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
>
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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
>
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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
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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
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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
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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
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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
>
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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
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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
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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
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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
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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>

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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

 

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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
>
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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
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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
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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
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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
>
>
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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
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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
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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.
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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.

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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

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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.
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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
>
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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

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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
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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
>
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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
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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
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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
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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
> 

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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

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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
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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
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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
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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
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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
>
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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.

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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

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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

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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.

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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.

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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

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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

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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
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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
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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
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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
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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



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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
>
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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
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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
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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
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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
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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
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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
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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
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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

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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
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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
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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
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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
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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
>
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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
>
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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
>
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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?
> 

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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.
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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
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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
>
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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
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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
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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
>
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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
>
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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 $${sounds_dir}
>
> Thanks
>
> Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bote_radio at botecomm.com  Wed Mar  4 19:35:32 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 4 Mar 2015 11:35:32 -0500
Subject: [Freeswitch-users] $${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

 

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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?
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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
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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?
>
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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
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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
>
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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."
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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
>
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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
 
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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.
>> 
>>  
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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
>
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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
>
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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."
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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

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

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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! 		 	   		  
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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
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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."
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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

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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

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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
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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
>
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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
>
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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
>
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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
>
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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,
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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?
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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
 		 	   		  
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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
>
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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?
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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

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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
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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
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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
>
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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
>
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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
>
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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

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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

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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
>
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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
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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
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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!
> > >> >
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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

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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.
> 

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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 		 	   		  
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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.
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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
>
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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/>
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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
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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.
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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
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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
>
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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,
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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
>
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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
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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.
>>> 
>> 

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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?

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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. 
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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,
> 

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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
>
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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
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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

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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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
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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
>
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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
>
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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
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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
>
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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

 

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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
>
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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
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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
>
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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

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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
>
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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

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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
>
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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
>
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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
>
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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
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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

 

 

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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
>
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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
>
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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.
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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
>>
>
>
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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.

 

 

 

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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


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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
>
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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
>
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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
>
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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
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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
>
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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
>
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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
>
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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
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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
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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
>
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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.
>> 

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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

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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?
>> 
> 

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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
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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
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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.
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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?

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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
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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 ? 

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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 ?

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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
>
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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
>
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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
>
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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 ? 
> 
>  
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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

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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

 

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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.

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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
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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
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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
>
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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
>
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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
>
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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


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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?
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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
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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 --------------
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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
>
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.=======================================================.
|            _____ ____     ____ _     ___              |
|           |  ___/ ___|   / ___| |   |_ _|             |
|           | |_  \___ \  | |   | |    | |              |
|           |  _|  ___) | | |___| |___ | |              |
|           |_|   |____/   \____|_____|___|             |
|                                                       |
.=======================================================.
| 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
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>> >
>> >     FreeSWITCH-users mailing list
>> >     FreeSWITCH-users at lists.freeswitch.org <mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>> >
>> >
>> >
>> >
>> > --
>> >
>> > */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
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>> > http://www.cluecon.com
>> >
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>> > 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
>>
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>> 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
>
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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


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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
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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
>>
>
>
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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
>>
>
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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
>
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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

 

 

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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
>
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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
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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?
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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
>>
>>
>>
>>
>