
Improvements in build system, cross platform support, and packaging:
* FS-7383 <http://ift.tt/19qOJBv>  Fixed a memory leak causing building
Freeswitch with clang AddressSanitizer enabled to fail
* FS-7149 <http://ift.tt/1Gl1nRp>  The Windows portion of the update to use
flite-2.0.0-release
The following bugs were squashed:
* FS-7300 <http://ift.tt/1Gep0Hk>  Fixed mod_rayo to properly handle errors
in mod_unimrcp
* FS-7385 <http://ift.tt/1DlAz2d>  Fixed a segfault caused by an invalid
contact URI and instead fail the call with 502 bad gateway if the outbound
leg returns a redirect with an invalid URI.
* FS-6954 <http://ift.tt/1yzQC5V>  Detect when we have T.38 in nomedia or
proxy media mode and apply same fixups as in media mode.
* FS-7391 <http://ift.tt/19qOJBD>  Corrected currency for mod_say_de it?s
now ?Ein Euro und Ein Cent? rather than ?Eins Euro und Eins Cent?
* FS-7396 <http://ift.tt/1DlAz2h>  Fix for WebRTC call setup hang when
Chrome 42 includes IPv6 ICE candidate
* FS-7386 <http://ift.tt/19qOJBG>  Allows auto-adjust to fix media endpoints
after receiving a 183 then a 200 from a different media endpoint, like in
the case of proxy forking scenarios

_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
http://www.freeswitchsolutions.com Official FreeSWITCH Sites
http://www.freeswitch.org http://confluence.freeswitch.org
http://www.cluecon.com FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/adc19a87/attachment.html 

From krice at freeswitch.org  Tue Mar 31 00:00:52 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 30 Mar 2015 15:00:52 -0500
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
 March 21st-27th
In-Reply-To: <D13F6580.E96CC%magnus.kelly@gmail.com>
Message-ID: <D13F15A4.99B4B%krice@freeswitch.org>

This is just a review of the commits that went into master. If you want to
get these you?ll have to build from git. The debian and yum repos are only
updated on releases

K


On 3/30/15, 2:43 PM, "Magnus Kelly" <magnus.kelly at gmail.com> wrote:

> Hello Ken,
> 
> Looks good ? it might be silly question but do you now push out so one can use
> likes of yum or apt-get ?
> 
> Regards
> Magnus
> 
> 
> From:  Ken Rice <krice at freeswitch.org>
> Reply-To:  FreeSWITCH Help <freeswitch-users at lists.freeswitch.org>
> Date:  Monday, 30 March 2015 19:53
> To:  FreeSWITCH Help <freeswitch-users at lists.freeswitch.org>
> Subject:  [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) March
> 21st-27th
> 
> New Post on freeswitch.org from Kathleen
> check it out at http://ift.tt/19qOGWv
> FreeSWITCH Week in Review (Master Branch) March 21st-27th
> 
> Hello, again. This passed week in the FreeSWITCH master branch we had 18
> commits. The features for this week include: support for mute in mod_verto and
> support for filtering on file:func to mod_logfile mapping.
> 
> Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over
> to freeswitch.com <https://freeswitch.com/>  to learn more about FreeSWITCH
> support.
> 
> New features that were added:
> * FS-7401 <http://ift.tt/19qOGWz>  Add support for mute in mod_verto
> * FS-7402 <http://ift.tt/1DlAwDQ>  Add support for filtering on file:func to
> mod_logfile mapping
> Improvements in build system, cross platform support, and packaging:
> * FS-7383 <http://ift.tt/19qOJBv>  Fixed a memory leak causing building
> Freeswitch with clang AddressSanitizer enabled to fail
> * FS-7149 <http://ift.tt/1Gl1nRp>  The Windows portion of the update to use
> flite-2.0.0-release
> The following bugs were squashed:
> * FS-7300 <http://ift.tt/1Gep0Hk>  Fixed mod_rayo to properly handle errors in
> mod_unimrcp 
> * FS-7385 <http://ift.tt/1DlAz2d>  Fixed a segfault caused by an invalid
> contact URI and instead fail the call with 502 bad gateway if the outbound leg
> returns a redirect with an invalid URI.
> * FS-6954 <http://ift.tt/1yzQC5V>  Detect when we have T.38 in nomedia or
> proxy media mode and apply same fixups as in media mode.
> * FS-7391 <http://ift.tt/19qOJBD>  Corrected currency for mod_say_de it?s now
> ?Ein Euro und Ein Cent? rather than ?Eins Euro und Eins Cent?
> * FS-7396 <http://ift.tt/1DlAz2h>  Fix for WebRTC call setup hang when Chrome
> 42 includes IPv6 ICE candidate
> * FS-7386 <http://ift.tt/19qOJBG>  Allows auto-adjust to fix media endpoints
> after receiving a 183 then a 200 from a different media endpoint, like in the
> case of proxy forking scenarios
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
> http://www.freeswitch.org http://confluence.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Twitter: @FreeSWITCH


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/f3681140/attachment.html 

From brian at freeswitch.org  Tue Mar 31 00:05:44 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 30 Mar 2015 15:05:44 -0500
Subject: [Freeswitch-users] Fax....How I hate you.
In-Reply-To: <CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>
References: <CADK-Bju4y1idRt_voi7M9Ek+hAU1OCWnzmz7moM1toUwcBbWWQ@mail.gmail.com>
	<CAKzWOxWKmq=4DwWG5DDnCm0BcbZ98QP_2jTVu=XZ-Lwmoe+J+g@mail.gmail.com>
	<CADK-BjtnsVQB7564gYkfONpcxjBL2D6HRUyk6hPvn4uWv=Mc9g@mail.gmail.com>
	<CADK-BjvS4T3_LHWD9XnO+B9L+BiPtrfbk6K3Ab80HROvC=P4RA@mail.gmail.com>
Message-ID: <CAEJMVkA0juMgHOnik9eu2n3wJrn7jOG=R==SfZ4-eeWBdKf=rw@mail.gmail.com>

Doing t.38?  Whats the log say?

On Mon, Mar 30, 2015 at 12:13 PM, Blake Priddy <bpriddy at bryantschools.org>
wrote:

> Didnt mean to send that just yet....
>
> On Mon, Mar 30, 2015 at 12:12 PM, Blake Priddy <bpriddy at bryantschools.org>
> wrote:
>
>>  condition  context  public  10
>> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>
>> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=22789633-b660-42bf-8f30-cc162d7db149&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>   condition  destination_number  ^(15012449620)$  20
>> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>
>> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=5cca2068-0e79-4433-953b-52b2e9375d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>   action  set  fax_enable_t38=true  26
>> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>
>> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=fc82bce2-3fc3-4d23-84a8-d5cae2087d94&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>   action  set  fax_enable_t38_request=true  27
>> <https://10.208.140.10/app/dialplan/dialplan_detail_edit.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>
>> <https://10.208.140.10/app/dialplan/dialplan_detail_delete.php?id=71d6fa72-7e1e-40f1-b13a-9130cae9c891&id2=5d6954c7-392d-4d45-8869-37bbde6ccca1&app_uuid=c03b422e-13a8-bd1b-e42b-b6b9b4d27ce4>
>>   action  transfer  109 XML default
>>
>> On Mon, Mar 30, 2015 at 11:59 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>
>>> Well said! Faxing is evil, especially in a VoIP environment.
>>>
>>> To add to what Brian said, I think we've seen some success with version
>>> 1.3.1 of the SPA firmware for that device.
>>>
>>> -MC
>>>
>>>
>>> On Mon, Mar 30, 2015 at 7:43 AM, Blake Priddy <bpriddy at bryantschools.org
>>> > wrote:
>>>
>>>> Is anyone out there having success with sending/receiving faxes with
>>>> Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I
>>>> have referred myself to this document
>>>> https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I
>>>> was just wanting to see what everyone is doing out there with flowroute for
>>>> the people who still must use faxing..
>>>>
>>>> --
>>>>
>>>>
>>>> *Blakelund Priddy*
>>>> Network & Systems Engineer
>>>> Bryant Public School District
>>>> Bryant, Arkansas 72022
>>>> http://www.bryantschools.org
>>>> p 501-653-5038
>>>> f 501-847-5656
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>> *Blakelund Priddy*
>> Network & Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org
>> p 501-653-5038
>> f 501-847-5656
>>
>
>
>
> --
>
>
> *Blakelund Priddy*
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
> p 501-653-5038
> f 501-847-5656
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/8c273a03/attachment-0001.html 

From blasterjr at gmail.com  Tue Mar 31 00:19:59 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Mon, 30 Mar 2015 14:19:59 -0600
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
Message-ID: <CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>

try changing the dofile line to the full path to the file, which is
probably /usr/local/freeswitch/scripts/presence.lua

so the line would look like this

dofile "/usr/local/freeswitch/scripts/presence.lua"

On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <raphael.lechner at gmail.com>
wrote:

> Thank you Chris for sharing your lua files with us.
>
> I tried but probably I missed something, because I got that error:
> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
> vm_blf_extension confirmed)
> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2:
> attempt to index global 'Presence' (a nil value)
> stack traceback:
> scripts/presence.lua:2: in main chunk
> [C]: in function 'dofile'
> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>
> Any idea?
>
> I have named the files set_presence.lua and presence.lua(for the
> functions) and both are located under scripts.
>
> Thanks!
>
>
> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>
> This can be done with lua, what i've done is created a small lua set of
> functions that make presence easy to handle.
>
> here's the presence function file:
> http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq
>
> then you setup a lua file (call it scripts/set_presence.lua) that is
> similar to the following
>
> dofile "presence.lua"
>
> state = argv[2]
> user = argv[1]
> domain = session:getVariable('domain')
>
> local p = Presence:new()
> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>
> p:set(state,false);
>
> you can execute this by adding the following line into your extension that
> controls it.
>
> <action application="lua" data="set_presence.lua vm_blf_extension
> confirmed"/>
>
> you can swap out the "vm_blf_extension" with anything you want, and as
> long as the phone "Subscribes" to this, it will work.
>
> You can use early, confirmed, terminated for the statuses.
>
>
> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> There's always the SEND_PRESENCE esl event...
>>
>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
>> wrote:
>>
>>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they
>>> had also a lamp for indicating if the ?today closed? playback or call
>>> forward to voicemail is enabled or not.
>>>
>>> Raphael
>>>
>>>
>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>>
>>> I have a customer who had this same request as well as one for a lamp
>>> indicating a call parked using ?valet parking?.  They are worried a parked
>>> call could get forgotten.  I would love to hear a solution to either/both.
>>>
>>> Sean
>>>
>>> *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 *Paul
>>> Atreides
>>> *Sent:* Monday, March 30, 2015 10:32 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>>>
>>> I want to have an indicator if the company voice mail is active or not.
>>>
>>> At the moment I am calling a number and setting a global variable to
>>> activate the central company voice mail.
>>> But I would like to use the BLF from a dummy account so that the user
>>> can see at the phone whether the voice mail is
>>>
>>> active or not.
>>>
>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> Can you describe a bit more exactly what you are trying to accomplish?
>>> Presence changes in reaction to events that happen in calls.
>>>
>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
>>> paul.atreides83 at googlemail.com> wrote:
>>>
>>> Hi,
>>>
>>> does someone know how to change the BLF lamp persistently? I found the
>>> channel variable
>>> presence id but this one will only last as long as the channel is
>>> active. Is there a way to change
>>> it permerently?
>>>
>>> Thanks for helping.
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/0555f223/attachment.html 

From raphael.lechner at gmail.com  Tue Mar 31 00:34:33 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Mon, 30 Mar 2015 22:34:33 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
	<CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
Message-ID: <509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>

still same error. It?s normal that when I call the presence.lua file directly with lua I got the same error?

tested with lua5.1 and lua5.2
root at rapbx:/usr/local/freeswitch# lua /usr/local/freeswitch/scripts/presence.lua
lua: /usr/local/freeswitch/scripts/presence.lua:1: attempt to index global 'Presence' (a nil value)
stack traceback:
	/usr/local/freeswitch/scripts/presence.lua:1: in main chunk
	[C]: in ?

calling via diaplan
EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua vm_blf_extension confirmed)
2015-03-30 22:25:57.535504 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
stack traceback:
	/usr/local/freeswitch/scripts/presence.lua:2: in main chunk
	[C]: in function 'dofile'
	/usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk

set_presence.lua -> https://pastebin.freeswitch.org/24060 <https://pastebin.freeswitch.org/24060>
presence.lua -> https://pastebin.freeswitch.org/24061 <https://pastebin.freeswitch.org/24061>

Thanks

> On 30 Mar 2015, at 22:19, Chris Tunbridge <blasterjr at gmail.com> wrote:
> 
> try changing the dofile line to the full path to the file, which is probably /usr/local/freeswitch/scripts/presence.lua
> 
> so the line would look like this
> 
> dofile "/usr/local/freeswitch/scripts/presence.lua"
> 
> On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
> Thank you Chris for sharing your lua files with us.
> 
> I tried but probably I missed something, because I got that error:
> EXECUTE sofia/internal/91 at 192.168.130.12 <mailto:sofia/internal/91 at 192.168.130.12> lua(set_presence.lua vm_blf_extension confirmed)
> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
> stack traceback:
> 	scripts/presence.lua:2: in main chunk
> 	[C]: in function 'dofile'
> 	/usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
> 
> Any idea?
> 
> I have named the files set_presence.lua and presence.lua(for the functions) and both are located under scripts.
> 
> Thanks!
>  
> 
>> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
>> 
>> This can be done with lua, what i've done is created a small lua set of functions that make presence easy to handle.
>> 
>> here's the presence function file: http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq <http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq>
>> 
>> then you setup a lua file (call it scripts/set_presence.lua) that is similar to the following
>> 
>> dofile "presence.lua"
>> 
>> state = argv[2]
>> user = argv[1]
>> domain = session:getVariable('domain')
>> 
>> local p = Presence:new()
>> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>> 
>> p:set(state,false);
>> 
>> you can execute this by adding the following line into your extension that controls it.
>> 
>> <action application="lua" data="set_presence.lua vm_blf_extension confirmed"/>
>> 
>> you can swap out the "vm_blf_extension" with anything you want, and as long as the phone "Subscribes" to this, it will work.
>> 
>> You can use early, confirmed, terminated for the statuses.
>> 
>> 
>> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>> wrote:
>> There's always the SEND_PRESENCE esl event...
>> 
>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they had also a lamp for indicating if the ?today closed? playback or call forward to voicemail is enabled or not.
>> 
>> Raphael
>> 
>> 
>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com <mailto:sdevoy at bizfocused.com>> wrote:
>>> 
>>> I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.
>>>  
>>> Sean
>>>  
>>> 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 Paul Atreides
>>> Sent: Monday, March 30, 2015 10:32 AM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] Changing BLF lamp persistently
>>>  
>>> I want to have an indicator if the company voice mail is active or not.
>>> 
>>> At the moment I am calling a number and setting a global variable to activate the central company voice mail.
>>> But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
>>> active or not.
>>> 
>>>  
>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.
>>>  
>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com <mailto:paul.atreides83 at googlemail.com>> wrote:
>>> Hi,
>>> 
>>> does someone know how to change the BLF lamp persistently? I found the channel variable
>>> presence id but this one will only last as long as the channel is active. Is there a way to change
>>> it permerently?
>>> 
>>> Thanks for helping.
>>> 
> 
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/e6e3a9b5/attachment-0001.html 

From blasterjr at gmail.com  Tue Mar 31 00:52:36 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Mon, 30 Mar 2015 14:52:36 -0600
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
	<CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
	<509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>
Message-ID: <CAO3hyzFOOqhoWt8NL4OWHTZk__gkyOC+AmRGJTZC8zCJeQa-rg@mail.gmail.com>

When i run lua /usr/local/freeswitch/scripts/presence.lua i get an empty
return with no errors, can you do lua -v so i can see what version of lua
you're on?

Also which version of FreeSWITCH are you running?

On Mon, Mar 30, 2015 at 2:34 PM, Raphael Lechner <raphael.lechner at gmail.com>
wrote:

> still same error. It?s normal that when I call the presence.lua file
> directly with lua I got the same error?
>
> tested with lua5.1 and lua5.2
> root at rapbx:/usr/local/freeswitch# lua
> /usr/local/freeswitch/scripts/presence.lua
> lua: /usr/local/freeswitch/scripts/presence.lua:1: attempt to index global
> 'Presence' (a nil value)
> stack traceback:
> /usr/local/freeswitch/scripts/presence.lua:1: in main chunk
> [C]: in ?
>
> calling via diaplan
> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
> vm_blf_extension confirmed)
> 2015-03-30 22:25:57.535504 [ERR] mod_lua.cpp:203
> /usr/local/freeswitch/scripts/presence.lua:2: attempt to index global
> 'Presence' (a nil value)
> stack traceback:
> /usr/local/freeswitch/scripts/presence.lua:2: in main chunk
> [C]: in function 'dofile'
> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>
> set_presence.lua -> https://pastebin.freeswitch.org/24060
> presence.lua -> https://pastebin.freeswitch.org/24061
>
> Thanks
>
> On 30 Mar 2015, at 22:19, Chris Tunbridge <blasterjr at gmail.com> wrote:
>
> try changing the dofile line to the full path to the file, which is
> probably /usr/local/freeswitch/scripts/presence.lua
>
> so the line would look like this
>
> dofile "/usr/local/freeswitch/scripts/presence.lua"
>
> On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <
> raphael.lechner at gmail.com> wrote:
>
>> Thank you Chris for sharing your lua files with us.
>>
>> I tried but probably I missed something, because I got that error:
>> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
>> vm_blf_extension confirmed)
>> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2:
>> attempt to index global 'Presence' (a nil value)
>> stack traceback:
>> scripts/presence.lua:2: in main chunk
>> [C]: in function 'dofile'
>> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>>
>> Any idea?
>>
>> I have named the files set_presence.lua and presence.lua(for the
>> functions) and both are located under scripts.
>>
>> Thanks!
>>
>>
>> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>>
>> This can be done with lua, what i've done is created a small lua set of
>> functions that make presence easy to handle.
>>
>> here's the presence function file:
>> http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq
>>
>> then you setup a lua file (call it scripts/set_presence.lua) that is
>> similar to the following
>>
>> dofile "presence.lua"
>>
>> state = argv[2]
>> user = argv[1]
>> domain = session:getVariable('domain')
>>
>> local p = Presence:new()
>> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>>
>> p:set(state,false);
>>
>> you can execute this by adding the following line into your extension
>> that controls it.
>>
>> <action application="lua" data="set_presence.lua vm_blf_extension
>> confirmed"/>
>>
>> you can swap out the "vm_blf_extension" with anything you want, and as
>> long as the phone "Subscribes" to this, it will work.
>>
>> You can use early, confirmed, terminated for the statuses.
>>
>>
>> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> There's always the SEND_PRESENCE esl event...
>>>
>>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
>>> wrote:
>>>
>>>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they
>>>> had also a lamp for indicating if the ?today closed? playback or call
>>>> forward to voicemail is enabled or not.
>>>>
>>>> Raphael
>>>>
>>>>
>>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>>>
>>>> I have a customer who had this same request as well as one for a lamp
>>>> indicating a call parked using ?valet parking?.  They are worried a parked
>>>> call could get forgotten.  I would love to hear a solution to either/both.
>>>>
>>>> Sean
>>>>
>>>> *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 *Paul
>>>> Atreides
>>>> *Sent:* Monday, March 30, 2015 10:32 AM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>>>>
>>>> I want to have an indicator if the company voice mail is active or not.
>>>>
>>>> At the moment I am calling a number and setting a global variable to
>>>> activate the central company voice mail.
>>>> But I would like to use the BLF from a dummy account so that the user
>>>> can see at the phone whether the voice mail is
>>>>
>>>> active or not.
>>>>
>>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>> Can you describe a bit more exactly what you are trying to accomplish?
>>>> Presence changes in reaction to events that happen in calls.
>>>>
>>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
>>>> paul.atreides83 at googlemail.com> wrote:
>>>>
>>>> Hi,
>>>>
>>>> does someone know how to change the BLF lamp persistently? I found the
>>>> channel variable
>>>> presence id but this one will only last as long as the channel is
>>>> active. Is there a way to change
>>>> it permerently?
>>>>
>>>> Thanks for helping.
>>>>
>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/03d82e9a/attachment.html 

From blasterjr at gmail.com  Tue Mar 31 00:55:11 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Mon, 30 Mar 2015 14:55:11 -0600
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAO3hyzFOOqhoWt8NL4OWHTZk__gkyOC+AmRGJTZC8zCJeQa-rg@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
	<CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
	<509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>
	<CAO3hyzFOOqhoWt8NL4OWHTZk__gkyOC+AmRGJTZC8zCJeQa-rg@mail.gmail.com>
Message-ID: <CAO3hyzFYuKdCUrB5ObFH9aiptqJ8jaobVrf5ymuNeXj6pVotRg@mail.gmail.com>

Actually on your paste of the presence.lua file, i noticed that you're
missing Presence = {} at the top, it should be the first item.

On Mon, Mar 30, 2015 at 2:52 PM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> When i run lua /usr/local/freeswitch/scripts/presence.lua i get an empty
> return with no errors, can you do lua -v so i can see what version of lua
> you're on?
>
> Also which version of FreeSWITCH are you running?
>
>
> On Mon, Mar 30, 2015 at 2:34 PM, Raphael Lechner <
> raphael.lechner at gmail.com> wrote:
>
>> still same error. It?s normal that when I call the presence.lua file
>> directly with lua I got the same error?
>>
>> tested with lua5.1 and lua5.2
>> root at rapbx:/usr/local/freeswitch# lua
>> /usr/local/freeswitch/scripts/presence.lua
>> lua: /usr/local/freeswitch/scripts/presence.lua:1: attempt to index
>> global 'Presence' (a nil value)
>> stack traceback:
>> /usr/local/freeswitch/scripts/presence.lua:1: in main chunk
>> [C]: in ?
>>
>> calling via diaplan
>> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
>> vm_blf_extension confirmed)
>> 2015-03-30 22:25:57.535504 [ERR] mod_lua.cpp:203
>> /usr/local/freeswitch/scripts/presence.lua:2: attempt to index global
>> 'Presence' (a nil value)
>> stack traceback:
>> /usr/local/freeswitch/scripts/presence.lua:2: in main chunk
>> [C]: in function 'dofile'
>> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>>
>> set_presence.lua -> https://pastebin.freeswitch.org/24060
>> presence.lua -> https://pastebin.freeswitch.org/24061
>>
>> Thanks
>>
>> On 30 Mar 2015, at 22:19, Chris Tunbridge <blasterjr at gmail.com> wrote:
>>
>> try changing the dofile line to the full path to the file, which is
>> probably /usr/local/freeswitch/scripts/presence.lua
>>
>> so the line would look like this
>>
>> dofile "/usr/local/freeswitch/scripts/presence.lua"
>>
>> On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <
>> raphael.lechner at gmail.com> wrote:
>>
>>> Thank you Chris for sharing your lua files with us.
>>>
>>> I tried but probably I missed something, because I got that error:
>>> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
>>> vm_blf_extension confirmed)
>>> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2:
>>> attempt to index global 'Presence' (a nil value)
>>> stack traceback:
>>> scripts/presence.lua:2: in main chunk
>>> [C]: in function 'dofile'
>>> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>>>
>>> Any idea?
>>>
>>> I have named the files set_presence.lua and presence.lua(for the
>>> functions) and both are located under scripts.
>>>
>>> Thanks!
>>>
>>>
>>> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>>>
>>> This can be done with lua, what i've done is created a small lua set of
>>> functions that make presence easy to handle.
>>>
>>> here's the presence function file:
>>> http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq
>>>
>>> then you setup a lua file (call it scripts/set_presence.lua) that is
>>> similar to the following
>>>
>>> dofile "presence.lua"
>>>
>>> state = argv[2]
>>> user = argv[1]
>>> domain = session:getVariable('domain')
>>>
>>> local p = Presence:new()
>>> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>>>
>>> p:set(state,false);
>>>
>>> you can execute this by adding the following line into your extension
>>> that controls it.
>>>
>>> <action application="lua" data="set_presence.lua vm_blf_extension
>>> confirmed"/>
>>>
>>> you can swap out the "vm_blf_extension" with anything you want, and as
>>> long as the phone "Subscribes" to this, it will work.
>>>
>>> You can use early, confirmed, terminated for the statuses.
>>>
>>>
>>> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <
>>> andrew at cassidywebservices.co.uk> wrote:
>>>
>>>> There's always the SEND_PRESENCE esl event...
>>>>
>>>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
>>>> wrote:
>>>>
>>>>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and
>>>>> they had also a lamp for indicating if the ?today closed? playback or call
>>>>> forward to voicemail is enabled or not.
>>>>>
>>>>> Raphael
>>>>>
>>>>>
>>>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>>>>
>>>>> I have a customer who had this same request as well as one for a lamp
>>>>> indicating a call parked using ?valet parking?.  They are worried a parked
>>>>> call could get forgotten.  I would love to hear a solution to either/both.
>>>>>
>>>>> Sean
>>>>>
>>>>> *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 *Paul
>>>>> Atreides
>>>>> *Sent:* Monday, March 30, 2015 10:32 AM
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>>>>>
>>>>> I want to have an indicator if the company voice mail is active or not.
>>>>>
>>>>> At the moment I am calling a number and setting a global variable to
>>>>> activate the central company voice mail.
>>>>> But I would like to use the BLF from a dummy account so that the user
>>>>> can see at the phone whether the voice mail is
>>>>>
>>>>> active or not.
>>>>>
>>>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>> Can you describe a bit more exactly what you are trying to
>>>>> accomplish?  Presence changes in reaction to events that happen in calls.
>>>>>
>>>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
>>>>> paul.atreides83 at googlemail.com> wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> does someone know how to change the BLF lamp persistently? I found the
>>>>> channel variable
>>>>> presence id but this one will only last as long as the channel is
>>>>> active. Is there a way to change
>>>>> it permerently?
>>>>>
>>>>> Thanks for helping.
>>>>>
>>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/08485ac8/attachment-0001.html 

From blasterjr at gmail.com  Tue Mar 31 00:55:30 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Mon, 30 Mar 2015 14:55:30 -0600
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAO3hyzFYuKdCUrB5ObFH9aiptqJ8jaobVrf5ymuNeXj6pVotRg@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
	<CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
	<509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>
	<CAO3hyzFOOqhoWt8NL4OWHTZk__gkyOC+AmRGJTZC8zCJeQa-rg@mail.gmail.com>
	<CAO3hyzFYuKdCUrB5ObFH9aiptqJ8jaobVrf5ymuNeXj6pVotRg@mail.gmail.com>
Message-ID: <CAO3hyzGFRA24-ZVx37tx8g+poMPdszLmoiKoAk6qbP2r=fb6bQ@mail.gmail.com>

And that would be because my paste was also missing it, i apologize for
that.

On Mon, Mar 30, 2015 at 2:55 PM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> Actually on your paste of the presence.lua file, i noticed that you're
> missing Presence = {} at the top, it should be the first item.
>
> On Mon, Mar 30, 2015 at 2:52 PM, Chris Tunbridge <blasterjr at gmail.com>
> wrote:
>
>> When i run lua /usr/local/freeswitch/scripts/presence.lua i get an empty
>> return with no errors, can you do lua -v so i can see what version of lua
>> you're on?
>>
>> Also which version of FreeSWITCH are you running?
>>
>>
>> On Mon, Mar 30, 2015 at 2:34 PM, Raphael Lechner <
>> raphael.lechner at gmail.com> wrote:
>>
>>> still same error. It?s normal that when I call the presence.lua file
>>> directly with lua I got the same error?
>>>
>>> tested with lua5.1 and lua5.2
>>> root at rapbx:/usr/local/freeswitch# lua
>>> /usr/local/freeswitch/scripts/presence.lua
>>> lua: /usr/local/freeswitch/scripts/presence.lua:1: attempt to index
>>> global 'Presence' (a nil value)
>>> stack traceback:
>>> /usr/local/freeswitch/scripts/presence.lua:1: in main chunk
>>> [C]: in ?
>>>
>>> calling via diaplan
>>> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
>>> vm_blf_extension confirmed)
>>> 2015-03-30 22:25:57.535504 [ERR] mod_lua.cpp:203
>>> /usr/local/freeswitch/scripts/presence.lua:2: attempt to index global
>>> 'Presence' (a nil value)
>>> stack traceback:
>>> /usr/local/freeswitch/scripts/presence.lua:2: in main chunk
>>> [C]: in function 'dofile'
>>> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>>>
>>> set_presence.lua -> https://pastebin.freeswitch.org/24060
>>> presence.lua -> https://pastebin.freeswitch.org/24061
>>>
>>> Thanks
>>>
>>> On 30 Mar 2015, at 22:19, Chris Tunbridge <blasterjr at gmail.com> wrote:
>>>
>>> try changing the dofile line to the full path to the file, which is
>>> probably /usr/local/freeswitch/scripts/presence.lua
>>>
>>> so the line would look like this
>>>
>>> dofile "/usr/local/freeswitch/scripts/presence.lua"
>>>
>>> On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <
>>> raphael.lechner at gmail.com> wrote:
>>>
>>>> Thank you Chris for sharing your lua files with us.
>>>>
>>>> I tried but probably I missed something, because I got that error:
>>>> EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua
>>>> vm_blf_extension confirmed)
>>>> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203
>>>> scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
>>>> stack traceback:
>>>> scripts/presence.lua:2: in main chunk
>>>> [C]: in function 'dofile'
>>>> /usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>>>>
>>>> Any idea?
>>>>
>>>> I have named the files set_presence.lua and presence.lua(for the
>>>> functions) and both are located under scripts.
>>>>
>>>> Thanks!
>>>>
>>>>
>>>> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com> wrote:
>>>>
>>>> This can be done with lua, what i've done is created a small lua set of
>>>> functions that make presence easy to handle.
>>>>
>>>> here's the presence function file:
>>>> http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq
>>>>
>>>> then you setup a lua file (call it scripts/set_presence.lua) that is
>>>> similar to the following
>>>>
>>>> dofile "presence.lua"
>>>>
>>>> state = argv[2]
>>>> user = argv[1]
>>>> domain = session:getVariable('domain')
>>>>
>>>> local p = Presence:new()
>>>> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>>>>
>>>> p:set(state,false);
>>>>
>>>> you can execute this by adding the following line into your extension
>>>> that controls it.
>>>>
>>>> <action application="lua" data="set_presence.lua vm_blf_extension
>>>> confirmed"/>
>>>>
>>>> you can swap out the "vm_blf_extension" with anything you want, and as
>>>> long as the phone "Subscribes" to this, it will work.
>>>>
>>>> You can use early, confirmed, terminated for the statuses.
>>>>
>>>>
>>>> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <
>>>> andrew at cassidywebservices.co.uk> wrote:
>>>>
>>>>> There's always the SEND_PRESENCE esl event...
>>>>>
>>>>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and
>>>>>> they had also a lamp for indicating if the ?today closed? playback or call
>>>>>> forward to voicemail is enabled or not.
>>>>>>
>>>>>> Raphael
>>>>>>
>>>>>>
>>>>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>>>>>
>>>>>> I have a customer who had this same request as well as one for a lamp
>>>>>> indicating a call parked using ?valet parking?.  They are worried a parked
>>>>>> call could get forgotten.  I would love to hear a solution to either/both.
>>>>>>
>>>>>> Sean
>>>>>>
>>>>>> *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 *Paul
>>>>>> Atreides
>>>>>> *Sent:* Monday, March 30, 2015 10:32 AM
>>>>>> *To:* FreeSWITCH Users Help
>>>>>> *Subject:* Re: [Freeswitch-users] Changing BLF lamp persistently
>>>>>>
>>>>>> I want to have an indicator if the company voice mail is active or
>>>>>> not.
>>>>>>
>>>>>> At the moment I am calling a number and setting a global variable to
>>>>>> activate the central company voice mail.
>>>>>> But I would like to use the BLF from a dummy account so that the user
>>>>>> can see at the phone whether the voice mail is
>>>>>>
>>>>>> active or not.
>>>>>>
>>>>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>> Can you describe a bit more exactly what you are trying to
>>>>>> accomplish?  Presence changes in reaction to events that happen in calls.
>>>>>>
>>>>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <
>>>>>> paul.atreides83 at googlemail.com> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> does someone know how to change the BLF lamp persistently? I found
>>>>>> the channel variable
>>>>>> presence id but this one will only last as long as the channel is
>>>>>> active. Is there a way to change
>>>>>> it permerently?
>>>>>>
>>>>>> Thanks for helping.
>>>>>>
>>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/43be1b8a/attachment.html 

From tfred31 at yahoo.com  Tue Mar 31 00:57:46 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Mon, 30 Mar 2015 13:57:46 -0700
Subject: [Freeswitch-users] Question re:  local_ip_v4
Message-ID: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>

We have 2 network cards on our FS server.
*  One IP goes to the 'outside' world  (IP = 107.1.nn.nn)
*  And the other IP goes to the 'inside' world  (IP = 192.168.nn.nn)

In spite of making suggested changes to   vars.xml   when FS launches it sets itself up to 'listen' to the 'outside' world (IP = 107 etc.) 
But the result of that is that none of the in-house SIP phones can register to it.

The changes I made (following suggestions found on the web)  to   vars.xml   are as follows:
   New entry:
        <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.nn.nn"/>   <!-- where nn values are 'real' values -->
   This new line was entered immediately before
       <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

Additionally another web reference suggested that the following 2 lines be added near the bottom of   vars.xml
      <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
      <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn"/>
        
PROBLEM:   None of that seems to work.   
                    Even after not only ReStarting the FS Service, but also after ReStarting the FS Server itself and then Starting the FS Service

When through FS_CLI  I run:    sofia  status     and/or   sofia status profile internal      both of them show the 107 IP instead of the 192 IP
Those changes have not caused FS to 'see' and 'listen to' the 192  port.

What am I doing wrong?

Your advice/suggestions would be greatly appreciated.

Thanks
TF





From Sharath.Kumar at meZocliq.com  Tue Mar 31 01:21:44 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 30 Mar 2015 21:21:44 +0000
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CAFiqYuk8kmyatFy-SzoRtG6xiasqasicaTaCTwtfL4Bcjz=rsA@mail.gmail.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
	<CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<55146F3B.3090605@freeswitch.org>
	<CAFiqYuk8kmyatFy-SzoRtG6xiasqasicaTaCTwtfL4Bcjz=rsA@mail.gmail.com>
Message-ID: <CY1PR0501MB109895E07917E10277E1643684F50@CY1PR0501MB1098.namprd05.prod.outlook.com>

Thanks for the help guys. I had 2 freeswitches being load balanced from a Kamailio sbc. 1 freeswitch was correctly configured while the other although had the right configuration in xmlcurl /xmlcdr did not load cdr module in module.conf!!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Friday, March 27, 2015 6:39 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

If you have submitting to a web server enabled you'll want to set log-http-and-disk to true (the default is false).

On 26 March 2015 at 20:42, I put the Who? in Mishehu <mishehu at freeswitch.org<mailto:mishehu at freeswitch.org>> wrote:
Ok, so if you set the log dir path, you are not seeing any actual files in that path?  There should be 1 xml file for each call channel that made it all the way to the CS_REPORTING state.  If you do not see that, then I'd have to guess that there is something else in your configuration that is impacting this.  If I am correct, then it would help if you would provide the config and the debug logs from the FS console or fs_cli.



--

Yossi Neiman



On 03/26/2015 03:18 PM, Sharath Kumar wrote:
Of course. I am only talking about successful calls and with the voice path present.

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 Anthony Minessale
Sent: Thursday, March 26, 2015 4:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Only if you actually have 2 legs, a failed call where the number is wrong etc will only have 1

On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
I am logging both legs. I believe the default behavior is to generate cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by commenting the entry. Neither seemed to matter. If I do see a CDR I only see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2 cdr request correct ?

Thank you Steven
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 Steven Ayre
Sent: Thursday, March 26, 2015 3:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] XML CDR unpredictable ?

Are you logging b-leg CDRs as well as a-leg?

On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
All,

I use mod_cdr_xml with xml_curl and receive messages on the backend PHP for call control logic. I am a little mystified by the XML CDR requests. I see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML requests but I do see for dialplan so the call does succeed. I don?t understand why the CDRs XML requests are intermittent? Also, I changed the config to log the b-leg cdr as well, it didn?t seem to make any impact.
So for each leg do I expect a CDR ? Also,if I do a redirect and then a bridge I only see the redirect app?s cdr and not the bridge apps. I need the bridge app?s cdr since It contains QOS metrics for that call.

A little help would be appreciated.

Thank you
Sharath

_________________________________________________________________________
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://confluence.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://confluence.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       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +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://confluence.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://confluence.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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150330/906e551d/attachment-0001.html 

From raphael.lechner at gmail.com  Tue Mar 31 12:55:34 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Tue, 31 Mar 2015 10:55:34 +0200
Subject: [Freeswitch-users] Changing BLF lamp persistently
In-Reply-To: <CAO3hyzGFRA24-ZVx37tx8g+poMPdszLmoiKoAk6qbP2r=fb6bQ@mail.gmail.com>
References: <CAP6oyxGbJxByR6WeB0+O_MnT+dZv=Dz73hBJpiTffSvtM3Aa4A@mail.gmail.com>
	<CAEwTsAjkWF4nvR4cpkkaThcQ--1+EYK4SgvxtBuHq=hiz6dMrg@mail.gmail.com>
	<CAP6oyxEo58m5A2eq4j43ZNsn=djvmdktbq31fqzW8b-wbHcADw@mail.gmail.com>
	<BY2PR0101MB093383794741368538DDD760BBF50@BY2PR0101MB0933.prod.exchangelabs.com>
	<5A134BF0-75AA-4C7B-BF62-C3E04A05321B@gmail.com>
	<CAMbW9oLU1q0YJO3oER125ebJ0-4-WqbOAc-nQmg0q_H5RG7RZA@mail.gmail.com>
	<CAO3hyzFUCoyp69JrGWejEB1pEM=aZOgNf0EQ=hE9k0KnOsSckA@mail.gmail.com>
	<EDAAFE5F-41F3-4A94-95CA-BEE1E7828ACD@gmail.com>
	<CAO3hyzGAOBSJoih4rM_AzjBjj5NHQ795jcwP3PbVAKH9qi9CzA@mail.gmail.com>
	<509BB674-0938-4470-8606-0BF2CFDCFC7D@gmail.com>
	<CAO3hyzFOOqhoWt8NL4OWHTZk__gkyOC+AmRGJTZC8zCJeQa-rg@mail.gmail.com>
	<CAO3hyzFYuKdCUrB5ObFH9aiptqJ8jaobVrf5ymuNeXj6pVotRg@mail.gmail.com>
	<CAO3hyzGFRA24-ZVx37tx8g+poMPdszLmoiKoAk6qbP2r=fb6bQ@mail.gmail.com>
Message-ID: <0CAA60F6-62C1-420C-A377-23C25C7A4A20@gmail.com>

Hi Chris,

Thank you. lua presence.lua is now working without error output.
Calling via dial plan seems to have another problem.
EXECUTE sofia/internal/91 at 192.168.130.12 lua(set_presence.lua vm_blf_extension confirmed)
2015-03-31 10:47:20.995581 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/presence.lua:6: attempt to index local 'self' (a nil value)
stack traceback:
	/usr/local/freeswitch/scripts/presence.lua:6: in function 'new'
	/usr/local/freeswitch/scripts/set_presence.lua:7: in main chunk

The line 6 contains the following:
  self.__index = self

Have you another idea?

The program versions are:
FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)
Lua 5.2.3  Copyright (C) 1994-2013 Lua.org, PUC-Rio
on ubuntu 14.04

Raphael

> On 30 Mar 2015, at 22:55, Chris Tunbridge <blasterjr at gmail.com> wrote:
> 
> And that would be because my paste was also missing it, i apologize for that.
> 
> On Mon, Mar 30, 2015 at 2:55 PM, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
> Actually on your paste of the presence.lua file, i noticed that you're missing Presence = {} at the top, it should be the first item.
> 
> On Mon, Mar 30, 2015 at 2:52 PM, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
> When i run lua /usr/local/freeswitch/scripts/presence.lua i get an empty return with no errors, can you do lua -v so i can see what version of lua you're on?
> 
> Also which version of FreeSWITCH are you running?
> 
> 
> On Mon, Mar 30, 2015 at 2:34 PM, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
> still same error. It?s normal that when I call the presence.lua file directly with lua I got the same error?
> 
> tested with lua5.1 and lua5.2
> root at rapbx:/usr/local/freeswitch# lua /usr/local/freeswitch/scripts/presence.lua
> lua: /usr/local/freeswitch/scripts/presence.lua:1: attempt to index global 'Presence' (a nil value)
> stack traceback:
> 	/usr/local/freeswitch/scripts/presence.lua:1: in main chunk
> 	[C]: in ?
> 
> calling via diaplan
> EXECUTE sofia/internal/91 at 192.168.130.12 <mailto:sofia/internal/91 at 192.168.130.12> lua(set_presence.lua vm_blf_extension confirmed)
> 2015-03-30 22:25:57.535504 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
> stack traceback:
> 	/usr/local/freeswitch/scripts/presence.lua:2: in main chunk
> 	[C]: in function 'dofile'
> 	/usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
> 
> set_presence.lua -> https://pastebin.freeswitch.org/24060 <https://pastebin.freeswitch.org/24060>
> presence.lua -> https://pastebin.freeswitch.org/24061 <https://pastebin.freeswitch.org/24061>
> 
> Thanks
> 
>> On 30 Mar 2015, at 22:19, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
>> 
>> try changing the dofile line to the full path to the file, which is probably /usr/local/freeswitch/scripts/presence.lua
>> 
>> so the line would look like this
>> 
>> dofile "/usr/local/freeswitch/scripts/presence.lua"
>> 
>> On Mon, Mar 30, 2015 at 1:20 PM, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
>> Thank you Chris for sharing your lua files with us.
>> 
>> I tried but probably I missed something, because I got that error:
>> EXECUTE sofia/internal/91 at 192.168.130.12 <mailto:sofia/internal/91 at 192.168.130.12> lua(set_presence.lua vm_blf_extension confirmed)
>> 2015-03-30 21:13:31.775263 [ERR] mod_lua.cpp:203 scripts/presence.lua:2: attempt to index global 'Presence' (a nil value)
>> stack traceback:
>> 	scripts/presence.lua:2: in main chunk
>> 	[C]: in function 'dofile'
>> 	/usr/local/freeswitch/scripts/set_presence.lua:1: in main chunk
>> 
>> Any idea?
>> 
>> I have named the files set_presence.lua and presence.lua(for the functions) and both are located under scripts.
>> 
>> Thanks!
>>  
>> 
>>> On 30 Mar 2015, at 20:43, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
>>> 
>>> This can be done with lua, what i've done is created a small lua set of functions that make presence easy to handle.
>>> 
>>> here's the presence function file: http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq <http://pastie.org/private/8qfyjbwzsmwm2a4uz7owq>
>>> 
>>> then you setup a lua file (call it scripts/set_presence.lua) that is similar to the following
>>> 
>>> dofile "presence.lua"
>>> 
>>> state = argv[2]
>>> user = argv[1]
>>> domain = session:getVariable('domain')
>>> 
>>> local p = Presence:new()
>>> p:init{user = user, domain = domain, uuid = 'custom_blf_key'};
>>> 
>>> p:set(state,false);
>>> 
>>> you can execute this by adding the following line into your extension that controls it.
>>> 
>>> <action application="lua" data="set_presence.lua vm_blf_extension confirmed"/>
>>> 
>>> you can swap out the "vm_blf_extension" with anything you want, and as long as the phone "Subscribes" to this, it will work.
>>> 
>>> You can use early, confirmed, terminated for the statuses.
>>> 
>>> 
>>> On Mon, Mar 30, 2015 at 9:37 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>> wrote:
>>> There's always the SEND_PRESENCE esl event...
>>> 
>>> On 30 March 2015 at 16:25, Raphael Lechner <raphael.lechner at gmail.com <mailto:raphael.lechner at gmail.com>> wrote:
>>> We migrated a customer from asterisk to FreeSWITCH 2 weeks ago and they had also a lamp for indicating if the ?today closed? playback or call forward to voicemail is enabled or not.
>>> 
>>> Raphael
>>> 
>>> 
>>>> On 30 Mar 2015, at 17:17, Sean Devoy <sdevoy at bizfocused.com <mailto:sdevoy at bizfocused.com>> wrote:
>>>> 
>>>> I have a customer who had this same request as well as one for a lamp indicating a call parked using ?valet parking?.  They are worried a parked call could get forgotten.  I would love to hear a solution to either/both.
>>>>  
>>>> Sean
>>>>  
>>>> 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 Paul Atreides
>>>> Sent: Monday, March 30, 2015 10:32 AM
>>>> To: FreeSWITCH Users Help
>>>> Subject: Re: [Freeswitch-users] Changing BLF lamp persistently
>>>>  
>>>> I want to have an indicator if the company voice mail is active or not.
>>>> 
>>>> At the moment I am calling a number and setting a global variable to activate the central company voice mail.
>>>> But I would like to use the BLF from a dummy account so that the user can see at the phone whether the voice mail is
>>>> active or not.
>>>> 
>>>>  
>>>> On Mon, Mar 30, 2015 at 4:08 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>> Can you describe a bit more exactly what you are trying to accomplish?  Presence changes in reaction to events that happen in calls.
>>>>  
>>>> On Mon, Mar 30, 2015 at 9:46 AM, Paul Atreides <paul.atreides83 at googlemail.com <mailto:paul.atreides83 at googlemail.com>> wrote:
>>>> Hi,
>>>> 
>>>> does someone know how to change the BLF lamp persistently? I found the channel variable
>>>> presence id but this one will only last as long as the channel is active. Is there a way to change
>>>> it permerently?
>>>> 
>>>> Thanks for helping.
>>>> 
>> 
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> 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://confluence.freeswitch.org <http://confluence.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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/af17546d/attachment-0001.html 

From vma at 440hz.fr  Tue Mar 31 13:02:07 2015
From: vma at 440hz.fr (Vallimamod Abdullah)
Date: Tue, 31 Mar 2015 11:02:07 +0200
Subject: [Freeswitch-users] Question re:  local_ip_v4
In-Reply-To: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>
References: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>
Message-ID: <3BC07844-9921-4ABF-82C3-664A6A920ECE@440hz.fr>

Hi,

Try to set the following in your internal profile:

    <param name="sip-ip" value="$${internal_sip_ip}?/>
    <param name="rtp-ip" value="$${internal_rtp_ip}"/>

along with the following in vars.xml:

     <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
     <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn?/>

Keep also in mind that you need to restart freeswitch everytime you change any X-PRE-PROCESS statement.

Best Regards,
Vallimamod
.

> On 30 Mar 2015, at 22:57, T Fred Farmington <tfred31 at yahoo.com> wrote:
> 
> We have 2 network cards on our FS server.
> *  One IP goes to the 'outside' world  (IP = 107.1.nn.nn)
> *  And the other IP goes to the 'inside' world  (IP = 192.168.nn.nn)
> 
> In spite of making suggested changes to   vars.xml   when FS launches it sets itself up to 'listen' to the 'outside' world (IP = 107 etc.) 
> But the result of that is that none of the in-house SIP phones can register to it.
> 
> The changes I made (following suggestions found on the web)  to   vars.xml   are as follows:
>   New entry:
>        <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.nn.nn"/>   <!-- where nn values are 'real' values -->
>   This new line was entered immediately before
>       <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
> 
> Additionally another web reference suggested that the following 2 lines be added near the bottom of   vars.xml
>      <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
>      <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn"/>
> 
> PROBLEM:   None of that seems to work.   
>                    Even after not only ReStarting the FS Service, but also after ReStarting the FS Server itself and then Starting the FS Service
> 
> When through FS_CLI  I run:    sofia  status     and/or   sofia status profile internal      both of them show the 107 IP instead of the 192 IP
> Those changes have not caused FS to 'see' and 'listen to' the 192  port.
> 
> What am I doing wrong?
> 
> Your advice/suggestions would be greatly appreciated.
> 
> Thanks
> TF



From brian at freeswitch.org  Tue Mar 31 17:19:34 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 31 Mar 2015 08:19:34 -0500
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <3BC07844-9921-4ABF-82C3-664A6A920ECE@440hz.fr>
References: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>
	<3BC07844-9921-4ABF-82C3-664A6A920ECE@440hz.fr>
Message-ID: <CAEJMVkCUXxOvRx+3=e3RKOQrQpTDMcYxKvkpXia3ftV6xhpggQ@mail.gmail.com>

local_ip_v4 and local_ip_v6 will always resolve to the IP on the interfaces
that can reach the internet.  Anything else such as your setup will require
you to set the values on the sofia profiles specifically.

On Tue, Mar 31, 2015 at 4:02 AM, Vallimamod Abdullah <vma at 440hz.fr> wrote:

> Hi,
>
> Try to set the following in your internal profile:
>
>     <param name="sip-ip" value="$${internal_sip_ip}?/>
>     <param name="rtp-ip" value="$${internal_rtp_ip}"/>
>
> along with the following in vars.xml:
>
>      <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
>      <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn?/>
>
> Keep also in mind that you need to restart freeswitch everytime you change
> any X-PRE-PROCESS statement.
>
> Best Regards,
> Vallimamod
> .
>
> > On 30 Mar 2015, at 22:57, T Fred Farmington <tfred31 at yahoo.com> wrote:
> >
> > We have 2 network cards on our FS server.
> > *  One IP goes to the 'outside' world  (IP = 107.1.nn.nn)
> > *  And the other IP goes to the 'inside' world  (IP = 192.168.nn.nn)
> >
> > In spite of making suggested changes to   vars.xml   when FS launches it
> sets itself up to 'listen' to the 'outside' world (IP = 107 etc.)
> > But the result of that is that none of the in-house SIP phones can
> register to it.
> >
> > The changes I made (following suggestions found on the web)  to
>  vars.xml   are as follows:
> >   New entry:
> >        <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.nn.nn"/>
>  <!-- where nn values are 'real' values -->
> >   This new line was entered immediately before
> >       <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
> >
> > Additionally another web reference suggested that the following 2 lines
> be added near the bottom of   vars.xml
> >      <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
> >      <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn"/>
> >
> > PROBLEM:   None of that seems to work.
> >                    Even after not only ReStarting the FS Service, but
> also after ReStarting the FS Server itself and then Starting the FS Service
> >
> > When through FS_CLI  I run:    sofia  status     and/or   sofia status
> profile internal      both of them show the 107 IP instead of the 192 IP
> > Those changes have not caused FS to 'see' and 'listen to' the 192  port.
> >
> > What am I doing wrong?
> >
> > Your advice/suggestions would be greatly appreciated.
> >
> > Thanks
> > TF
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/f4568c8f/attachment.html 

From brian at freeswitch.org  Tue Mar 31 17:20:17 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 31 Mar 2015 08:20:17 -0500
Subject: [Freeswitch-users] XML CDR unpredictable ?
In-Reply-To: <CY1PR0501MB109895E07917E10277E1643684F50@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB109807ADDA314012CD2D31D484080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAFiqYukLKNTJgH8N99yaE4BEeKxS+PfLWB2b083rNLyNN53RCw@mail.gmail.com>
	<CY1PR0501MB1098253AB5FA75FD5D5A553684080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAKbxfG-8cN0J-R6V=D-BXY+WA6nebnRYsobSzm1x7je6rJwOjw@mail.gmail.com>
	<CY1PR0501MB1098359E5632F1162DDA450F84080@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<55146F3B.3090605@freeswitch.org>
	<CAFiqYuk8kmyatFy-SzoRtG6xiasqasicaTaCTwtfL4Bcjz=rsA@mail.gmail.com>
	<CY1PR0501MB109895E07917E10277E1643684F50@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAEJMVkC=L+PbiEPYN8Ju7JfMWrZ3hp32Kv3BKvCPUwwqeb5T5Q@mail.gmail.com>

That sounds like a fun one to debug!  Glad you found out what was up, has
us all scratching our heads on this one.

On Mon, Mar 30, 2015 at 4:21 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Thanks for the help guys. I had 2 freeswitches being load balanced from
> a Kamailio sbc. 1 freeswitch was correctly configured while the other
> although had the right configuration in xmlcurl /xmlcdr did not load cdr
> module in module.conf!!
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Friday, March 27, 2015 6:39 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> If you have submitting to a web server enabled you'll want to
> set log-http-and-disk to true (the default is false).
>
>
>
> On 26 March 2015 at 20:42, I put the Who? in Mishehu <
> mishehu at freeswitch.org> wrote:
>
>  Ok, so if you set the log dir path, you are not seeing any actual files
> in that path?  There should be 1 xml file for each call channel that made
> it all the way to the CS_REPORTING state.  If you do not see that, then I'd
> have to guess that there is something else in your configuration that is
> impacting this.  If I am correct, then it would help if you would provide
> the config and the debug logs from the FS console or fs_cli.
>
>
>  --
>
> Yossi Neiman
>
>
>
> On 03/26/2015 03:18 PM, Sharath Kumar wrote:
>
>    Of course. I am only talking about successful calls and with the voice
> path present.
>
>
>
> *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 *Anthony
> Minessale
> *Sent:* Thursday, March 26, 2015 4:07 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> Only if you actually have 2 legs, a failed call where the number is wrong
> etc will only have 1
>
>
>
> On Thu, Mar 26, 2015 at 2:59 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>  I am logging both legs. I believe the default behavior is to generate
> cdrs for b-leg as well. I tried by adding ?log-b-leg? to true and also by
> commenting the entry. Neither seemed to matter. If I do see a CDR I only
> see 1 request. I thought by enabling both ?a? and ?b? legs I should see 2
> cdr request correct ?
>
>
>
> Thank you Steven
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
> *Sent:* Thursday, March 26, 2015 3:40 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] XML CDR unpredictable ?
>
>
>
> Are you logging b-leg CDRs as well as a-leg?
>
>
>
> On 26 March 2015 at 18:48, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>      All,
>
>
>
> I use mod_cdr_xml with xml_curl and receive messages on the backend PHP
> for call control logic. I am a little mystified by the XML CDR requests. I
> see in the logs CS_HANGUP to CS_REPORTING but I don?t always see the XML
> requests but I do see for dialplan so the call does succeed. I don?t
> understand why the CDRs XML requests are intermittent? Also, I changed the
> config to log the b-leg cdr as well, it didn?t seem to make any impact.
>
> So for each leg do I expect a CDR ? Also,if I do a redirect and then a
> bridge I only see the redirect app?s cdr and not the bridge apps. I need
> the bridge app?s cdr since It contains QOS metrics for that call.
>
>
>
> A little help would be appreciated.
>
>
>
> Thank you
>
> Sharath
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
>
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
>
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
>
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/b73b3784/attachment-0001.html 

From steveayre at gmail.com  Tue Mar 31 17:21:26 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 Mar 2015 14:21:26 +0100
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>
References: <1427749066.12206.YahooMailBasic@web160206.mail.bf1.yahoo.com>
Message-ID: <CAFiqYun7C1Jbb5CkEYub9scZyuJ7-jCngXXjGcybS37021AhHQ@mail.gmail.com>

You want to configure 2 sofia profiles, one listening on each IP.

It might be simpler to define the IPs within the profiles rather than in
vars.conf.xml (where you're just setting variables that are referenced
elsewhere).


On 30 March 2015 at 21:57, T Fred Farmington <tfred31 at yahoo.com> wrote:

> We have 2 network cards on our FS server.
> *  One IP goes to the 'outside' world  (IP = 107.1.nn.nn)
> *  And the other IP goes to the 'inside' world  (IP = 192.168.nn.nn)
>
> In spite of making suggested changes to   vars.xml   when FS launches it
> sets itself up to 'listen' to the 'outside' world (IP = 107 etc.)
> But the result of that is that none of the in-house SIP phones can
> register to it.
>
> The changes I made (following suggestions found on the web)  to
>  vars.xml   are as follows:
>    New entry:
>         <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.nn.nn"/>   <!--
> where nn values are 'real' values -->
>    This new line was entered immediately before
>        <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
> Additionally another web reference suggested that the following 2 lines be
> added near the bottom of   vars.xml
>       <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.nn.nn"/>
>       <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.nn.nn"/>
>
> PROBLEM:   None of that seems to work.
>                     Even after not only ReStarting the FS Service, but
> also after ReStarting the FS Server itself and then Starting the FS Service
>
> When through FS_CLI  I run:    sofia  status     and/or   sofia status
> profile internal      both of them show the 107 IP instead of the 192 IP
> Those changes have not caused FS to 'see' and 'listen to' the 192  port.
>
> What am I doing wrong?
>
> Your advice/suggestions would be greatly appreciated.
>
> Thanks
> TF
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/0ec3d90a/attachment.html 

From tfred31 at yahoo.com  Tue Mar 31 17:35:54 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 06:35:54 -0700
Subject: [Freeswitch-users] Question re:  local_ip_v4
In-Reply-To: <3BC07844-9921-4ABF-82C3-664A6A920ECE@440hz.fr>
Message-ID: <1427808954.95364.YahooMailBasic@web160203.mail.bf1.yahoo.com>

Hello,

I thank you for your reply.

I already had the parameters  (internal_sip_ip & internal_rtp_ip)  you describe defined in   vars.xml  
And I added the 2 new parameter definitions  (sip_ip & rtp_ip)  to    internal.xml

I then did a ReStart of FreeSWITCH
However, in spite of the definitions, the result was un-changed.
FS_CLI showed the  107.1.nn.nn  'outside' IP, but did not show the 'inside'  192.168.nn.nn  IP

Any other suggestions?

Thanks
TF

--------------------------------------------
On Tue, 3/31/15, Vallimamod Abdullah <vma at 440hz.fr> wrote:

 Subject: Re: [Freeswitch-users] Question re:  local_ip_v4
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Tuesday, March 31, 2015, 3:02 AM
 
 Hi,
 
 Try to set the following in your internal
 profile:
 
 ? ? <param
 name="sip-ip"
 value="$${internal_sip_ip}?/>
 ?
 ? <param name="rtp-ip"
 value="$${internal_rtp_ip}"/>
 
 along with the following in
 vars.xml:
 
 ?
 ???<X-PRE-PROCESS cmd="set"
 data="internal_sip_ip=192.168.nn.nn"/>
 ? ???<X-PRE-PROCESS
 cmd="set"
 data="internal_rtp_ip=192.168.nn.nn?/>
 
 Keep also in mind that you
 need to restart freeswitch everytime you change any
 X-PRE-PROCESS statement.
 
 Best Regards,
 Vallimamod
 .
 
 > On 30 Mar 2015, at
 22:57, T Fred Farmington <tfred31 at yahoo.com>
 wrote:
 > 
 > We have 2
 network cards on our FS server.
 > *? One
 IP goes to the 'outside' world? (IP =
 107.1.nn.nn)
 > *? And the other IP goes
 to the 'inside' world? (IP = 192.168.nn.nn)
 > 
 > In spite of making
 suggested changes to???vars.xml???when
 FS launches it sets itself up to 'listen' to the
 'outside' world (IP = 107 etc.) 
 > But the result of that is that none of the
 in-house SIP phones can register to it.
 >
 
 > The changes I made (following
 suggestions found on the web)?
 to???vars.xml???are as follows:
 >???New entry:
 >? ? ? ? <X-PRE-PROCESS
 cmd="set"
 data="local_ip_v4=192.168.nn.nn"/>???<!--
 where nn values are 'real' values -->
 >???This new line was entered
 immediately before
 >? ?
 ???<X-PRE-PROCESS cmd="set"
 data="domain=$${local_ip_v4}"/>
 > 
 > Additionally another
 web reference suggested that the following 2 lines be added
 near the bottom of???vars.xml
 >? ? ? <X-PRE-PROCESS
 cmd="set"
 data="internal_sip_ip=192.168.nn.nn"/>
 >? ? ? <X-PRE-PROCESS
 cmd="set"
 data="internal_rtp_ip=192.168.nn.nn"/>
 > 
 >
 PROBLEM:???None of that seems to
 work.???
 >? ? ? ? ? ? ?
 ? ? ? Even after not only ReStarting the FS Service, but
 also after ReStarting the FS Server itself and then Starting
 the FS Service
 > 
 >
 When through FS_CLI? I run:? ? sofia? status?
 ???and/or???sofia status profile
 internal? ? ? both of them show the 107 IP instead of the
 192 IP
 > Those changes have not caused FS
 to 'see' and 'listen to' the 192? port.
 > 
 > What am I doing
 wrong?
 > 
 > Your
 advice/suggestions would be greatly appreciated.
 > 
 > Thanks
 > TF
 
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


From tfred31 at yahoo.com  Tue Mar 31 18:27:29 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 07:27:29 -0700
Subject: [Freeswitch-users] Question re:  local_ip_v4
In-Reply-To: <3BC07844-9921-4ABF-82C3-664A6A920ECE@440hz.fr>
Message-ID: <1427812049.29239.YahooMailBasic@web160204.mail.bf1.yahoo.com>

An additional piece of information that may be helpful in determining what is going wrong.....

After having the FS service up and running I attempt to Register a SIP phone by doing a ReStart on it.

Its extension is  105   and there is an associated  'extension' xml file for 105 in the  conf\directory\default   directory   (note - it used to work on another FS server)

When that Registration attempt is 'seen' by FS I get the following message through FS_CLI:
      Can't find user [105 at 192.168.nn.nn]  from <the phone's IP address>
      You must define a domain called  <FS 'inside' IP address 192.168.nn.nn>  in your directory and add a user with the ID '105' attribute 
      and you must configure your device to use the proper domain in its authentication credentials

I am sure that message is telling me something important, but I thought that I had already done everything that it is now telling me to do again.
1.  I have extension 105 defined in the directory
2.  I hope to have configured FS to 'see' domain 192.168.nn.nn  as its internal SIP IP
3.  I have the phone configured to Register on FS's 'internal' domain

What is wrong?
And, specifically what do I need to do to resolve this?

Thanks
TF


     


--------------------------------------------
On Tue, 3/31/15, Vallimamod Abdullah <vma at 440hz.fr> wrote:

 Subject: Re: [Freeswitch-users] Question re:  local_ip_v4
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Tuesday, March 31, 2015, 3:02 AM
 
 Hi,
 
 Try to set the following in your internal
 profile:
 
 ? ? <param
 name="sip-ip"
 value="$${internal_sip_ip}?/>
 ?
 ? <param name="rtp-ip"
 value="$${internal_rtp_ip}"/>
 
 along with the following in
 vars.xml:
 
 ?
 ???<X-PRE-PROCESS cmd="set"
 data="internal_sip_ip=192.168.nn.nn"/>
 ? ???<X-PRE-PROCESS
 cmd="set"
 data="internal_rtp_ip=192.168.nn.nn?/>
 
 Keep also in mind that you
 need to restart freeswitch everytime you change any
 X-PRE-PROCESS statement.
 
 Best Regards,
 Vallimamod
 .
 
 > On 30 Mar 2015, at
 22:57, T Fred Farmington <tfred31 at yahoo.com>
 wrote:
 > 
 > We have 2
 network cards on our FS server.
 > *? One
 IP goes to the 'outside' world? (IP =
 107.1.nn.nn)
 > *? And the other IP goes
 to the 'inside' world? (IP = 192.168.nn.nn)
 > 
 > In spite of making
 suggested changes to???vars.xml???when
 FS launches it sets itself up to 'listen' to the
 'outside' world (IP = 107 etc.) 
 > But the result of that is that none of the
 in-house SIP phones can register to it.
 >
 
 > The changes I made (following
 suggestions found on the web)?
 to???vars.xml???are as follows:
 >???New entry:
 >? ? ? ? <X-PRE-PROCESS
 cmd="set"
 data="local_ip_v4=192.168.nn.nn"/>???<!--
 where nn values are 'real' values -->
 >???This new line was entered
 immediately before
 >? ?
 ???<X-PRE-PROCESS cmd="set"
 data="domain=$${local_ip_v4}"/>
 > 
 > Additionally another
 web reference suggested that the following 2 lines be added
 near the bottom of???vars.xml
 >? ? ? <X-PRE-PROCESS
 cmd="set"
 data="internal_sip_ip=192.168.nn.nn"/>
 >? ? ? <X-PRE-PROCESS
 cmd="set"
 data="internal_rtp_ip=192.168.nn.nn"/>
 > 
 >
 PROBLEM:???None of that seems to
 work.???
 >? ? ? ? ? ? ?
 ? ? ? Even after not only ReStarting the FS Service, but
 also after ReStarting the FS Server itself and then Starting
 the FS Service
 > 
 >
 When through FS_CLI? I run:? ? sofia? status?
 ???and/or???sofia status profile
 internal? ? ? both of them show the 107 IP instead of the
 192 IP
 > Those changes have not caused FS
 to 'see' and 'listen to' the 192? port.
 > 
 > What am I doing
 wrong?
 > 
 > Your
 advice/suggestions would be greatly appreciated.
 > 
 > Thanks
 > TF
 
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


From tfred31 at yahoo.com  Tue Mar 31 19:11:49 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 08:11:49 -0700
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <CAFiqYun7C1Jbb5CkEYub9scZyuJ7-jCngXXjGcybS37021AhHQ@mail.gmail.com>
Message-ID: <1427814709.82232.YahooMailBasic@web160203.mail.bf1.yahoo.com>

Thank you for your reply.

OK, but pardon my ignorance, but how do I do that?

I am guessing here, but does that mean that for the 'inside' IP (192.168.nn.nn) I use the    conf\sip_profiles\internal.xml  file and define   sip_ip and rtp_ip  specifically -- not using  the $$(internal_etc...) references?
If so, OK

But how do I define the 'outside' (107.1.nnn.nn) sip?
Do I do it in the    conf\sip_profiles\external.xml    file?
And, if so, what do I put there?

Thanks,
TF




--------------------------------------------
On Tue, 3/31/15, Steven Ayre <steveayre at gmail.com> wrote:

 Subject: Re: [Freeswitch-users] Question re: local_ip_v4
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Tuesday, March 31, 2015, 7:21 AM
 
 You want
 to configure 2 sofia profiles, one listening on each
 IP.
 It might be simpler to
 define the IPs within the profiles rather than in
 vars.conf.xml (where you're just setting variables that
 are referenced elsewhere).
 
 On 30 March 2015 at 21:57,
 T Fred Farmington <tfred31 at yahoo.com>
 wrote:
 We have 2
 network cards on our FS server.
 
 *? One IP goes to the 'outside' world? (IP =
 107.1.nn.nn)
 
 *? And the other IP goes to the 'inside' world?
 (IP = 192.168.nn.nn)
 
 
 
 In spite of making suggested changes to? ?vars.xml?
 ?when FS launches it sets itself up to 'listen' to
 the 'outside' world (IP = 107 etc.)
 
 But the result of that is that none of the in-house SIP
 phones can register to it.
 
 
 
 The changes I made (following suggestions found on the
 web)? to? ?vars.xml? ?are as follows:
 
 ? ?New entry:
 
 ? ? ? ? <X-PRE-PROCESS cmd="set"
 data="local_ip_v4=192.168.nn.nn"/>? ?<!--
 where nn values are 'real' values -->
 
 ? ?This new line was entered immediately before
 
 ? ? ? ?<X-PRE-PROCESS cmd="set"
 data="domain=$${local_ip_v4}"/>
 
 
 
 Additionally another web reference suggested that the
 following 2 lines be added near the bottom of?
 ?vars.xml
 
 ? ? ? <X-PRE-PROCESS cmd="set"
 data="internal_sip_ip=192.168.nn.nn"/>
 
 ? ? ? <X-PRE-PROCESS cmd="set"
 data="internal_rtp_ip=192.168.nn.nn"/>
 
 
 
 PROBLEM:? ?None of that seems to work.
 
 ? ? ? ? ? ? ? ? ? ? Even after not only ReStarting
 the FS Service, but also after ReStarting the FS Server
 itself and then Starting the FS Service
 
 
 
 When through FS_CLI? I run:? ? sofia? status? ?
 ?and/or? ?sofia status profile internal? ? ? both of
 them show the 107 IP instead of the 192 IP
 
 Those changes have not caused FS to 'see' and
 'listen to' the 192? port.
 
 
 
 What am I doing wrong?
 
 
 
 Your advice/suggestions would be greatly appreciated.
 
 
 
 Thanks
 
 TF
 
 
 
 
 
 
 
 
 
 _________________________________________________________________________
 
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 
 http://www.freeswitchsolutions.com
 
 
 
 Official FreeSWITCH Sites
 
 http://www.freeswitch.org
 
 http://confluence.freeswitch.org
 
 http://www.cluecon.com
 
 
 
 FreeSWITCH-users mailing list
 
 FreeSWITCH-users at lists.freeswitch.org
 
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 
 http://www.freeswitch.org
 
 
 
 -----Inline Attachment Follows-----
 
 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 
 consulting at freeswitch.org
 http://www.freeswitchsolutions.com
 
 Official FreeSWITCH Sites
 http://www.freeswitch.org
 http://confluence.freeswitch.org
 http://www.cluecon.com
 
 FreeSWITCH-users mailing
 list
 FreeSWITCH-users at lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


From paul.atreides83 at googlemail.com  Tue Mar 31 19:29:05 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Tue, 31 Mar 2015 17:29:05 +0200
Subject: [Freeswitch-users] Caller ID prefix missing
Message-ID: <CAP6oyxEhz80ETuJk+Lqa9b5F3iwy26A4TQzfAAw0qaXGQFTzMw@mail.gmail.com>

Hi,

when I place a call over the gateway to the PTSN I see the dialed number
without
the leading 0 prefix in the history of my Grandstream GXP2140. Is there a
way
to add the 0 back again?

Thank you

<include>
 <extension name="isdn-outbound">
  <condition field="destination_number" expression="^0(\d+)$">
   <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
   <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
   <action application="bridge" data="sofia/gateway/isdn/$1"/>
  </condition>
 </extension>
</include>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/700853c9/attachment.html 

From randomdev4 at gmail.com  Tue Mar 31 19:59:07 2015
From: randomdev4 at gmail.com (Tim Smith)
Date: Tue, 31 Mar 2015 16:59:07 +0100
Subject: [Freeswitch-users] User registering on wrong sofia profile ??
Message-ID: <CA+HuS5E9ZVHeXZ=vtsagJQfjbn5HzWxe82BT7H0EVmXP=drJZw@mail.gmail.com>

Hi,

Any ideas why user 2110 is coming up on external rather than internal ?

2005|default|v.example.com|default|sofia/internal/sip:2005 at 10.10.10.1:5060
2110|default|v.example.com|default|sofia/external/sip:2110 at 10.10.10.2:61039


The directory XML for user 2110 is no different to that for 2005, so
I've no idea why Freeswitch is behaving like this ?

I've taken a look around the sofia xml files but can't find anywhere
in there that might be distinguishing between user extension numbers.

My vars.xml is correctly set so that v.example.com (not my real
domain) is the registration domain.

Any ideas ???


From tfred31 at yahoo.com  Tue Mar 31 19:59:50 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 08:59:50 -0700
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <CAFiqYun7C1Jbb5CkEYub9scZyuJ7-jCngXXjGcybS37021AhHQ@mail.gmail.com>
Message-ID: <1427817590.71213.YahooMailBasic@web160205.mail.bf1.yahoo.com>

OK

Within the file:   conf/directory/default.xml    I specifically defined the   <domain name="192.168.nn.nn">    to be the 'inside' IP which the SIP phones are to use.

And in the  conf/sip_profiles/internal.xml    I defined
       sip-ip  <==  'inside' IP
       rtp-ip  <==  'inside' IP

      ext-sip-ip   <==  'outside' IP
      ext-rtp-ip   <==  'outside' IP

Just in case I also put the same definitions into   conf/sip_profiles/external.xml

After a FS ReStart I still only see FS_CLI (sofia status) referring to the 'outside'  IP

And on doing a ReStart of the SIP phone, I still get the same error message as before as seen on FS_CLI
      Can't find user [105 at 192.168.nn.nn]  from <the phone's IP address>
      You must define a domain called  <FS 'inside' IP address 192.168.nn.nn>  in your directory and add a user with the ID '105' attribute
      and you must configure your device to use the proper domain in its authentication credentials

Obviously I am still missing something.

Suggestions?

Thanks
TF



From andrew at cassidywebservices.co.uk  Tue Mar 31 20:04:18 2015
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 31 Mar 2015 17:04:18 +0100
Subject: [Freeswitch-users] Caller ID prefix missing
In-Reply-To: <CAP6oyxEhz80ETuJk+Lqa9b5F3iwy26A4TQzfAAw0qaXGQFTzMw@mail.gmail.com>
References: <CAP6oyxEhz80ETuJk+Lqa9b5F3iwy26A4TQzfAAw0qaXGQFTzMw@mail.gmail.com>
Message-ID: <CAMbW9o+DZPhpHWWYw1c+j=KAcvBJyuR2yo_nK-OSCUy4NftuRQ@mail.gmail.com>

change $1 to $0 or 0$1

On 31 March 2015 at 16:29, Paul Atreides <paul.atreides83 at googlemail.com>
wrote:

> Hi,
>
> when I place a call over the gateway to the PTSN I see the dialed number
> without
> the leading 0 prefix in the history of my Grandstream GXP2140. Is there a
> way
> to add the 0 back again?
>
> Thank you
>
> <include>
>  <extension name="isdn-outbound">
>   <condition field="destination_number" expression="^0(\d+)$">
>    <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>    <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>    <action application="bridge" data="sofia/gateway/isdn/$1"/>
>   </condition>
>  </extension>
> </include>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/0f4d04bd/attachment.html 

From steveayre at gmail.com  Tue Mar 31 22:15:40 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 Mar 2015 19:15:40 +0100
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <1427814709.82232.YahooMailBasic@web160203.mail.bf1.yahoo.com>
References: <CAFiqYun7C1Jbb5CkEYub9scZyuJ7-jCngXXjGcybS37021AhHQ@mail.gmail.com>
	<1427814709.82232.YahooMailBasic@web160203.mail.bf1.yahoo.com>
Message-ID: <CAFiqYu=eAo=EjQL51e_JsZxx+zk3CD6ywqckzovSWqt827SO2w@mail.gmail.com>

Re internal.xml and external.xml - yes

There's nothing special about the internal/external profiles by the way.
They're just an example configuration. You can have as many or as few as
you like.

On 31 March 2015 at 16:11, T Fred Farmington <tfred31 at yahoo.com> wrote:

> Thank you for your reply.
>
> OK, but pardon my ignorance, but how do I do that?
>
> I am guessing here, but does that mean that for the 'inside' IP
> (192.168.nn.nn) I use the    conf\sip_profiles\internal.xml  file and
> define   sip_ip and rtp_ip  specifically -- not using  the
> $$(internal_etc...) references?
> If so, OK
>
> But how do I define the 'outside' (107.1.nnn.nn) sip?
> Do I do it in the    conf\sip_profiles\external.xml    file?
> And, if so, what do I put there?
>
> Thanks,
> TF
>
>
>
>
> --------------------------------------------
> On Tue, 3/31/15, Steven Ayre <steveayre at gmail.com> wrote:
>
>  Subject: Re: [Freeswitch-users] Question re: local_ip_v4
>  To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>  Date: Tuesday, March 31, 2015, 7:21 AM
>
>  You want
>  to configure 2 sofia profiles, one listening on each
>  IP.
>  It might be simpler to
>  define the IPs within the profiles rather than in
>  vars.conf.xml (where you're just setting variables that
>  are referenced elsewhere).
>
>  On 30 March 2015 at 21:57,
>  T Fred Farmington <tfred31 at yahoo.com>
>  wrote:
>  We have 2
>  network cards on our FS server.
>
>  *  One IP goes to the 'outside' world  (IP =
>  107.1.nn.nn)
>
>  *  And the other IP goes to the 'inside' world
>  (IP = 192.168.nn.nn)
>
>
>
>  In spite of making suggested changes to   vars.xml
>   when FS launches it sets itself up to 'listen' to
>  the 'outside' world (IP = 107 etc.)
>
>  But the result of that is that none of the in-house SIP
>  phones can register to it.
>
>
>
>  The changes I made (following suggestions found on the
>  web)  to   vars.xml   are as follows:
>
>     New entry:
>
>          <X-PRE-PROCESS cmd="set"
>  data="local_ip_v4=192.168.nn.nn"/>   <!--
>  where nn values are 'real' values -->
>
>     This new line was entered immediately before
>
>         <X-PRE-PROCESS cmd="set"
>  data="domain=$${local_ip_v4}"/>
>
>
>
>  Additionally another web reference suggested that the
>  following 2 lines be added near the bottom of
>   vars.xml
>
>        <X-PRE-PROCESS cmd="set"
>  data="internal_sip_ip=192.168.nn.nn"/>
>
>        <X-PRE-PROCESS cmd="set"
>  data="internal_rtp_ip=192.168.nn.nn"/>
>
>
>
>  PROBLEM:   None of that seems to work.
>
>                      Even after not only ReStarting
>  the FS Service, but also after ReStarting the FS Server
>  itself and then Starting the FS Service
>
>
>
>  When through FS_CLI  I run:    sofia  status
>   and/or   sofia status profile internal      both of
>  them show the 107 IP instead of the 192 IP
>
>  Those changes have not caused FS to 'see' and
>  'listen to' the 192  port.
>
>
>
>  What am I doing wrong?
>
>
>
>  Your advice/suggestions would be greatly appreciated.
>
>
>
>  Thanks
>
>  TF
>
>
>
>
>
>
>
>
>
>  _________________________________________________________________________
>
>  Professional FreeSWITCH Consulting Services:
>
>  consulting at freeswitch.org
>
>  http://www.freeswitchsolutions.com
>
>
>
>  Official FreeSWITCH Sites
>
>  http://www.freeswitch.org
>
>  http://confluence.freeswitch.org
>
>  http://www.cluecon.com
>
>
>
>  FreeSWITCH-users mailing list
>
>  FreeSWITCH-users at lists.freeswitch.org
>
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>  http://www.freeswitch.org
>
>
>
>  -----Inline Attachment Follows-----
>
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
>
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
>
>  FreeSWITCH-users mailing
>  list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/f14b6570/attachment-0001.html 

From steveayre at gmail.com  Tue Mar 31 22:17:41 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 Mar 2015 19:17:41 +0100
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <1427817590.71213.YahooMailBasic@web160205.mail.bf1.yahoo.com>
References: <CAFiqYun7C1Jbb5CkEYub9scZyuJ7-jCngXXjGcybS37021AhHQ@mail.gmail.com>
	<1427817590.71213.YahooMailBasic@web160205.mail.bf1.yahoo.com>
Message-ID: <CAFiqYu=gggGCm8+L3pZanF-3FdseYXX4v4tuasrCKOMeGMuLiw@mail.gmail.com>

You want to have ext-sip-ip and ext-rtp-ip the same as sip-ip and rtp-ip.
Those settings are to handle some NAT situations where you're listening on
your LAN IP but outside are visible on your nated WAN IP.

>From what you descibed your box is assigned your external address directly
in which case you need 2 profiles

On 31 March 2015 at 16:59, T Fred Farmington <tfred31 at yahoo.com> wrote:

> OK
>
> Within the file:   conf/directory/default.xml    I specifically defined
> the   <domain name="192.168.nn.nn">    to be the 'inside' IP which the SIP
> phones are to use.
>
> And in the  conf/sip_profiles/internal.xml    I defined
>        sip-ip  <==  'inside' IP
>        rtp-ip  <==  'inside' IP
>
>       ext-sip-ip   <==  'outside' IP
>       ext-rtp-ip   <==  'outside' IP
>
> Just in case I also put the same definitions into
>  conf/sip_profiles/external.xml
>
> After a FS ReStart I still only see FS_CLI (sofia status) referring to the
> 'outside'  IP
>
> And on doing a ReStart of the SIP phone, I still get the same error
> message as before as seen on FS_CLI
>       Can't find user [105 at 192.168.nn.nn]  from <the phone's IP address>
>       You must define a domain called  <FS 'inside' IP address
> 192.168.nn.nn>  in your directory and add a user with the ID '105' attribute
>       and you must configure your device to use the proper domain in its
> authentication credentials
>
> Obviously I am still missing something.
>
> Suggestions?
>
> Thanks
> TF
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/9c35907e/attachment.html 

From tfred31 at yahoo.com  Tue Mar 31 22:34:19 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 11:34:19 -0700
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <CAFiqYu=gggGCm8+L3pZanF-3FdseYXX4v4tuasrCKOMeGMuLiw@mail.gmail.com>
Message-ID: <1427826859.83834.YahooMailBasic@web160204.mail.bf1.yahoo.com>

Again, thank you for your reply.

OK, I'll make the change to   ext-sip-ip  and ext-rtp-ip   and make them the same IP address as   sip-ip  and rtp-ip  respectively

I'll will readily set up 2 profiles once I understand how to do so.

There is a simple description named:   SETTING THE SIP PROFILES TO USE DIFFERENT ETHERNET PORTS  on the previous FreeSWITCH wiki ( http://wiki.freeswitch.org/wiki/Multi_home_tutorial#INTERNAL_LAN )

And I did what they said to do there (except for making the mistake of not following their  ext-sip-ip &  ext-rtp-ip settings).   
But that did not seem to do what I need to do.
I'll try it again with the   ext-    settings corrected and see how it goes.

What about the phone registration problem where it cannot find the user [105 at 192.168.nn.nn] ?
I thought that I had everything in place for that to work, but I cannot register any SIP phones because of that problem.

Thanks,
TF





--------------------------------------------
On Tue, 3/31/15, Steven Ayre <steveayre at gmail.com> wrote:

 Subject: Re: [Freeswitch-users] Question re: local_ip_v4
 To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
 Date: Tuesday, March 31, 2015, 12:17 PM
 
You want to have ext-sip-ip and ext-rtp-ip the same as sip-ip and rtp-ip. 
Those settings are to handle some NAT situations where you're listening on your LAN IP but outside are visible on your nated WAN IP.

>From what you descibed your box is assigned your external address directly in which case you need 2 profiles 




From ing.antonyam at gmail.com  Tue Mar 31 23:21:59 2015
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Tue, 31 Mar 2015 13:21:59 -0600
Subject: [Freeswitch-users] Error Mod_lua.
Message-ID: <CABFbAiXfhHg0nPfwgBtpqdKphERVdMrk7n9R4=HrdMWByGSOmQ@mail.gmail.com>

Hi,

Install fusionpbx to manage freeswitch, perform the configuration of lua
module enable these lines in lua.conf.xml

<param name = "xml-handler-script" value = "app.lua xml_handler" />
  <param name = "xml-handler-bindings" value = "dialplan | directory |
configuration" />

I am using the script for the configuration that comes in the pbx in the
following path:

/ var / www / html / fusionpbx / resources / install / scripts

copy them to the following path freeswitch:

/ usr / local / freeswitch / scripts /

but when you start freeswitch shows me the following errors have some idea:

2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
/usr/local/freeswitch/scripts/resources/config.lua: No such file or
directory
stack traceback:
[C]: in function 'dofile'
/usr/local/freeswitch/scripts/app.lua:30: in main chunk
2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
error!
2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
/usr/local/freeswitch/scripts/resources/config.lua: No such file or
directory
stack traceback:
[C]: in function 'dofile'
/usr/local/freeswitch/scripts/app.lua:30: in main chunk
2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
error!
2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
/usr/local/freeswitch/scripts/resources/config.lua: No such file or
directory
stack traceback:
[C]: in function 'dofile'
/usr/local/freeswitch/scripts/app.lua:30: in main chunk
2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
error!

Regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/1a020ab9/attachment.html 

From steveayre at gmail.com  Tue Mar 31 23:32:54 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 Mar 2015 20:32:54 +0100
Subject: [Freeswitch-users] User registering on wrong sofia profile ??
In-Reply-To: <CA+HuS5E9ZVHeXZ=vtsagJQfjbn5HzWxe82BT7H0EVmXP=drJZw@mail.gmail.com>
References: <CA+HuS5E9ZVHeXZ=vtsagJQfjbn5HzWxe82BT7H0EVmXP=drJZw@mail.gmail.com>
Message-ID: <CAFiqYukDDH5H-RuMvFpQAockGizyX6j6FTj=uCDNyegAEZSANA@mail.gmail.com>

It's showing the SIP URI that lets you call them. That'll vary depending on
the profile registered to, as you'll want to reply along the same route.

It probably means they're not registering to the same port as user 2005 is
- internal is probably on 5060 and external 5080 if you're basing it off
the default config.

If both users are callable then it's likely not an issue.

Steve




On 31 March 2015 at 16:59, Tim Smith <randomdev4 at gmail.com> wrote:

> Hi,
>
> Any ideas why user 2110 is coming up on external rather than internal ?
>
> 2005|default|v.example.com|default|sofia/internal/sip:2005 at 10.10.10.1:5060
> 2110|default|v.example.com|default|sofia/external/
> sip:2110 at 10.10.10.2:61039
>
>
> The directory XML for user 2110 is no different to that for 2005, so
> I've no idea why Freeswitch is behaving like this ?
>
> I've taken a look around the sofia xml files but can't find anywhere
> in there that might be distinguishing between user extension numbers.
>
> My vars.xml is correctly set so that v.example.com (not my real
> domain) is the registration domain.
>
> Any ideas ???
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/9152db36/attachment.html 

From steveayre at gmail.com  Tue Mar 31 23:35:12 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 31 Mar 2015 20:35:12 +0100
Subject: [Freeswitch-users] Error Mod_lua.
In-Reply-To: <CABFbAiXfhHg0nPfwgBtpqdKphERVdMrk7n9R4=HrdMWByGSOmQ@mail.gmail.com>
References: <CABFbAiXfhHg0nPfwgBtpqdKphERVdMrk7n9R4=HrdMWByGSOmQ@mail.gmail.com>
Message-ID: <CAFiqYunDX4ZoSB+SNV_ChnsZ-xaLoF9yK--gbfBNZUzo+bqQsQ@mail.gmail.com>

It's probably a file permission problem. Linux returns the same 'no such
file' error for both missing files and permission errors.

Does the 'freeswitch' user have +r on the file and +rx on every directory
component in the path - either through user (u), group (g) or other (o)?

On 31 March 2015 at 20:21, Antony Aguirre Morales <ing.antonyam at gmail.com>
wrote:

> Hi,
>
> Install fusionpbx to manage freeswitch, perform the configuration of lua
> module enable these lines in lua.conf.xml
>
> <param name = "xml-handler-script" value = "app.lua xml_handler" />
>   <param name = "xml-handler-bindings" value = "dialplan | directory |
> configuration" />
>
> I am using the script for the configuration that comes in the pbx in the
> following path:
>
> / var / www / html / fusionpbx / resources / install / scripts
>
> copy them to the following path freeswitch:
>
> / usr / local / freeswitch / scripts /
>
> but when you start freeswitch shows me the following errors have some idea:
>
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
> /usr/local/freeswitch/scripts/resources/config.lua: No such file or
> directory
> stack traceback:
> [C]: in function 'dofile'
> /usr/local/freeswitch/scripts/app.lua:30: in main chunk
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
> error!
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
> /usr/local/freeswitch/scripts/resources/config.lua: No such file or
> directory
> stack traceback:
> [C]: in function 'dofile'
> /usr/local/freeswitch/scripts/app.lua:30: in main chunk
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
> error!
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open
> /usr/local/freeswitch/scripts/resources/config.lua: No such file or
> directory
> stack traceback:
> [C]: in function 'dofile'
> /usr/local/freeswitch/scripts/app.lua:30: in main chunk
> 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute
> error!
>
> Regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/7e9dd352/attachment-0001.html 

From paul.atreides83 at googlemail.com  Tue Mar 31 23:37:43 2015
From: paul.atreides83 at googlemail.com (Paul Atreides)
Date: Tue, 31 Mar 2015 21:37:43 +0200
Subject: [Freeswitch-users] Caller ID prefix missing
In-Reply-To: <CAMbW9o+DZPhpHWWYw1c+j=KAcvBJyuR2yo_nK-OSCUy4NftuRQ@mail.gmail.com>
References: <CAP6oyxEhz80ETuJk+Lqa9b5F3iwy26A4TQzfAAw0qaXGQFTzMw@mail.gmail.com>
	<CAMbW9o+DZPhpHWWYw1c+j=KAcvBJyuR2yo_nK-OSCUy4NftuRQ@mail.gmail.com>
Message-ID: <CAP6oyxFFfWGP+2YV77m880OO84yNd6mVwgxg6o7ozxJ+9Yk94g@mail.gmail.com>

Hi Andrew,

I think my description was not so good. My gateway accepts 012456789 but I
want the user to dial 001234567.
Can I remove the leading 0 and still tell the sip client that the user
dialed with the extra 0?


On Tue, Mar 31, 2015 at 6:04 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> change $1 to $0 or 0$1
>
> On 31 March 2015 at 16:29, Paul Atreides <paul.atreides83 at googlemail.com>
> wrote:
>
>> Hi,
>>
>> when I place a call over the gateway to the PTSN I see the dialed number
>> without
>> the leading 0 prefix in the history of my Grandstream GXP2140. Is there a
>> way
>> to add the 0 back again?
>>
>> Thank you
>>
>> <include>
>>  <extension name="isdn-outbound">
>>   <condition field="destination_number" expression="^0(\d+)$">
>>    <action application="set"
>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>    <action application="set"
>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>    <action application="bridge" data="sofia/gateway/isdn/$1"/>
>>   </condition>
>>  </extension>
>> </include>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150331/2e626537/attachment.html 

From tfred31 at yahoo.com  Tue Mar 31 23:41:16 2015
From: tfred31 at yahoo.com (T Fred Farmington)
Date: Tue, 31 Mar 2015 12:41:16 -0700
Subject: [Freeswitch-users] Question re: local_ip_v4
In-Reply-To: <CAFiqYu=gggGCm8+L3pZanF-3FdseYXX4v4tuasrCKOMeGMuLiw@mail.gmail.com>
Message-ID: <1427830876.97153.YahooMailBasic@web160203.mail.bf1.yahoo.com>

Thanks to everyone for their assistance in trying to resolve this problem.

It seemed like I did worked.  
Or, more likely, I did not do enough towards configuring EVERYTHING so as to resolve it.

Consequently I have decided to abandon this hardware approach.
I am certain that one of you FS gurus could have gotten it to work as intended, but it was more than my novice skills could tackle.

I had managed to get a working FS server on a virtual Windows workstation and I was trying to get it running on a non-Virtual box.

With the abandonment of the dual network card approach in mind, I have now copied EVERYTHING from the virtual workstation's FreeSWITCH directory over to the new box and configured its network card like the virtual workstation.

Now I have a new problem, but since its topic is different I will address it in a new thread

Thanks again for all your support and advice.
TF