From david.villasmil.work at gmail.com  Sun Jan  1 20:04:26 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 01 Jan 2017 17:04:26 +0000
Subject: [Freeswitch-users] Farsi language support in FreeSwitch
In-Reply-To: <CAKbxfG87HBNftfkPBX0dPRDELVcbN0Y3RCPN4ndjp4g-MSAO7w@mail.gmail.com>
References: <CALR0W69FPiBfANybZLHURhSN4jDn=BiR4cx2_yh1g4gmaNUd-A@mail.gmail.com>
	<CALR0W6_JjUeL_K7eMPE1FjEJ9mqTGqgiN0ABGrFZWz1BNVuD=g@mail.gmail.com>
	<CAKbxfG87HBNftfkPBX0dPRDELVcbN0Y3RCPN4ndjp4g-MSAO7w@mail.gmail.com>
Message-ID: <CAFGRPVrqs3xdrakGCFvVKtX4U9=C2hzsYCY1ucwU81cRbg_Sjg@mail.gmail.com>

You'd need to record your own language. Look at the existing recordings and
make your own in exactly the same way under a different directory. After
that just configure freeswitch to use the files in the directory you
created.
On Fri, Dec 30, 2016 at 3:53 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Hi,
>
> We don't have a module for that lang but its possible to create one.
>
> Call lengths can be determined from the CDRS.
>
>
> On Fri, Dec 30, 2016 at 2:06 PM, Chandramouli P <mouli123 at gmail.com>
> wrote:
>
> Hello List,
>
> Any update would be appreciated.
>
> Thank you,
> Chandramouli.
>
>
> On Fri, Dec 30, 2016 at 6:55 PM, Chandramouli P <mouli123 at gmail.com>
> wrote:
>
> Hello,
>
> I am planning to use FreeSwitch to implement a small project. I have the
> below queries:
>
> 1) Does FreeSwitch supports "Farsi" language?
> 2) Is it possible to say current date and time in "Farsi" language"
> 3) Is it possible to calculate the active call length in FreeSwitch? I
> mean that let us assume ten calls are active on a single channel
> (concurrent calls) and they are listening my welcome greeting or file. Now,
> I want to calculate the active call length of the users. Is it possible or
> not?
>
> Thanks in advance,
> Chandramouli.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From kamil.nigmatullin at gmail.com  Sun Jan  1 21:45:04 2017
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Mon, 2 Jan 2017 00:45:04 +0600
Subject: [Freeswitch-users] Abandoned members ESL problem
In-Reply-To: <CABMX_w8TYjdh74KsCOy9OCcUjOcaBSk9nL=sgEhr2UJ5iEC+bw@mail.gmail.com>
References: <CA+XtiefaQZHoaf-4UFaUdtg_Ti70svRboqPJvdHuRCLbMN0UXA@mail.gmail.com>
	<CABMX_w89+gwuO5SaxbTO+xwJ5K=7njfaXwwW68t4ocA1_O+xfA@mail.gmail.com>
	<CA+XtiecP+ZC8DVhqQHGnEtK_e8RX1Xjrb70A0o=LJsa-hvA8mg@mail.gmail.com>
	<CABMX_w8TYjdh74KsCOy9OCcUjOcaBSk9nL=sgEhr2UJ5iEC+bw@mail.gmail.com>
Message-ID: <CA+XtieeQ7PB7NzE4K4zj3HoWKtDEzbto5dJkEDBp3FjvopCGSQ@mail.gmail.com>

Thanks. I have posted to JIRA.
https://freeswitch.org/jira/browse/ESL-124


2016-12-30 1:16 GMT+06:00 ?talo Rossi <italo at freeswitch.org>:

> File a JIRA with steps to reproduce please
>
> On Thu, Dec 29, 2016 at 4:09 PM, Kamil Nigmatullin <
> kamil.nigmatullin at gmail.com> wrote:
>
>> freeswitch at freeswitch> version
>> FreeSWITCH Version 1.6.13+git~20161214T213702Z~d422498d0f~64bit (git
>> d422498 2016-12-14 21:37:02Z 64bit)
>>
>>
>> 2016-12-29 21:44 GMT+06:00 ?talo Rossi <italo at freeswitch.org>:
>>
>>> Which version?
>>>
>>> On Mon, Dec 26, 2016 at 2:49 AM, Kamil Nigmatullin <
>>> kamil.nigmatullin at gmail.com> wrote:
>>>
>>>> I am creating some kind of monitoring for callcenter. Found one thing.
>>>> Whan a Caller (Memeber) joined queue 1, then member-queue-start event
>>>> appears. But if caller left the queue, and joined in a few minutes later (discard_abandoned_after
>>>> = 3600) the same queue, then member-queue-start event is not shown.
>>>> Maybe it is right but how can I find out  if memeber rejoined the queue?
>>>>
>>>> --
>>>> Kamil Nigmatullin
>>>> Tel: 77272323748
>>>> mob: 7 (707) 2517003 <(707)%20251-7003>
>>>> Skype: kamil.nigmatullin
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>> italo at freeswitch.org
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Kamil Nigmatullin
>> Tel: 77272323748
>> mob: 7 (707) 2517003 <(707)%20251-7003>
>> Skype: kamil.nigmatullin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From therebel22 at gmail.com  Mon Jan  2 00:21:28 2017
From: therebel22 at gmail.com (Marc S)
Date: Sun, 1 Jan 2017 22:21:28 +0100
Subject: [Freeswitch-users] Count calls on specific gateway on failover
	bridge scenario
Message-ID: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>

Hello,

i'm trying to get current calls count on a specific gateway in failover
bridge scenario :

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bridge#mod_dptools:bridge-ImplementingFailover

I have tried :

show calls
show channels

but gateway name doesn't appears on list

sofia status
sofia status gateway GATEWAY_NAME

but no help, no counter of current calls.

Any ideas ?

Thanks,
Marc
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From mouli123 at gmail.com  Mon Jan  2 09:41:44 2017
From: mouli123 at gmail.com (Chandramouli P)
Date: Mon, 2 Jan 2017 12:11:44 +0530
Subject: [Freeswitch-users] Farsi language support in FreeSwitch
In-Reply-To: <CAKbxfG87HBNftfkPBX0dPRDELVcbN0Y3RCPN4ndjp4g-MSAO7w@mail.gmail.com>
References: <CALR0W69FPiBfANybZLHURhSN4jDn=BiR4cx2_yh1g4gmaNUd-A@mail.gmail.com>
	<CALR0W6_JjUeL_K7eMPE1FjEJ9mqTGqgiN0ABGrFZWz1BNVuD=g@mail.gmail.com>
	<CAKbxfG87HBNftfkPBX0dPRDELVcbN0Y3RCPN4ndjp4g-MSAO7w@mail.gmail.com>
Message-ID: <CALR0W69xZY00mFzM0RQOZUkoW61S5awMn+k1ngFYMommYUZ3tQ@mail.gmail.com>

Hello Anthony,

Thanks for your reply. I am aware that we can determine the call length
from the CDRs. But, what I doubt Is it possible to determine the active
call length (during the call and not after completion of the call)?

Thank you,
Chandramouli.


On Sat, Dec 31, 2016 at 2:22 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Hi,
>
> We don't have a module for that lang but its possible to create one.
>
> Call lengths can be determined from the CDRS.
>
>
> On Fri, Dec 30, 2016 at 2:06 PM, Chandramouli P <mouli123 at gmail.com>
> wrote:
>
>> Hello List,
>>
>> Any update would be appreciated.
>>
>> Thank you,
>> Chandramouli.
>>
>>
>> On Fri, Dec 30, 2016 at 6:55 PM, Chandramouli P <mouli123 at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> I am planning to use FreeSwitch to implement a small project. I have the
>>> below queries:
>>>
>>> 1) Does FreeSwitch supports "Farsi" language?
>>> 2) Is it possible to say current date and time in "Farsi" language"
>>> 3) Is it possible to calculate the active call length in FreeSwitch? I
>>> mean that let us assume ten calls are active on a single channel
>>> (concurrent calls) and they are listening my welcome greeting or file. Now,
>>> I want to calculate the active call length of the users. Is it possible or
>>> not?
>>>
>>> Thanks in advance,
>>> Chandramouli.
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mouli123 at gmail.com  Mon Jan  2 09:45:48 2017
From: mouli123 at gmail.com (Chandramouli P)
Date: Mon, 2 Jan 2017 12:15:48 +0530
Subject: [Freeswitch-users] Possible to use Sofia SIP Stack for commercial
	use?
Message-ID: <CALR0W6_BMFS4864A5HevWVdEktgRBTXCsioN7Eai3+y3MpEz7w@mail.gmail.com>

Hello,

Is it possible to use Sofia sip stack for free of cost in commercial
application? I want to integrate Sofia sip stack with Android and develop
an application and release it commercially. Is there any licensing
restrictions?

Thank you,
Chandramouli.
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From david.villasmil.work at gmail.com  Mon Jan  2 10:34:29 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 02 Jan 2017 07:34:29 +0000
Subject: [Freeswitch-users] Farsi language support in FreeSwitch
In-Reply-To: <CALR0W69xZY00mFzM0RQOZUkoW61S5awMn+k1ngFYMommYUZ3tQ@mail.gmail.com>
References: <CALR0W69FPiBfANybZLHURhSN4jDn=BiR4cx2_yh1g4gmaNUd-A@mail.gmail.com>
	<CALR0W6_JjUeL_K7eMPE1FjEJ9mqTGqgiN0ABGrFZWz1BNVuD=g@mail.gmail.com>
	<CAKbxfG87HBNftfkPBX0dPRDELVcbN0Y3RCPN4ndjp4g-MSAO7w@mail.gmail.com>
	<CALR0W69xZY00mFzM0RQOZUkoW61S5awMn+k1ngFYMommYUZ3tQ@mail.gmail.com>
Message-ID: <CAFGRPVreMmuRHMmCsm=0WS-9iJkwf5sp5aH9mnfVwJv2Rf94_w@mail.gmail.com>

Quickest way, configure odbc in the core (
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/6586721).
You will get a table containing all dialogs (ongoing calls), you should be
able to get it using that.
Another way is to use ESL, and get events, you can get connects, hangups,
etc. Take a look at
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/1048924

Regards,

David.


On Mon, Jan 2, 2017 at 2:42 AM Chandramouli P <mouli123 at gmail.com> wrote:

Hello Anthony,

Thanks for your reply. I am aware that we can determine the call length
from the CDRs. But, what I doubt Is it possible to determine the active
call length (during the call and not after completion of the call)?

Thank you,
Chandramouli.


On Sat, Dec 31, 2016 at 2:22 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

Hi,

We don't have a module for that lang but its possible to create one.

Call lengths can be determined from the CDRS.


On Fri, Dec 30, 2016 at 2:06 PM, Chandramouli P <mouli123 at gmail.com> wrote:

Hello List,

Any update would be appreciated.

Thank you,
Chandramouli.


On Fri, Dec 30, 2016 at 6:55 PM, Chandramouli P <mouli123 at gmail.com> wrote:

Hello,

I am planning to use FreeSwitch to implement a small project. I have the
below queries:

1) Does FreeSwitch supports "Farsi" language?
2) Is it possible to say current date and time in "Farsi" language"
3) Is it possible to calculate the active call length in FreeSwitch? I mean
that let us assume ten calls are active on a single channel (concurrent
calls) and they are listening my welcome greeting or file. Now, I want to
calculate the active call length of the users. Is it possible or not?

Thanks in advance,
Chandramouli.






_________________________________________________________________________


Professional FreeSWITCH Consulting Services:


consulting at freeswitch.org


http://www.freeswitchsolutions.com





Official FreeSWITCH Sites


http://www.freeswitch.org


http://confluence.freeswitch.org


http://www.cluecon.com





FreeSWITCH-users mailing list


FreeSWITCH-users at lists.freeswitch.org


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


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


http://www.freeswitch.org




-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA





_________________________________________________________________________


Professional FreeSWITCH Consulting Services:


consulting at freeswitch.org


http://www.freeswitchsolutions.com





Official FreeSWITCH Sites


http://www.freeswitch.org


http://confluence.freeswitch.org


http://www.cluecon.com





FreeSWITCH-users mailing list


FreeSWITCH-users at lists.freeswitch.org


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


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


http://www.freeswitch.org




_________________________________________________________________________


Professional FreeSWITCH Consulting Services:


consulting at freeswitch.org


http://www.freeswitchsolutions.com





Official FreeSWITCH Sites


http://www.freeswitch.org


http://confluence.freeswitch.org


http://www.cluecon.com





FreeSWITCH-users mailing list


FreeSWITCH-users at lists.freeswitch.org


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


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


http://www.freeswitch.org
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From devang.nathwani31589 at gmail.com  Mon Jan  2 13:06:44 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Mon, 2 Jan 2017 15:36:44 +0530
Subject: [Freeswitch-users] Codec transcoding PCMA to PMCU
In-Reply-To: <CAEJMVkChxr1mdk7+9h=DpTrDTqapOHYqUkUSj_NKvpcpi1LzrA@mail.gmail.com>
References: <CAGqqW99s1fYvZN8dg_hngWPhBaDfdNz+d1eJ-A6W_GhKxyfdNQ@mail.gmail.com>
	<CAEJMVkChxr1mdk7+9h=DpTrDTqapOHYqUkUSj_NKvpcpi1LzrA@mail.gmail.com>
Message-ID: <CAGOm0PJvcKpbGZsy=f8Je3R3nQCK9XV-8OF6MGCAQOBnXPEG5g@mail.gmail.com>

Hello,

This is our dialplan,

<section name="dialplan" description="My Dialplan">
<context name="default">
<extension name="22290915300123">
<condition field="destination_number" expression="22290915300123">
<action application="set"
data="effective_destination_number=22290915300123"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="export" data="media_bug_answer_req=true"/>
<action application="export" data="RECORD_STEREO=true"/>
<action application="export" data="record_sample_rate=8000"/>
<action application="export" data="execute_on_answer=record_session
/home/shared/recordings/c9b95b66-efcb-45a9-89b7-b6e39f4a2c4b.wav"/>
<action application="set" data="absolute_codec_string=PCMU"/>
<action application="bridge" data="sofia/gateway/Twilio/+34915300123"/>
</condition>
</extension>
</context>
</section>

On Tue, Dec 27, 2016 at 8:51 PM, Brian West <brian at freeswitch.org> wrote:

> We would need the full dialplan that you're executing to accomplish this.
>
> On Tue, Dec 27, 2016 at 5:27 AM, Ketan Kothari <kkothari157 at gmail.com>
> wrote:
>
>> Hello,
>>
>> I have configured PCMA codec in my zoipper soft-phone(UAC) and tried to
>> call to 123456789 using our gateway twillo(UAS).
>> Gateway only accept PCMU codec. I have also configured absolute codec
>> =PCMU but still freeswitch sending PCMA to twillo(Provider gateway UAS). I
>> have got error Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED in
>> freeswitch.
>>
>> Once i have added PCMU codec in my zoiper(UAC) it's working.
>>
>>
>> [image: Inline image 1]
>>
>>
>> What am i missing here?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 (50% Discount using code FreeSwitch50)
> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50)
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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 ashwinrath at gmail.com  Mon Jan  2 14:20:01 2017
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Mon, 2 Jan 2017 16:50:01 +0530
Subject: [Freeswitch-users] Possible to use Sofia SIP Stack for
	commercial use?
In-Reply-To: <CALR0W6_BMFS4864A5HevWVdEktgRBTXCsioN7Eai3+y3MpEz7w@mail.gmail.com>
References: <CALR0W6_BMFS4864A5HevWVdEktgRBTXCsioN7Eai3+y3MpEz7w@mail.gmail.com>
Message-ID: <CAA_Nwu6weELm+Z6qkhfLdLXeVhnL4GtHqLYR7GL2v3zumrCKxA@mail.gmail.com>

Yes you can. Sofia is licensed under LGPL which allows you to use it for
proprietary apps. More here :  http://www.gnu.org/licenses/why-not-lgpl.html

On 2 January 2017 at 12:15, Chandramouli P <mouli123 at gmail.com> wrote:

> Hello,
>
> Is it possible to use Sofia sip stack for free of cost in commercial
> application? I want to integrate Sofia sip stack with Android and develop
> an application and release it commercially. Is there any licensing
> restrictions?
>
> Thank you,
> Chandramouli.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ashwin Kumar Rath
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From vma at vallimamod.org  Mon Jan  2 14:27:33 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 2 Jan 2017 12:27:33 +0100
Subject: [Freeswitch-users] Count calls on specific gateway on failover
 bridge scenario
In-Reply-To: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>
References: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>
Message-ID: <51494FC6-A226-47DE-96A9-BB574A74AD09@vallimamod.org>

Hi Marc,

AFAIK, there is no per-gateway call count directly available. You can only get the per-profile call count with "sofia status". But if you are only routing inbound calls to these gateways, you can easily retrieve the numbers with:

fs_cli -x 'show channels' | grep inbound | cut -d',' -f11,12

You will get something like:

bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX

So you just have to count the lines corresponding to each gateway to get your numbers.

Best Regards,
-- 
Vallimamod Abdullah
vma at vallimamod.org
.

> On 1 Jan 2017, at 22:21, Marc S <therebel22 at gmail.com> wrote:
> 
> Hello,
> 
> i'm trying to get current calls count on a specific gateway in failover bridge scenario :
> 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bridge#mod_dptools:bridge-ImplementingFailover
> 
> I have tried :
> 
> show calls
> show channels
> 
> but gateway name doesn't appears on list
> 
> sofia status
> sofia status gateway GATEWAY_NAME
> 
> but no help, no counter of current calls.
> 
> Any ideas ?
> 
> Thanks,
> Marc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From therebel22 at gmail.com  Mon Jan  2 16:00:43 2017
From: therebel22 at gmail.com (Marc S)
Date: Mon, 2 Jan 2017 14:00:43 +0100
Subject: [Freeswitch-users] Count calls on specific gateway on failover
 bridge scenario
In-Reply-To: <51494FC6-A226-47DE-96A9-BB574A74AD09@vallimamod.org>
References: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>
	<51494FC6-A226-47DE-96A9-BB574A74AD09@vallimamod.org>
Message-ID: <CAMs117WoVOfq0iYxgkFtsYcBZxhB1xscfeHxj2QanG7yu6XXqg@mail.gmail.com>

Hi Vallimamod,

Thanks for reply. Here the result :

signal_bridge,3c77e89c-d0ea-11e6-8e01-c7c4c1737996
signal_bridge,cf214562-d0ea-11e6-8eb6-c7c4c1737996

Problem is that "show channels" not giving me any sofia/gateway/.. string
because, i think, i use brige like this :

<action application="bridge" data=
"sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>


Marc

2017-01-02 12:27 GMT+01:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi Marc,
>
> AFAIK, there is no per-gateway call count directly available. You can only
> get the per-profile call count with "sofia status". But if you are only
> routing inbound calls to these gateways, you can easily retrieve the
> numbers with:
>
> fs_cli -x 'show channels' | grep inbound | cut -d',' -f11,12
>
> You will get something like:
>
> bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
>
> So you just have to count the lines corresponding to each gateway to get
> your numbers.
>
> Best Regards,
> --
> Vallimamod Abdullah
> vma at vallimamod.org
> .
>
> > On 1 Jan 2017, at 22:21, Marc S <therebel22 at gmail.com> wrote:
> >
> > Hello,
> >
> > i'm trying to get current calls count on a specific gateway in failover
> bridge scenario :
> >
> > https://freeswitch.org/confluence/display/FREESWITCH/
> mod_dptools%3A+bridge#mod_dptools:bridge-ImplementingFailover
> >
> > I have tried :
> >
> > show calls
> > show channels
> >
> > but gateway name doesn't appears on list
> >
> > sofia status
> > sofia status gateway GATEWAY_NAME
> >
> > but no help, no counter of current calls.
> >
> > Any ideas ?
> >
> > Thanks,
> > Marc
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From italo at freeswitch.org  Mon Jan  2 16:35:42 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 2 Jan 2017 10:35:42 -0300
Subject: [Freeswitch-users] Abandoned members ESL problem
In-Reply-To: <CA+XtieeQ7PB7NzE4K4zj3HoWKtDEzbto5dJkEDBp3FjvopCGSQ@mail.gmail.com>
References: <CA+XtiefaQZHoaf-4UFaUdtg_Ti70svRboqPJvdHuRCLbMN0UXA@mail.gmail.com>
	<CABMX_w89+gwuO5SaxbTO+xwJ5K=7njfaXwwW68t4ocA1_O+xfA@mail.gmail.com>
	<CA+XtiecP+ZC8DVhqQHGnEtK_e8RX1Xjrb70A0o=LJsa-hvA8mg@mail.gmail.com>
	<CABMX_w8TYjdh74KsCOy9OCcUjOcaBSk9nL=sgEhr2UJ5iEC+bw@mail.gmail.com>
	<CA+XtieeQ7PB7NzE4K4zj3HoWKtDEzbto5dJkEDBp3FjvopCGSQ@mail.gmail.com>
Message-ID: <CABMX_w9=BDE1b1eY437t3neEjPebgdLOBsi+TvfbvvGduq6MSw@mail.gmail.com>

You opened in the wrong project so I had to move the JIRA to the right one:
https://freeswitch.org/jira/browse/FS-9905

On Sun, Jan 1, 2017 at 3:45 PM, Kamil Nigmatullin <
kamil.nigmatullin at gmail.com> wrote:

> Thanks. I have posted to JIRA.
> https://freeswitch.org/jira/browse/ESL-124
>
>
> 2016-12-30 1:16 GMT+06:00 ?talo Rossi <italo at freeswitch.org>:
>
>> File a JIRA with steps to reproduce please
>>
>> On Thu, Dec 29, 2016 at 4:09 PM, Kamil Nigmatullin <
>> kamil.nigmatullin at gmail.com> wrote:
>>
>>> freeswitch at freeswitch> version
>>> FreeSWITCH Version 1.6.13+git~20161214T213702Z~d422498d0f~64bit (git
>>> d422498 2016-12-14 21:37:02Z 64bit)
>>>
>>>
>>> 2016-12-29 21:44 GMT+06:00 ?talo Rossi <italo at freeswitch.org>:
>>>
>>>> Which version?
>>>>
>>>> On Mon, Dec 26, 2016 at 2:49 AM, Kamil Nigmatullin <
>>>> kamil.nigmatullin at gmail.com> wrote:
>>>>
>>>>> I am creating some kind of monitoring for callcenter. Found one thing.
>>>>> Whan a Caller (Memeber) joined queue 1, then member-queue-start event
>>>>> appears. But if caller left the queue, and joined in a few minutes later (discard_abandoned_after
>>>>> = 3600) the same queue, then member-queue-start event is not shown.
>>>>> Maybe it is right but how can I find out  if memeber rejoined the queue?
>>>>>
>>>>> --
>>>>> Kamil Nigmatullin
>>>>> Tel: 77272323748
>>>>> mob: 7 (707) 2517003 <(707)%20251-7003>
>>>>> Skype: kamil.nigmatullin
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> ?talo Rossi
>>>> italo at freeswitch.org
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Kamil Nigmatullin
>>> Tel: 77272323748
>>> mob: 7 (707) 2517003 <(707)%20251-7003>
>>> Skype: kamil.nigmatullin
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>> italo at freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003 <(707)%20251-7003>
> Skype: kamil.nigmatullin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From vma at vallimamod.org  Mon Jan  2 18:34:15 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 2 Jan 2017 16:34:15 +0100
Subject: [Freeswitch-users] Count calls on specific gateway on failover
 bridge scenario
In-Reply-To: <CAMs117WoVOfq0iYxgkFtsYcBZxhB1xscfeHxj2QanG7yu6XXqg@mail.gmail.com>
References: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>
	<51494FC6-A226-47DE-96A9-BB574A74AD09@vallimamod.org>
	<CAMs117WoVOfq0iYxgkFtsYcBZxhB1xscfeHxj2QanG7yu6XXqg@mail.gmail.com>
Message-ID: <6F635258-320B-40CB-B1E3-B3B6BE94A23B@vallimamod.org>

Hi Marc,

Are you by any chance in bypass media mode?
If so that could explain why the application name is set to signal_bridge. I have tried with your dialplan in "normal" mode and I actually get the full dial url in the 12th field.

Unfortunately, it won't help either in your case as you won't know which gateway did actually take the call. But you can set continue_on_fail and add a second bridge app:

<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/gateway/primary/dialstring"/>
<action application="bridge" data="sofia/gateway/secondary/dialstring"/>

This way you will get the specific dial url in the channel list.

Best Regards,
-- 
Vallimamod Abdullah
vma at vallimamod.org
.

> On 2 Jan 2017, at 14:00, Marc S <therebel22 at gmail.com> wrote:
> 
> Hi Vallimamod,
> 
> Thanks for reply. Here the result :
> 
> signal_bridge,3c77e89c-d0ea-11e6-8e01-c7c4c1737996
> signal_bridge,cf214562-d0ea-11e6-8eb6-c7c4c1737996
> 
> Problem is that "show channels" not giving me any sofia/gateway/.. string because, i think, i use brige like this :
> 
> <action application="bridge" data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
> 
> 
> Marc
> 
> 2017-01-02 12:27 GMT+01:00 Vallimamod Abdullah <vma at vallimamod.org>:
> Hi Marc,
> 
> AFAIK, there is no per-gateway call count directly available. You can only get the per-profile call count with "sofia status". But if you are only routing inbound calls to these gateways, you can easily retrieve the numbers with:
> 
> fs_cli -x 'show channels' | grep inbound | cut -d',' -f11,12
> 
> You will get something like:
> 
> bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
> bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
> 
> So you just have to count the lines corresponding to each gateway to get your numbers.
> 
> Best Regards,
> --
> Vallimamod Abdullah
> vma at vallimamod.org
> .
> 
> > On 1 Jan 2017, at 22:21, Marc S <therebel22 at gmail.com> wrote:
> >
> > Hello,
> >
> > i'm trying to get current calls count on a specific gateway in failover bridge scenario :
> >
> > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bridge#mod_dptools:bridge-ImplementingFailover
> >
> > I have tried :
> >
> > show calls
> > show channels
> >
> > but gateway name doesn't appears on list
> >
> > sofia status
> > sofia status gateway GATEWAY_NAME
> >
> > but no help, no counter of current calls.
> >
> > Any ideas ?
> >
> > Thanks,
> > Marc
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From therebel22 at gmail.com  Mon Jan  2 18:57:56 2017
From: therebel22 at gmail.com (Marc S)
Date: Mon, 2 Jan 2017 16:57:56 +0100
Subject: [Freeswitch-users] Count calls on specific gateway on failover
 bridge scenario
In-Reply-To: <6F635258-320B-40CB-B1E3-B3B6BE94A23B@vallimamod.org>
References: <CAMs117Xkfx4cRs=s-ZRJ_D1LQYYmUPXo1AFZaQsL23gXzVa=MA@mail.gmail.com>
	<51494FC6-A226-47DE-96A9-BB574A74AD09@vallimamod.org>
	<CAMs117WoVOfq0iYxgkFtsYcBZxhB1xscfeHxj2QanG7yu6XXqg@mail.gmail.com>
	<6F635258-320B-40CB-B1E3-B3B6BE94A23B@vallimamod.org>
Message-ID: <CAMs117XxewFcTpdA8g56KJSmR7U=rYDEGZaNONnVHfA3Op1OSw@mail.gmail.com>

Yes ! It is in bypass mode !

I'm going to test continue_on_fail.

Thanks a lot !
Marc

2017-01-02 16:34 GMT+01:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi Marc,
>
> Are you by any chance in bypass media mode?
> If so that could explain why the application name is set to signal_bridge.
> I have tried with your dialplan in "normal" mode and I actually get the
> full dial url in the 12th field.
>
> Unfortunately, it won't help either in your case as you won't know which
> gateway did actually take the call. But you can set continue_on_fail and
> add a second bridge app:
>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="bridge" data="sofia/gateway/primary/dialstring"/>
> <action application="bridge" data="sofia/gateway/secondary/dialstring"/>
>
> This way you will get the specific dial url in the channel list.
>
> Best Regards,
> --
> Vallimamod Abdullah
> vma at vallimamod.org
> .
>
> > On 2 Jan 2017, at 14:00, Marc S <therebel22 at gmail.com> wrote:
> >
> > Hi Vallimamod,
> >
> > Thanks for reply. Here the result :
> >
> > signal_bridge,3c77e89c-d0ea-11e6-8e01-c7c4c1737996
> > signal_bridge,cf214562-d0ea-11e6-8eb6-c7c4c1737996
> >
> > Problem is that "show channels" not giving me any sofia/gateway/..
> string because, i think, i use brige like this :
> >
> > <action application="bridge" data="sofia/gateway/primary/
> dialstring|sofia/gateway/secondary/dialstring"/>
> >
> >
> > Marc
> >
> > 2017-01-02 12:27 GMT+01:00 Vallimamod Abdullah <vma at vallimamod.org>:
> > Hi Marc,
> >
> > AFAIK, there is no per-gateway call count directly available. You can
> only get the per-profile call count with "sofia status". But if you are
> only routing inbound calls to these gateways, you can easily retrieve the
> numbers with:
> >
> > fs_cli -x 'show channels' | grep inbound | cut -d',' -f11,12
> >
> > You will get something like:
> >
> > bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> > bridge,{ignore_display_update=true}sofia/gateway/gw1/+XXX
> > bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
> > bridge,{ignore_display_update=true}sofia/gateway/gw2/+XXX
> >
> > So you just have to count the lines corresponding to each gateway to get
> your numbers.
> >
> > Best Regards,
> > --
> > Vallimamod Abdullah
> > vma at vallimamod.org
> > .
> >
> > > On 1 Jan 2017, at 22:21, Marc S <therebel22 at gmail.com> wrote:
> > >
> > > Hello,
> > >
> > > i'm trying to get current calls count on a specific gateway in
> failover bridge scenario :
> > >
> > > https://freeswitch.org/confluence/display/FREESWITCH/
> mod_dptools%3A+bridge#mod_dptools:bridge-ImplementingFailover
> > >
> > > I have tried :
> > >
> > > show calls
> > > show channels
> > >
> > > but gateway name doesn't appears on list
> > >
> > > sofia status
> > > sofia status gateway GATEWAY_NAME
> > >
> > > but no help, no counter of current calls.
> > >
> > > Any ideas ?
> > >
> > > Thanks,
> > > Marc
> > > ____________________________________________________________
> _____________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vbvbrj at gmail.com  Tue Jan  3 12:40:07 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Tue, 3 Jan 2017 11:40:07 +0200
Subject: [Freeswitch-users] Combine non-auth and auth calls on same profile
 security consideration.
Message-ID: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>

Hello.

There where separate sip profiles for registered accounts and anonymous 
inbound calls: internal and external. All calls and registrations on 
internal a authenticated and routed to "default" context. For external 
profile inbound calls are not authenticated and are routed to "public" 
context.

In order for some one to call inbound (s)he have to use :5080 port, 
which is somehow inconvenient. So I decided to combine in one profile 
named "example" on port 5060.

<profile name="internal_77.89.245.34">
<settings>
<param name="context" value="public"/>
<param name="auth-calls" value="false"/>
<param name="auth-all-packets" value="false"/>
<param name="inbound-reg-force-matching-username" value="true"/>
<param name="force-register-domain" value="default"/>
<param name="force-subscription-domain" value="default"/>
<param name="force-register-db-domain" value="default"/>
</settings>
</profile>
<domain name="$${domain}">
<variables>
<variable name="inbound-reg-force-matching-username" value="true"/>
<variable name="user_context" value="default"/>
<groups>
<group name="default">
<users>
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
</users>
</group>
</groups>
</variables>
</domain>

Registration was working, calling to those registration was working, but 
some phones hit public context, some default context. I started to dig 
whats happening and found that D-Link phones always do authenticated 
calls when they a registered, while Stephen's phones does 
unauthenticated and go to public.

Then I found a mention that registrations allow only to find the phone 
to call, while calls does not necessary authenticate. So I used 
Anthony's solution in public context on the end:

<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>

Now when a registered phone hits public context and no other conditions 
are met, they are rejected and call authenticated hitting default context.

Then in CDR I see two lines: one with rejected and one with success. 
This is not well, so found a hint and put:
<anti-action application="set" data="process_cdr=false"/>

And now is working somewhat correct except that calls to public numbers 
which does not require authentication does not get vars for the 
registered extension.

My questions are:

1) Does this type of combination affect security?
2) How to impose all registered phones to make authenticated calls 
always? So they will not go first thru public context and then to default?

-- 
Mimiko desu.


From mike at jerris.com  Tue Jan  3 19:43:30 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Jan 2017 11:43:30 -0500
Subject: [Freeswitch-users] Codec transcoding PCMA to PMCU
In-Reply-To: <CAGOm0PJvcKpbGZsy=f8Je3R3nQCK9XV-8OF6MGCAQOBnXPEG5g@mail.gmail.com>
References: <CAGqqW99s1fYvZN8dg_hngWPhBaDfdNz+d1eJ-A6W_GhKxyfdNQ@mail.gmail.com>
	<CAEJMVkChxr1mdk7+9h=DpTrDTqapOHYqUkUSj_NKvpcpi1LzrA@mail.gmail.com>
	<CAGOm0PJvcKpbGZsy=f8Je3R3nQCK9XV-8OF6MGCAQOBnXPEG5g@mail.gmail.com>
Message-ID: <F45F4CE8-4DEC-4159-9A3F-08A2041E62DF@jerris.com>

your original diagram said you have inherit_codec=true.  You?r description says you want to change the codec not inherit it so you?d need to remove that setting.  Setting that is overriding your absolute_codec_string

> On Jan 2, 2017, at 5:06 AM, devang nathwani <devang.nathwani31589 at gmail.com> wrote:
> 
> Hello,
> 
> This is our dialplan,
> 
> <section name="dialplan" description="My Dialplan">
> <context name="default">
> <extension name="22290915300123">
> <condition field="destination_number" expression="22290915300123">
> <action application="set" data="effective_destination_number=22290915300123"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="export" data="media_bug_answer_req=true"/>
> <action application="export" data="RECORD_STEREO=true"/>
> <action application="export" data="record_sample_rate=8000"/>
> <action application="export" data="execute_on_answer=record_session /home/shared/recordings/c9b95b66-efcb-45a9-89b7-b6e39f4a2c4b.wav"/>
> <action application="set" data="absolute_codec_string=PCMU"/>
> <action application="bridge" data="sofia/gateway/Twilio/+34915300123"/>
> </condition>
> </extension>
> </context>
> </section>
> 
> On Tue, Dec 27, 2016 at 8:51 PM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
> We would need the full dialplan that you're executing to accomplish this.
> 
> On Tue, Dec 27, 2016 at 5:27 AM, Ketan Kothari <kkothari157 at gmail.com <mailto:kkothari157 at gmail.com>> wrote:
> Hello,
> 
> I have configured PCMA codec in my zoipper soft-phone(UAC) and tried to call to 123456789 using our gateway twillo(UAS).
> Gateway only accept PCMU codec. I have also configured absolute codec =PCMU but still freeswitch sending PCMA to twillo(Provider gateway UAS). I have got error Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED in freeswitch.
> 
> Once i have added PCMU codec in my zoiper(UAC) it's working.
> 
> 
> <Adiptel-fs-codec-issue.png>
> 
> 
> What am i missing here?
> 
> 
> 
> _________________________________________________________________________
> 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/> (50% Discount using code FreeSwitch50)
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> (50% Discount using code FreeSwitch50)
> https://www.gofundme.com/freeswitch_ubuntu <https://www.gofundme.com/freeswitch_ubuntu>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> Skype:briankwest
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <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 kkothari157 at gmail.com  Mon Jan  2 13:04:16 2017
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Mon, 2 Jan 2017 15:34:16 +0530
Subject: [Freeswitch-users] Codec transcoding PCMA to PMCU
In-Reply-To: <CAGqqW99s1fYvZN8dg_hngWPhBaDfdNz+d1eJ-A6W_GhKxyfdNQ@mail.gmail.com>
References: <CAGqqW99s1fYvZN8dg_hngWPhBaDfdNz+d1eJ-A6W_GhKxyfdNQ@mail.gmail.com>
Message-ID: <CAGqqW9-zN8pdmqQcy_zpyqc6QNd9b_=knke1X3DX2OyOhU-qsg@mail.gmail.com>

Hello,

This is my dialplan.

<section name="dialplan" description="My Dialplan">
<context name="default">
<extension name="22290915300123">
<condition field="destination_number" expression="22290915300123">
<action application="set"
data="effective_destination_number=22290915300123"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="export" data="media_bug_answer_req=true"/>
<action application="export" data="RECORD_STEREO=true"/>
<action application="export" data="record_sample_rate=8000"/>
<action application="export" data="execute_on_answer=record_session
/home/shared/recordings/c9b95b66-efcb-45a9-89b7-b6e39f4a2c4b.wav"/>
<action application="set" data="absolute_codec_string=PCMU"/>
<action application="bridge" data="sofia/gateway/Twilio/+34915300123"/>
</condition>
</extension>
</context>
</section>

On Tue, Dec 27, 2016 at 4:57 PM, Ketan Kothari <kkothari157 at gmail.com>
wrote:

> Hello,
>
> I have configured PCMA codec in my zoipper soft-phone(UAC) and tried to
> call to 123456789 using our gateway twillo(UAS).
> Gateway only accept PCMU codec. I have also configured absolute codec
> =PCMU but still freeswitch sending PCMA to twillo(Provider gateway UAS). I
> have got error Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED in
> freeswitch.
>
> Once i have added PCMU codec in my zoiper(UAC) it's working.
>
>
> [image: Inline image 1]
>
>
> What am i missing here?
>
>
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From 568691 at gmail.com  Tue Jan  3 23:33:04 2017
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Tue, 3 Jan 2017 22:33:04 +0200
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
Message-ID: <CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>

I'd recommend you to use HAPROXY as load-balancer for websockets as it can
be simplier configured as statefull proxy.

2016-12-24 12:05 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> Hello guys
> I want configure user frendly WebRTC server based on FreeSwitch and SipML5
> client.
>
> It can be easy done in FreeSwitch and NGINX is bounded to different
> IP/ports. But if you wants use one IP and 443 port then you will try
> configre NGINX to proxy all reuests line "/fs-socket/" to FreeSwitch port
> 7443.
>
> It is works fine but FreeSwitch cannot not see real client IP address like
> folowing. Captured on Amazon server.
>
> freeswitch at ip-172-31-29-87.us-west-2.compute.internal> sofia status
> profile internal reg
>
> Registrations:
> ============================================================
> =====================================
> Call-ID:     f1e8c7ca-8f50-4285-fd1a-148d2f1d1b88
> User:       23 at 46.218.201.23
> Contact:     "23" <sips:23 at df7jal23ls0d.invalid;
> rtcweb-breaker=no;transport=wss;fs_nat=yes;fs_path=sips%
> 3A23%40172.31.29.87%3A37244%3Brtcweb-breaker%3Dno%3Btransport%3Dwss>
> Agent:       IM-client/OMA1.0 sipML5-v1.2016.03.04
> Status:     Registered(TLS-NAT)(unknown) EXP(2016-12-24 09:42:43)
> EXPSECS(230)
> Ping-Status: Reachable
> Ping-Time: 0.00
> Host:       ip-172-31-29-87.us-west-2.compute.internal
> IP:         172.31.29.87
> Port:       37244
> Auth-User:   23
> Auth-Realm: 46.218.201.23
> MWI-Account: 23 at 46.218.201.23
>
> Total items returned: 1
> ============================================================
> =====================================
>
> Displayed real IP address of NGINX
>
> Also when received INVITE then variables like network_addr will cantain
> real IP of NGINX.
>
> Then you can try confgire nginx like ng
>
> proxy_set_header X-Real-IP $remote_addr;
> proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for;
>
>
> But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every
> SIP message not only connection establishing. NGINX not understand SIP
> messages and cannot do it.
>
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this
> case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port
> with http server?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Alexandru Covalschi
VoIP engineer and system administrator
tel: +37367398493
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From s.safarov at gmail.com  Tue Jan  3 23:47:03 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 03 Jan 2017 20:47:03 +0000
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
Message-ID: <CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>

Hello Alexander
But in haproxy FreeSwitch will see IP address of haproxy server for all
cases.
How to see real IP of WEbRTC device in FS when used proxy?

Sergey

??, 3 ???. 2017 ?. ? 23:34, Alexandru Covalschi <568691 at gmail.com>:

> I'd recommend you to use HAPROXY as load-balancer for websockets as it can
> be simplier configured as statefull proxy.
>
> 2016-12-24 12:05 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>
> Hello guys
> I want configure user frendly WebRTC server based on FreeSwitch and SipML5
> client.
>
> It can be easy done in FreeSwitch and NGINX is bounded to different
> IP/ports. But if you wants use one IP and 443 port then you will try
> configre NGINX to proxy all reuests line "/fs-socket/" to FreeSwitch port
> 7443.
>
> It is works fine but FreeSwitch cannot not see real client IP address like
> folowing. Captured on Amazon server.
>
> freeswitch at ip-172-31-29-87.us-west-2.compute.internal> sofia status
> profile internal reg
>
> Registrations:
>
> =================================================================================================
> Call-ID:     f1e8c7ca-8f50-4285-fd1a-148d2f1d1b88
> User:       23 at 46.218.201.23
> Contact:     "23" <sips:23 at df7jal23ls0d.invalid
> ;rtcweb-breaker=no;transport=wss;fs_nat=yes;fs_path=sips%3A23%40172.31.29.87%3A37244%3Brtcweb-breaker%3Dno%3Btransport%3Dwss>
> Agent:       IM-client/OMA1.0 sipML5-v1.2016.03.04
> Status:     Registered(TLS-NAT)(unknown) EXP(2016-12-24 09:42:43)
> EXPSECS(230)
> Ping-Status: Reachable
> Ping-Time: 0.00
> Host:       ip-172-31-29-87.us-west-2.compute.internal
> IP:         172.31.29.87
> Port:       37244
> Auth-User:   23
> Auth-Realm: 46.218.201.23
> MWI-Account: 23 at 46.218.201.23
>
> Total items returned: 1
>
> =================================================================================================
>
> Displayed real IP address of NGINX
>
> Also when received INVITE then variables like network_addr will cantain
> real IP of NGINX.
>
> Then you can try confgire nginx like ng
>
> proxy_set_header X-Real-IP $remote_addr;
> proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for;
>
>
> But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every
> SIP message not only connection establishing. NGINX not understand SIP
> messages and cannot do it.
>
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this
> case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port
> with http server?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Alexandru Covalschi
> VoIP engineer and system administrator
> tel: +37367398493 <+373%20673%2098%20493>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 568691 at gmail.com  Wed Jan  4 00:02:49 2017
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Tue, 3 Jan 2017 23:02:49 +0200
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
Message-ID: <937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>

Why do you need it actually? Freeswitch will see proxy IP, yes, but the websocket connection will be bind to one of the ports of haproxy and it should work fine. 
>But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing
First of all Freeswitch doesn't require those headers.
Well for that you should use STUN and set that in your clients application. Another way is to use Kamailio, yes, it can add those on the fly, but again - you must put it instead of the proxy server (which you 
may actually want to do as it is a SIP proxy by-desing and it works with websockets pretty fine).
________________________________
Alexandru Covalschi
VoIP Engineer and System Administrator
tel: +373 673 98 493

> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com> ???????(?):
> 
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port with http server?
> 

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From s.safarov at gmail.com  Wed Jan  4 00:43:24 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 03 Jan 2017 21:43:24 +0000
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
Message-ID: <CAHtxdDfsT-eGxtb9SbQ92SsO85dOy=ZuKuMcKqziewPxbrO7BA@mail.gmail.com>

For particular IP addresses I want write custom dialplan.
And use one entry point for HTTP and WebSocket traffic - nginx or similar
wevserver over https URI.

Sergey

??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com>:

> Why do you need it actually? Freeswitch will see proxy IP, yes, but the
> websocket connection will be bind to one of the ports of haproxy and it
> should work fine.
> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every
> SIP message not only connection establishing
> First of all Freeswitch doesn't require those headers.
> Well for that you should use STUN and set that in your clients
> application. Another way is to use Kamailio, yes, it can add those on the
> fly, but again - you must put it instead of the proxy server (which you
> may actually want to do as it is a SIP proxy by-desing and it works with
> websockets pretty fine).
> ________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <+373%20673%2098%20493>
>
> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com> ???????(?):
>
>
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this
> case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port
> with http server?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 568691 at gmail.com  Wed Jan  4 00:50:02 2017
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Tue, 3 Jan 2017 23:50:02 +0200
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDfsT-eGxtb9SbQ92SsO85dOy=ZuKuMcKqziewPxbrO7BA@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
	<CAHtxdDfsT-eGxtb9SbQ92SsO85dOy=ZuKuMcKqziewPxbrO7BA@mail.gmail.com>
Message-ID: <DA5434D4-2F69-4429-8484-8DC926A0FF6B@gmail.com>

Then you must specify SIP headers from SIPML5 directly as I said before. Actually having webserver serving also as websocket proxy is kinda terrible idea as webrtc clients will share same threads with simple web ones. 

________________________________
Alexandru Covalschi
VoIP Engineer and System Administrator
tel: +373 673 98 493

> 3 ???. 2017 ?., ? 23:43, Sergey Safarov <s.safarov at gmail.com> ???????(?):
> 
> For particular IP addresses I want write custom dialplan.
> And use one entry point for HTTP and WebSocket traffic - nginx or similar wevserver over https URI.
> 
> Sergey
> 
> ??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com <mailto:568691 at gmail.com>>:
> Why do you need it actually? Freeswitch will see proxy IP, yes, but the websocket connection will be bind to one of the ports of haproxy and it should work fine. 
> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing
> First of all Freeswitch doesn't require those headers.
> Well for that you should use STUN and set that in your clients application. Another way is to use Kamailio, yes, it can add those on the fly, but again - you must put it instead of the proxy server (which you 
> may actually want to do as it is a SIP proxy by-desing and it works with websockets pretty fine).
> ________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <tel:+373%20673%2098%20493>
>> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> ???????(?):
> 
>> 
>> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header.
>> Are you can suggest other way to publish FreeSwitch socket on same port with http server?
>> 
> _________________________________________________________________________
> 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 s.safarov at gmail.com  Wed Jan  4 00:51:58 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 03 Jan 2017 21:51:58 +0000
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
Message-ID: <CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>

??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com>:

> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every
> SIP message not only connection establishing
> First of all Freeswitch doesn't require those headers.
>
This header required to auth particular SIP/WebRTC clients by IP addresses
via proxy daemons.

Well for that you should use STUN and set that in your clients application.
> Another way is to use Kamailio, yes, it can add those on the fly, but again
> - you must put it instead of the proxy server (which you
> may actually want to do as it is a SIP proxy by-desing and it works with
> websockets pretty fine).
>
 If i place Kamailio instead of http proxy server, then i must have two
entry points:
1) for http trafic - http server;
2) web-socket trafic - Kamailio.

But i wants avoid configuration with more than one entry point.

Sergey

________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <+373%20673%2098%20493>
>
> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com> ???????(?):
>
>
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this
> case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port
> with http server?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 568691 at gmail.com  Wed Jan  4 00:55:39 2017
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Tue, 3 Jan 2017 23:55:39 +0200
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
	<CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
Message-ID: <F8823253-FEC3-48C7-B507-5C162E6E6B3D@gmail.com>

If you go deeper into websockets you'll understand that WS traffic has very little common with HTTP (actually it is more like HTTP/2) and it is generally a good idea to separate http and ws traffic,
You can join @ru_freeswitch channel on telegram, as what we are discussing now is offtopic on that list.

________________________________
Alexandru Covalschi
VoIP Engineer and System Administrator
tel: +373 673 98 493

> 3 ???. 2017 ?., ? 23:51, Sergey Safarov <s.safarov at gmail.com> ???????(?):
> 
> ??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com <mailto:568691 at gmail.com>>:
> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing
> First of all Freeswitch doesn't require those headers.
> This header required to auth particular SIP/WebRTC clients by IP addresses via proxy daemons.
> 
> Well for that you should use STUN and set that in your clients application. Another way is to use Kamailio, yes, it can add those on the fly, but again - you must put it instead of the proxy server (which you 
> may actually want to do as it is a SIP proxy by-desing and it works with websockets pretty fine).
>  If i place Kamailio instead of http proxy server, then i must have two entry points:
> 1) for http trafic - http server;
> 2) web-socket trafic - Kamailio.
> 
> But i wants avoid configuration with more than one entry point.
> 
> Sergey
> 
> ________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <tel:+373%20673%2098%20493>
>> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> ???????(?):
> 
>> 
>> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header.
>> Are you can suggest other way to publish FreeSwitch socket on same port with http server?
>> 
> _________________________________________________________________________
> 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 s.safarov at gmail.com  Wed Jan  4 00:58:52 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 03 Jan 2017 21:58:52 +0000
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
	<CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
Message-ID: <CAHtxdDc_W_adnkSfYg-JdO5G=xJ31BO9SKHVZHiXjVMd7gARyA@mail.gmail.com>

I think nginx web server is well designed daemon for thousand of clients.
As example
https://www.scalescale.com/tips/nginx/nginx-benchmarking-using-siedge/#


??, 4 ???. 2017 ?. ? 0:51, Sergey Safarov <s.safarov at gmail.com>:

> ??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com>:
>
> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every
> SIP message not only connection establishing
>
> First of all Freeswitch doesn't require those headers.
>
> This header required to auth particular SIP/WebRTC clients by IP addresses
> via proxy daemons.
>
> Well for that you should use STUN and set that in your clients
> application. Another way is to use Kamailio, yes, it can add those on the
> fly, but again - you must put it instead of the proxy server (which you
> may actually want to do as it is a SIP proxy by-desing and it works with
> websockets pretty fine).
>
>  If i place Kamailio instead of http proxy server, then i must have two
> entry points:
> 1) for http trafic - http server;
> 2) web-socket trafic - Kamailio.
>
> But i wants avoid configuration with more than one entry point.
>
> Sergey
>
> ________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <+373%20673%2098%20493>
>
> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com> ???????(?):
>
>
> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this
> case Kamailio can parce http headers and add requred SIP header.
> Are you can suggest other way to publish FreeSwitch socket on same port
> with http server?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 568691 at gmail.com  Wed Jan  4 01:01:57 2017
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Wed, 4 Jan 2017 00:01:57 +0200
Subject: [Freeswitch-users] NGINX + Kamailio + FreeSwitch
In-Reply-To: <CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
References: <CAHtxdDcSieTBsNwrdAc2bJ+Ng5yC+V+7o3BBwJHQ5YVXxwOdNg@mail.gmail.com>
	<CAD_iGr_oM4JZncvOyeeRvxbpzKFmm2hHYVKWUpaVCCSZE8A6uQ@mail.gmail.com>
	<CAHtxdDc+RrsiRoghjbVKcQRR3LfGu+ac=Gw9Jk1GjRu1UgEX1Q@mail.gmail.com>
	<937C69B1-2DD6-4F68-9C4C-4916552398C0@gmail.com>
	<CAHtxdDc03cy2igKqL10QbWZ1ktYnYd+ZMZ0M9JS2PmwAFh6xLw@mail.gmail.com>
Message-ID: <61584BEF-E808-4DE8-9817-B061C1F4E18D@gmail.com>

That example is not about SIP over WebRTC. Well another way which you can try is to perform an HTTP 301 redirect to the address:port you want to reach, however I still think putting Kamailio instead of nginx will let you do a lot more flexible stuff with SIP not changing the client part.

________________________________
Alexandru Covalschi
VoIP Engineer and System Administrator
tel: +373 673 98 493

> 3 ???. 2017 ?., ? 23:51, Sergey Safarov <s.safarov at gmail.com> ???????(?):
> 
> ??, 4 ???. 2017 ?. ? 0:03, Alexandru Covalschi <568691 at gmail.com <mailto:568691 at gmail.com>>:
> >But FreeSwitch wants SIP headers "X-AUTH-IP" and "X-AUTH-PORT" in every SIP message not only connection establishing
> First of all Freeswitch doesn't require those headers.
> This header required to auth particular SIP/WebRTC clients by IP addresses via proxy daemons.
> 
> Well for that you should use STUN and set that in your clients application. Another way is to use Kamailio, yes, it can add those on the fly, but again - you must put it instead of the proxy server (which you 
> may actually want to do as it is a SIP proxy by-desing and it works with websockets pretty fine).
>  If i place Kamailio instead of http proxy server, then i must have two entry points:
> 1) for http trafic - http server;
> 2) web-socket trafic - Kamailio.
> 
> But i wants avoid configuration with more than one entry point.
> 
> Sergey
> 
> ________________________________
> Alexandru Covalschi
> VoIP Engineer and System Administrator
> tel: +373 673 98 493 <tel:+373%20673%2098%20493>
>> 3 ???. 2017 ?., ? 22:47, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> ???????(?):
> 
>> 
>> Then you will try cofigure Kamailio between  NGINX and FreeSwitch. In this case Kamailio can parce http headers and add requred SIP header.
>> Are you can suggest other way to publish FreeSwitch socket on same port with http server?
>> 
> _________________________________________________________________________
> 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 loi.dangthanh at gmail.com  Wed Jan  4 13:15:32 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Wed, 4 Jan 2017 17:15:32 +0700
Subject: [Freeswitch-users] Running show codec via fs_cli
Message-ID: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>

Hi guys, I'm connecting to my backgrounded FS via fs_cli, things are pretty
straightforward, and connection is OK. But I found this:
`show codec` returns 0 total.

I can make sure CORE_PCM_MODULE is loaded, via log file and api
`module_exists`.

Then I tried to `reload` CORE_PCM_MODULE, and guess what, `show codec`
works properly, contains codecs informations.

Same issue happened with CORE_SPEEX_MODULE and CORE_VPX_MODULE as well.

FS running normally before and after reloading module(s).
Not sure if this is a bug on `show codec` or I just miss something.

More information, connecting to a foreground running FS doesn't have the
same issue.

rgds,

Loi Dang Thanh
Phone : +84. 1224.735.448
Email : loi.dangthanh at gmail.com
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From loi.dangthanh at gmail.com  Wed Jan  4 13:49:34 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Wed, 4 Jan 2017 17:49:34 +0700
Subject: [Freeswitch-users] Codec transcoding PCMA to PMCU
In-Reply-To: <CAGqqW9-zN8pdmqQcy_zpyqc6QNd9b_=knke1X3DX2OyOhU-qsg@mail.gmail.com>
References: <CAGqqW99s1fYvZN8dg_hngWPhBaDfdNz+d1eJ-A6W_GhKxyfdNQ@mail.gmail.com>
	<CAGqqW9-zN8pdmqQcy_zpyqc6QNd9b_=knke1X3DX2OyOhU-qsg@mail.gmail.com>
Message-ID: <CAGsrq_tXPUO1gcy6TCGJuYnKegf0w3K1ZNBPumANkLs3TYvJzA@mail.gmail.com>

hi Ketan, pls check if `inbound-proxy-media` is true in your sip profile.
if yes, set it to false.
beside, export absolute_codec_string=PCMU to calling leg.

<action application="bridge" data="{absolute_codec_string=PCMU
}sofia/gateway/Twilio/+34915300123"/>

or

<action application="export" data="absolute_codec_string=PCMU"/>

good luck.
rgds,

Loi Dang Thanh
Phone : +84. 1224.735.448
Email : loi.dangthanh at gmail.com

On Mon, Jan 2, 2017 at 5:04 PM, Ketan Kothari <kkothari157 at gmail.com> wrote:

> Hello,
>
> This is my dialplan.
>
> <section name="dialplan" description="My Dialplan">
> <context name="default">
> <extension name="22290915300123">
> <condition field="destination_number" expression="22290915300123">
> <action application="set" data="effective_destination_
> number=22290915300123"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="export" data="media_bug_answer_req=true"/>
> <action application="export" data="RECORD_STEREO=true"/>
> <action application="export" data="record_sample_rate=8000"/>
> <action application="export" data="execute_on_answer=record_session
> /home/shared/recordings/c9b95b66-efcb-45a9-89b7-b6e39f4a2c4b.wav"/>
> <action application="set" data="absolute_codec_string=PCMU"/>
> <action application="bridge" data="sofia/gateway/Twilio/+34915300123"/>
> </condition>
> </extension>
> </context>
> </section>
>
> On Tue, Dec 27, 2016 at 4:57 PM, Ketan Kothari <kkothari157 at gmail.com>
> wrote:
>
>> Hello,
>>
>> I have configured PCMA codec in my zoipper soft-phone(UAC) and tried to
>> call to 123456789 using our gateway twillo(UAS).
>> Gateway only accept PCMU codec. I have also configured absolute codec
>> =PCMU but still freeswitch sending PCMA to twillo(Provider gateway UAS). I
>> have got error Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED in
>> freeswitch.
>>
>> Once i have added PCMU codec in my zoiper(UAC) it's working.
>>
>>
>> [image: Inline image 1]
>>
>>
>> What am i missing here?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Jan  4 20:07:40 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 4 Jan 2017 11:07:40 -0600
Subject: [Freeswitch-users] Combine non-auth and auth calls on same
 profile security consideration.
In-Reply-To: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>
References: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>
Message-ID: <CAKbxfG8KmeBuwGiWQbUMcRVHYqoSYVgwqEWGhPGD1xjvQ580aw@mail.gmail.com>

On Tue, Jan 3, 2017 at 3:40 AM, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
>
> There where separate sip profiles for registered accounts and anonymous
> inbound calls: internal and external. All calls and registrations on
> internal a authenticated and routed to "default" context. For external
> profile inbound calls are not authenticated and are routed to "public"
> context.
>
> In order for some one to call inbound (s)he have to use :5080 port,
> which is somehow inconvenient. So I decided to combine in one profile
> named "example" on port 5060.
>
> <profile name="internal_77.89.245.34">
> <settings>
> <param name="context" value="public"/>
> <param name="auth-calls" value="false"/>
> <param name="auth-all-packets" value="false"/>
> <param name="inbound-reg-force-matching-username" value="true"/>
> <param name="force-register-domain" value="default"/>
> <param name="force-subscription-domain" value="default"/>
> <param name="force-register-db-domain" value="default"/>
> </settings>
> </profile>
> <domain name="$${domain}">
> <variables>
> <variable name="inbound-reg-force-matching-username" value="true"/>
> <variable name="user_context" value="default"/>
> <groups>
> <group name="default">
> <users>
> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
> </users>
> </group>
> </groups>
> </variables>
> </domain>
>
> Registration was working, calling to those registration was working, but
> some phones hit public context, some default context. I started to dig
> whats happening and found that D-Link phones always do authenticated
> calls when they a registered, while Stephen's phones does
> unauthenticated and go to public.
>
> Then I found a mention that registrations allow only to find the phone
> to call, while calls does not necessary authenticate. So I used
> Anthony's solution in public context on the end:
>
> <extension name="check_auth" continue="true">
> <condition field="${sip_authorized}" expression="^true$" break="never">
> <anti-action application="respond" data="407"/>
> </condition>
> </extension>
>
> Now when a registered phone hits public context and no other conditions
> are met, they are rejected and call authenticated hitting default context.
>
> Then in CDR I see two lines: one with rejected and one with success.
> This is not well, so found a hint and put:
> <anti-action application="set" data="process_cdr=false"/>
>
> And now is working somewhat correct except that calls to public numbers
> which does not require authentication does not get vars for the
> registered extension.
>
> My questions are:
>
> 1) Does this type of combination affect security?
> 2) How to impose all registered phones to make authenticated calls
> always? So they will not go first thru public context and then to default?
>
>
Its a lot to go through for the vanity of not having to type 5080 once in a
config box, but that's just my opinion ;)

You can use the set_user app to make unauthenticated calls get the same
data as authenticated calls would have on a specified exten.




> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From brian at freeswitch.org  Thu Jan  5 02:59:48 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 Jan 2017 17:59:48 -0600
Subject: [Freeswitch-users] Running show codec via fs_cli
In-Reply-To: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>
References: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>
Message-ID: <CAEJMVkA5KyLCuXdJ9yZ2W35dn0JuBmqJkp2+CNryqV001PNhTg@mail.gmail.com>

That probably means your sqlite db is toast or you started with the db
disabled, When you type show codecs, it doesn't actually walk the in memory
data to show you that, Its pulling it from the database on disk, same with
channels and many others.

/b


On Wed, Jan 4, 2017 at 4:15 AM, L?i ??ng <loi.dangthanh at gmail.com> wrote:

> Hi guys, I'm connecting to my backgrounded FS via fs_cli, things are
> pretty straightforward, and connection is OK. But I found this:
> `show codec` returns 0 total.
>
> I can make sure CORE_PCM_MODULE is loaded, via log file and api
> `module_exists`.
>
> Then I tried to `reload` CORE_PCM_MODULE, and guess what, `show codec`
> works properly, contains codecs informations.
>
> Same issue happened with CORE_SPEEX_MODULE and CORE_VPX_MODULE as well.
>
> FS running normally before and after reloading module(s).
> Not sure if this is a bug on `show codec` or I just miss something.
>
> More information, connecting to a foreground running FS doesn't have the
> same issue.
>
> rgds,
>
> Loi Dang Thanh
> Phone : +84. 1224.735.448
> Email : loi.dangthanh at gmail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com (50% Discount using code FreeSwitch50)
http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50)
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From david.villasmil.work at gmail.com  Thu Jan  5 07:14:51 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 05 Jan 2017 04:14:51 +0000
Subject: [Freeswitch-users] Fs_path
Message-ID: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>

Hello guys,

I've been wondering where the "fs_path" parameter comes from? How is it
built, etc? I've been seeing cases where when the client has no "fs_path",
fs can't find the user and returns temp unavailable...

Thanks all,

David
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From loi.dangthanh at gmail.com  Thu Jan  5 09:48:26 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Thu, 5 Jan 2017 13:48:26 +0700
Subject: [Freeswitch-users] Running show codec via fs_cli
In-Reply-To: <CAEJMVkA5KyLCuXdJ9yZ2W35dn0JuBmqJkp2+CNryqV001PNhTg@mail.gmail.com>
References: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>
	<CAEJMVkA5KyLCuXdJ9yZ2W35dn0JuBmqJkp2+CNryqV001PNhTg@mail.gmail.com>
Message-ID: <CAGsrq_vPRcyYdzYjzbBhqxu556PRuxeOxTLaeopSUr4o--gAVg@mail.gmail.com>

Hi Brian, tks for pointing out the scenario.
afaik,
1. In start, FS firstly open db connection
2. In loading core modules, CORE_PCM_MODULE, FS put codec informations into
sqlitedb (default)
3. `show codecs` in mod_commands just to retrieve that these informations
from db.
Everythings ok when FS is running foreground, I can check the log in
console and it's straightforward.

The problem was, when I'm running FS in background, looks like step 2 was
not executed, so my step 3 return `0 total`.
Not sure what happened, since logfile module is loaded after loading core
module, so that I could not check logfile for details in step 1 and 2.

Not sure if there is difference between running in foreground and
background affect this.

rgds,

Loi Dang Thanh
Phone : +84. 1224.735.448
Email : loi.dangthanh at gmail.com

On Thu, Jan 5, 2017 at 6:59 AM, Brian West <brian at freeswitch.org> wrote:

> That probably means your sqlite db is toast or you started with the db
> disabled, When you type show codecs, it doesn't actually walk the in memory
> data to show you that, Its pulling it from the database on disk, same with
> channels and many others.
>
> /b
>
>
> On Wed, Jan 4, 2017 at 4:15 AM, L?i ??ng <loi.dangthanh at gmail.com> wrote:
>
>> Hi guys, I'm connecting to my backgrounded FS via fs_cli, things are
>> pretty straightforward, and connection is OK. But I found this:
>> `show codec` returns 0 total.
>>
>> I can make sure CORE_PCM_MODULE is loaded, via log file and api
>> `module_exists`.
>>
>> Then I tried to `reload` CORE_PCM_MODULE, and guess what, `show codec`
>> works properly, contains codecs informations.
>>
>> Same issue happened with CORE_SPEEX_MODULE and CORE_VPX_MODULE as well.
>>
>> FS running normally before and after reloading module(s).
>> Not sure if this is a bug on `show codec` or I just miss something.
>>
>> More information, connecting to a foreground running FS doesn't have the
>> same issue.
>>
>> rgds,
>>
>> Loi Dang Thanh
>> Phone : +84. 1224.735.448
>> Email : loi.dangthanh at gmail.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50)
> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50)
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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 v.zakhozhai at gmail.com  Thu Jan  5 15:11:42 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 5 Jan 2017 14:11:42 +0200
Subject: [Freeswitch-users] Fs_path
In-Reply-To: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
References: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
Message-ID: <CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>

Hi David,

there is no magick with fs_path. fs_path contains value from Path header of
a request which comes from sip proxy, i.e. kamailio. It means that replies
and SIP requests from freeswitch to UAC will traverse SIP proxy which
should be on the path.

For example,
Request: UAC ==> SIP proxy/load balancer ==> freeswitch box
Reply: freeswitch box ==> SIP proxy/load balancer ==> UAC
Request (INVITE): freeswitch box ==> SIP proxy/load balancer ==> UAC

Without path specified freeswitch will send direct messages to UAC:

Request: UAC ==> SIP proxy ==> freeswitch box
Reply: freeswitch box ==> UAC

This behaviour may be undesirable.

So SIP proxy inserts itself to Path header and freeswitch recognizes it and
add fs_path tag to contact.

2017-01-05 6:14 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:

> Hello guys,
>
> I've been wondering where the "fs_path" parameter comes from? How is it
> built, etc? I've been seeing cases where when the client has no "fs_path",
> fs can't find the user and returns temp unavailable...
>
> Thanks all,
>
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
? ?????????,
????????? ???????
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From vbvbrj at gmail.com  Thu Jan  5 15:57:55 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 5 Jan 2017 14:57:55 +0200
Subject: [Freeswitch-users] Combine non-auth and auth calls on same
 profile security consideration.
In-Reply-To: <CAKbxfG8KmeBuwGiWQbUMcRVHYqoSYVgwqEWGhPGD1xjvQ580aw@mail.gmail.com>
References: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>
	<CAKbxfG8KmeBuwGiWQbUMcRVHYqoSYVgwqEWGhPGD1xjvQ580aw@mail.gmail.com>
Message-ID: <9aed8282-b05f-b599-93ba-ead07f60b333@gmail.com>

On 04.01.2017 19:07, Anthony Minessale wrote:
>     My questions are:
>
>     1) Does this type of combination affect security?
>     2) How to impose all registered phones to make authenticated calls
>     always? So they will not go first thru public context and then to
>     default?
>
>
> Its a lot to go through for the vanity of not having to type 5080 once
> in a config box, but that's just my opinion ;)
>
> You can use the set_user app to make unauthenticated calls get the same
> data as authenticated calls would have on a specified exten.

Anthony thank you for suggestion.

Taking this public dialplan:

<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<extension name="sip_nat_detected" continue="true">
<condition field="${sip_nat_detected}" expression="^true$">
<action application="set" inline="true" 
data="sip-force-contact=NDLB-connectile-dysfunction"/>
</condition>
<extension name="public_did">
<condition field="destination_number" expression="^([5-8]\d\d)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
</extension>
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="set" data="process_cdr=false"/>
<anti-action application="respond" data="407"/>
</condition>
</extension>
</context>

Where to put the set_user app? If I'll put it before "check_auth" 
extension like:

<action application="set_user" data="${caller_id_number}@${domain}"/>

Then any one calling from internet could set theirs caller_id_number to 
internal's one and act on behalf of some registered user to fraud.

Or may be first extension in public dialplan to put something which will 
check, based on caller_id_number, if there is a registered user and 
impose to make authenticate call, like:

<extension name="check_user_registered">
<condition field="${sofia_contact(${caller_id_number})}" 
expression="^error/" break="on-false">
<anti-action application="set" data="process_cdr=false"/>
<anti-action application="respond" data="407"/>
</condition>
</extension>

PS: Yes, its not to big to add :5080, but take callers that want to call 
from mobile via internet using a sip uri. Even myself forgets to add 
port number at the end. :)

-- 
Mimiko desu.


From rtreleaven at bunnykick.ca  Thu Jan  5 16:19:47 2017
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Thu, 5 Jan 2017 08:19:47 -0500
Subject: [Freeswitch-users] Combine non-auth and auth calls on same
 profile security consideration.
In-Reply-To: <9aed8282-b05f-b599-93ba-ead07f60b333@gmail.com>
References: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>
	<CAKbxfG8KmeBuwGiWQbUMcRVHYqoSYVgwqEWGhPGD1xjvQ580aw@mail.gmail.com>
	<9aed8282-b05f-b599-93ba-ead07f60b333@gmail.com>
Message-ID: <CAHzfiZ6bNZkkfgSwQevB+krJWza8sqJG=UwyiJvm10XcF6GKiA@mail.gmail.com>

Wouldn't it be simpler to make :5060 the unauthenticated port and :5080 the
authenticated?

On Jan 5, 2017 7:58 AM, "Mimiko" <vbvbrj at gmail.com> wrote:

> On 04.01.2017 19:07, Anthony Minessale wrote:
> >     My questions are:
> >
> >     1) Does this type of combination affect security?
> >     2) How to impose all registered phones to make authenticated calls
> >     always? So they will not go first thru public context and then to
> >     default?
> >
> >
> > Its a lot to go through for the vanity of not having to type 5080 once
> > in a config box, but that's just my opinion ;)
> >
> > You can use the set_user app to make unauthenticated calls get the same
> > data as authenticated calls would have on a specified exten.
>
> Anthony thank you for suggestion.
>
> Taking this public dialplan:
>
> <context name="public">
> <extension name="unloop">
> <condition field="${unroll_loops}" expression="^true$"/>
> <condition field="${sip_looped_call}" expression="^true$">
> <action application="deflect" data="${destination_number}"/>
> </condition>
> </extension>
> <extension name="sip_nat_detected" continue="true">
> <condition field="${sip_nat_detected}" expression="^true$">
> <action application="set" inline="true"
> data="sip-force-contact=NDLB-connectile-dysfunction"/>
> </condition>
> <extension name="public_did">
> <condition field="destination_number" expression="^([5-8]\d\d)$">
> <action application="transfer" data="$1 XML default"/>
> </condition>
> </extension>
> </extension>
> <extension name="check_auth" continue="true">
> <condition field="${sip_authorized}" expression="^true$" break="never">
> <anti-action application="set" data="process_cdr=false"/>
> <anti-action application="respond" data="407"/>
> </condition>
> </extension>
> </context>
>
> Where to put the set_user app? If I'll put it before "check_auth"
> extension like:
>
> <action application="set_user" data="${caller_id_number}@${domain}"/>
>
> Then any one calling from internet could set theirs caller_id_number to
> internal's one and act on behalf of some registered user to fraud.
>
> Or may be first extension in public dialplan to put something which will
> check, based on caller_id_number, if there is a registered user and
> impose to make authenticate call, like:
>
> <extension name="check_user_registered">
> <condition field="${sofia_contact(${caller_id_number})}"
> expression="^error/" break="on-false">
> <anti-action application="set" data="process_cdr=false"/>
> <anti-action application="respond" data="407"/>
> </condition>
> </extension>
>
> PS: Yes, its not to big to add :5080, but take callers that want to call
> from mobile via internet using a sip uri. Even myself forgets to add
> port number at the end. :)
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jose.lopes at itcenter.com.pt  Thu Jan  5 17:35:45 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Thu, 5 Jan 2017 14:35:45 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call to
	sip external voicemail
Message-ID: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>

Hello Guys,

I have audio cut off at the begin of the verto call to  FreeSwitch that
redirect to sip external voicemail (Access voicemail mailbox) .

This happen when I use PCMU at verto codecs and sip codecs (if i use opus
at verto codecs, there is no issue, but this causes audio transcoding) .

At dialplan i used the example "Bridging from WebRTC (mod_verto) to
PSTN/ITSPs" from
https://freeswitch.org/confluence/display/FREESWITCH/mod_verto.
I notice if i remove the playback action, there is no issue. But I need the
playback action to send rtp packets to verto client.

I simulate this using another FreeSwitch as external voicemail server and I
only listen "id followed by pound" from the initial message of voicemail
("Please enter your id followed by pound").
The log of this call is at https://pastebin.freeswitch.org/view/507fa115

What I can do to use PCMU at verto codecs and sip codecs on type of call?
Should i open a issue on FreeSwitch JIRA ?


Best regards,
Jose Lopes
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From mike at jerris.com  Thu Jan  5 17:42:23 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 05 Jan 2017 14:42:23 +0000
Subject: [Freeswitch-users] Running show codec via fs_cli
In-Reply-To: <CAGsrq_vPRcyYdzYjzbBhqxu556PRuxeOxTLaeopSUr4o--gAVg@mail.gmail.com>
References: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>
	<CAEJMVkA5KyLCuXdJ9yZ2W35dn0JuBmqJkp2+CNryqV001PNhTg@mail.gmail.com>
	<CAGsrq_vPRcyYdzYjzbBhqxu556PRuxeOxTLaeopSUr4o--gAVg@mail.gmail.com>
Message-ID: <CAEwTsAh2i1sNFSxUEaRL45q=T5k1fktwAY3WOB1VtW+nbGuL1A@mail.gmail.com>

what version of freeswitch?

On Thu, Jan 5, 2017 at 1:54 AM L?i ??ng <loi.dangthanh at gmail.com> wrote:

> Hi Brian, tks for pointing out the scenario.
> afaik,
> 1. In start, FS firstly open db connection
> 2. In loading core modules, CORE_PCM_MODULE, FS put codec informations
> into sqlitedb (default)
> 3. `show codecs` in mod_commands just to retrieve that these informations
> from db.
> Everythings ok when FS is running foreground, I can check the log in
> console and it's straightforward.
>
> The problem was, when I'm running FS in background, looks like step 2 was
> not executed, so my step 3 return `0 total`.
> Not sure what happened, since logfile module is loaded after loading core
> module, so that I could not check logfile for details in step 1 and 2.
>
> Not sure if there is difference between running in foreground and
> background affect this.
>
> rgds,
>
> Loi Dang Thanh
> Phone : +84. 1224.735.448
> Email : loi.dangthanh at gmail.com
>
>
>
> On Thu, Jan 5, 2017 at 6:59 AM, Brian West <brian at freeswitch.org> wrote:
>
> That probably means your sqlite db is toast or you started with the db
> disabled, When you type show codecs, it doesn't actually walk the in memory
> data to show you that, Its pulling it from the database on disk, same with
> channels and many others.
>
> /b
>
>
> On Wed, Jan 4, 2017 at 4:15 AM, L?i ??ng <loi.dangthanh at gmail.com> wrote:
>
> Hi guys, I'm connecting to my backgrounded FS via fs_cli, things are
> pretty straightforward, and connection is OK. But I found this:
> `show codec` returns 0 total.
>
> I can make sure CORE_PCM_MODULE is loaded, via log file and api
> `module_exists`.
>
> Then I tried to `reload` CORE_PCM_MODULE, and guess what, `show codec`
> works properly, contains codecs informations.
>
> Same issue happened with CORE_SPEEX_MODULE and CORE_VPX_MODULE as well.
>
> FS running normally before and after reloading module(s).
> Not sure if this is a bug on `show codec` or I just miss something.
>
> More information, connecting to a foreground running FS doesn't have the
> same issue.
>
> rgds,
>
> Loi Dang Thanh
> Phone : +84. 1224.735.448
> Email : loi.dangthanh at gmail.com
>
>
>
>
>
> _________________________________________________________________________
>
>
> Professional FreeSWITCH Consulting Services:
>
>
> consulting at freeswitch.org
>
>
> http://www.freeswitchsolutions.com
>
>
>
>
>
> Official FreeSWITCH Sites
>
>
> http://www.freeswitch.org
>
>
> http://confluence.freeswitch.org
>
>
> http://www.cluecon.com
>
>
>
>
>
> FreeSWITCH-users mailing list
>
>
> FreeSWITCH-users at lists.freeswitch.org
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
> http://www.freeswitch.org
>
>
>
>
> --
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> *Brian West*
> brian at freeswitch.org
>
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50)
> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50)
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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
>
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From mike at jerris.com  Thu Jan  5 17:45:30 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 05 Jan 2017 14:45:30 +0000
Subject: [Freeswitch-users] Fs_path
In-Reply-To: <CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>
References: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
	<CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>
Message-ID: <CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>

only partially correct here.  The main place fs_path is inserted is based
on the nat detection settings you configure in freeswitch itself.  When we
get a registration that's detected behind nat we write down the ip/port the
request came from in fs_path before we store the registration, the. when
calling that ref we will send the invite to that ip/port

On Thu, Jan 5, 2017 at 7:16 AM Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
wrote:

> Hi David,
>
> there is no magick with fs_path. fs_path contains value from Path header
> of a request which comes from sip proxy, i.e. kamailio. It means that
> replies and SIP requests from freeswitch to UAC will traverse SIP proxy
> which should be on the path.
>
> For example,
> Request: UAC ==> SIP proxy/load balancer ==> freeswitch box
> Reply: freeswitch box ==> SIP proxy/load balancer ==> UAC
> Request (INVITE): freeswitch box ==> SIP proxy/load balancer ==> UAC
>
> Without path specified freeswitch will send direct messages to UAC:
>
> Request: UAC ==> SIP proxy ==> freeswitch box
> Reply: freeswitch box ==> UAC
>
> This behaviour may be undesirable.
>
> So SIP proxy inserts itself to Path header and freeswitch recognizes it
> and add fs_path tag to contact.
>
> 2017-01-05 6:14 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>
> :
>
> Hello guys,
>
> I've been wondering where the "fs_path" parameter comes from? How is it
> built, etc? I've been seeing cases where when the client has no "fs_path",
> fs can't find the user and returns temp unavailable...
>
> Thanks all,
>
> David
>
>
>
> _________________________________________________________________________
>
>
> Professional FreeSWITCH Consulting Services:
>
>
> consulting at freeswitch.org
>
>
> http://www.freeswitchsolutions.com
>
>
>
>
>
> Official FreeSWITCH Sites
>
>
> http://www.freeswitch.org
>
>
> http://confluence.freeswitch.org
>
>
> http://www.cluecon.com
>
>
>
>
>
> FreeSWITCH-users mailing list
>
>
> FreeSWITCH-users at lists.freeswitch.org
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
> http://www.freeswitch.org
>
>
>
>
> --
> ? ?????????,
> ????????? ???????
>
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
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From bipin at xbipin.com  Thu Jan  5 18:29:39 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 5 Jan 2017 19:29:39 +0400
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
Message-ID: <4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>

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From v.zakhozhai at gmail.com  Thu Jan  5 18:45:25 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Thu, 5 Jan 2017 17:45:25 +0200
Subject: [Freeswitch-users] Fs_path
In-Reply-To: <CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>
References: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
	<CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>
	<CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>
Message-ID: <CANsY4D+W6+eC9GHe3rePGw8KwxuPkOfMSmGpbu208pPSN3jBMA@mail.gmail.com>

Michael,

thank you for clarification of fs_path tag's meaning. I guess it is David's
use case.

2017-01-05 16:45 GMT+02:00 Michael Jerris <mike at jerris.com>:

> only partially correct here.  The main place fs_path is inserted is based
> on the nat detection settings you configure in freeswitch itself.  When we
> get a registration that's detected behind nat we write down the ip/port the
> request came from in fs_path before we store the registration, the. when
> calling that ref we will send the invite to that ip/port
>
> On Thu, Jan 5, 2017 at 7:16 AM Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
> wrote:
>
>> Hi David,
>>
>> there is no magick with fs_path. fs_path contains value from Path header
>> of a request which comes from sip proxy, i.e. kamailio. It means that
>> replies and SIP requests from freeswitch to UAC will traverse SIP proxy
>> which should be on the path.
>>
>> For example,
>> Request: UAC ==> SIP proxy/load balancer ==> freeswitch box
>> Reply: freeswitch box ==> SIP proxy/load balancer ==> UAC
>> Request (INVITE): freeswitch box ==> SIP proxy/load balancer ==> UAC
>>
>> Without path specified freeswitch will send direct messages to UAC:
>>
>> Request: UAC ==> SIP proxy ==> freeswitch box
>> Reply: freeswitch box ==> UAC
>>
>> This behaviour may be undesirable.
>>
>> So SIP proxy inserts itself to Path header and freeswitch recognizes it
>> and add fs_path tag to contact.
>>
>> 2017-01-05 6:14 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com
>> >:
>>
>> Hello guys,
>>
>> I've been wondering where the "fs_path" parameter comes from? How is it
>> built, etc? I've been seeing cases where when the client has no "fs_path",
>> fs can't find the user and returns temp unavailable...
>>
>> Thanks all,
>>
>> David
>>
>>
>>
>> _________________________________________________________________________
>>
>>
>> Professional FreeSWITCH Consulting Services:
>>
>>
>> consulting at freeswitch.org
>>
>>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>>
>> http://www.freeswitch.org
>>
>>
>> http://confluence.freeswitch.org
>>
>>
>> http://www.cluecon.com
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> ? ?????????,
>> ????????? ???????
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>> Professional FreeSWITCH Consulting Services:
>>
>> consulting at freeswitch.org
>>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> http://www.freeswitch.org
>>
>> http://confluence.freeswitch.org
>>
>> http://www.cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
? ?????????,
????????? ???????
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From brian at freeswitch.org  Thu Jan  5 18:47:20 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Jan 2017 09:47:20 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
Message-ID: <CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>

Prefix them with silence_stream://2000 or 3000 and it should go away.

/b


On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> i have the same issue, i think its related to slow audio setup during the
> call
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
> to sip external voicemail
> From: Jos? Lopes <jose.lopes at itcenter.com.pt> <jose.lopes at itcenter.com.pt>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/5/2017, 6:35:45 PM
>
> Hello Guys,
>
> I have audio cut off at the begin of the verto call to  FreeSwitch that
> redirect to sip external voicemail (Access voicemail mailbox) .
>
> This happen when I use PCMU at verto codecs and sip codecs (if i use opus
> at verto codecs, there is no issue, but this causes audio transcoding) .
>
> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
> PSTN/ITSPs" from https://freeswitch.org/confluence/display/FREESWITCH/
> mod_verto.
> I notice if i remove the playback action, there is no issue. But I need
> the playback action to send rtp packets to verto client.
>
> I simulate this using another FreeSwitch as external voicemail server and
> I only listen "id followed by pound" from the initial message of voicemail
> ("Please enter your id followed by pound").
> The log of this call is at https://pastebin.freeswitch.org/view/507fa115
>
> What I can do to use PCMU at verto codecs and sip codecs on type of call?
> Should i open a issue on FreeSwitch JIRA ?
>
>
> Best regards,
> Jose Lopes
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From loi.dangthanh at gmail.com  Thu Jan  5 18:48:58 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Thu, 5 Jan 2017 22:48:58 +0700
Subject: [Freeswitch-users] Running show codec via fs_cli
In-Reply-To: <CAEwTsAh2i1sNFSxUEaRL45q=T5k1fktwAY3WOB1VtW+nbGuL1A@mail.gmail.com>
References: <CAGsrq_ufyR1=yGHkwTvBG9PfaBLSZie_ePf_i9GjzB1ckqXD8w@mail.gmail.com>
	<CAEJMVkA5KyLCuXdJ9yZ2W35dn0JuBmqJkp2+CNryqV001PNhTg@mail.gmail.com>
	<CAGsrq_vPRcyYdzYjzbBhqxu556PRuxeOxTLaeopSUr4o--gAVg@mail.gmail.com>
	<CAEwTsAh2i1sNFSxUEaRL45q=T5k1fktwAY3WOB1VtW+nbGuL1A@mail.gmail.com>
Message-ID: <CAGsrq_sBoGww9+mrmnnq2KfOLBa5+U_v3eA2F7RTjYVQ=XqG4Q@mail.gmail.com>

It's 1.6.13, NOT from git.

rgds,

Loi Dang Thanh
Phone : +84. 1224.735.448
Email : loi.dangthanh at gmail.com

On Thu, Jan 5, 2017 at 9:42 PM, Michael Jerris <mike at jerris.com> wrote:

> what version of freeswitch?
>
> On Thu, Jan 5, 2017 at 1:54 AM L?i ??ng <loi.dangthanh at gmail.com> wrote:
>
>> Hi Brian, tks for pointing out the scenario.
>> afaik,
>> 1. In start, FS firstly open db connection
>> 2. In loading core modules, CORE_PCM_MODULE, FS put codec informations
>> into sqlitedb (default)
>> 3. `show codecs` in mod_commands just to retrieve that these informations
>> from db.
>> Everythings ok when FS is running foreground, I can check the log in
>> console and it's straightforward.
>>
>> The problem was, when I'm running FS in background, looks like step 2 was
>> not executed, so my step 3 return `0 total`.
>> Not sure what happened, since logfile module is loaded after loading core
>> module, so that I could not check logfile for details in step 1 and 2.
>>
>> Not sure if there is difference between running in foreground and
>> background affect this.
>>
>> rgds,
>>
>> Loi Dang Thanh
>> Phone : +84. 1224.735.448
>> Email : loi.dangthanh at gmail.com
>>
>>
>>
>> On Thu, Jan 5, 2017 at 6:59 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> That probably means your sqlite db is toast or you started with the db
>> disabled, When you type show codecs, it doesn't actually walk the in memory
>> data to show you that, Its pulling it from the database on disk, same with
>> channels and many others.
>>
>> /b
>>
>>
>> On Wed, Jan 4, 2017 at 4:15 AM, L?i ??ng <loi.dangthanh at gmail.com> wrote:
>>
>> Hi guys, I'm connecting to my backgrounded FS via fs_cli, things are
>> pretty straightforward, and connection is OK. But I found this:
>> `show codec` returns 0 total.
>>
>> I can make sure CORE_PCM_MODULE is loaded, via log file and api
>> `module_exists`.
>>
>> Then I tried to `reload` CORE_PCM_MODULE, and guess what, `show codec`
>> works properly, contains codecs informations.
>>
>> Same issue happened with CORE_SPEEX_MODULE and CORE_VPX_MODULE as well.
>>
>> FS running normally before and after reloading module(s).
>> Not sure if this is a bug on `show codec` or I just miss something.
>>
>> More information, connecting to a foreground running FS doesn't have the
>> same issue.
>>
>> rgds,
>>
>> Loi Dang Thanh
>> Phone : +84. 1224.735.448
>> Email : loi.dangthanh at gmail.com
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>>
>> Professional FreeSWITCH Consulting Services:
>>
>>
>> consulting at freeswitch.org
>>
>>
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>>
>> http://www.freeswitch.org
>>
>>
>> http://confluence.freeswitch.org
>>
>>
>> http://www.cluecon.com
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com (50% Discount using code FreeSwitch50)
>> http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50)
>> https://www.gofundme.com/freeswitch_ubuntu
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *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 vbvbrj at gmail.com  Thu Jan  5 19:43:50 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 5 Jan 2017 18:43:50 +0200
Subject: [Freeswitch-users] Combine non-auth and auth calls on same
 profile security consideration.
In-Reply-To: <CAHzfiZ6bNZkkfgSwQevB+krJWza8sqJG=UwyiJvm10XcF6GKiA@mail.gmail.com>
References: <30fc247d-c185-08e2-a7de-348a0a657b16@gmail.com>
	<CAKbxfG8KmeBuwGiWQbUMcRVHYqoSYVgwqEWGhPGD1xjvQ580aw@mail.gmail.com>
	<9aed8282-b05f-b599-93ba-ead07f60b333@gmail.com>
	<CAHzfiZ6bNZkkfgSwQevB+krJWza8sqJG=UwyiJvm10XcF6GKiA@mail.gmail.com>
Message-ID: <9e1102f1-4311-66d2-82ee-cc8077d58319@gmail.com>

On 05.01.2017 15:19, Russell Treleaven wrote:
> Wouldn't it be simpler to make :5060 the unauthenticated port and :5080
> the authenticated?

Maybe. For now I have more registered users than anonymous calls from 
internet. So, when configuring some device I easily can forget about 
port. :)
Any way, its more about my questions, than switching ports. I considered 
this option and is last resort.

-- 
Mimiko desu.


From david.villasmil.work at gmail.com  Thu Jan  5 19:44:50 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 05 Jan 2017 16:44:50 +0000
Subject: [Freeswitch-users] Fs_path
In-Reply-To: <CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>
References: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
	<CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>
	<CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>
Message-ID: <CAFGRPVoBOJf5nAjinyy_GTmYm66+hLNo5XpNTwAhJUjX=HWVRg@mail.gmail.com>

So if i understand correctly, even if fs_path is not there, there is no
reason for fs not FINDING the registered user, i.e. Not even sending an
invite request to said user and stating in the cli the user was not found?

On Thu, Jan 5, 2017 at 10:46 AM Michael Jerris <mike at jerris.com> wrote:

> only partially correct here.  The main place fs_path is inserted is based
> on the nat detection settings you configure in freeswitch itself.  When we
> get a registration that's detected behind nat we write down the ip/port the
> request came from in fs_path before we store the registration, the. when
> calling that ref we will send the invite to that ip/port
>
> On Thu, Jan 5, 2017 at 7:16 AM Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
> wrote:
>
> Hi David,
>
> there is no magick with fs_path. fs_path contains value from Path header
> of a request which comes from sip proxy, i.e. kamailio. It means that
> replies and SIP requests from freeswitch to UAC will traverse SIP proxy
> which should be on the path.
>
> For example,
> Request: UAC ==> SIP proxy/load balancer ==> freeswitch box
> Reply: freeswitch box ==> SIP proxy/load balancer ==> UAC
> Request (INVITE): freeswitch box ==> SIP proxy/load balancer ==> UAC
>
> Without path specified freeswitch will send direct messages to UAC:
>
> Request: UAC ==> SIP proxy ==> freeswitch box
> Reply: freeswitch box ==> UAC
>
> This behaviour may be undesirable.
>
> So SIP proxy inserts itself to Path header and freeswitch recognizes it
> and add fs_path tag to contact.
>
> 2017-01-05 6:14 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>
> :
>
> Hello guys,
>
> I've been wondering where the "fs_path" parameter comes from? How is it
> built, etc? I've been seeing cases where when the client has no "fs_path",
> fs can't find the user and returns temp unavailable...
>
> Thanks all,
>
> David
>
>
>
> _________________________________________________________________________
>
>
> Professional FreeSWITCH Consulting Services:
>
>
> consulting at freeswitch.org
>
>
> http://www.freeswitchsolutions.com
>
>
>
>
>
> Official FreeSWITCH Sites
>
>
> http://www.freeswitch.org
>
>
> http://confluence.freeswitch.org
>
>
> http://www.cluecon.com
>
>
>
>
>
> FreeSWITCH-users mailing list
>
>
> FreeSWITCH-users at lists.freeswitch.org
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>
> http://www.freeswitch.org
>
>
>
>
> --
> ? ?????????,
> ????????? ???????
>
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vbvbrj at gmail.com  Thu Jan  5 19:49:04 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 5 Jan 2017 18:49:04 +0200
Subject: [Freeswitch-users] DTMF SIP Info or rfc2833 (rfc 4733)
Message-ID: <7339cecd-431a-ff69-49f1-0bb8e5e30c6d@gmail.com>

Hello.

What is better to use as standard for DTMF: SIP Info or rfc2833 ? In sip 
profile I can set <param name="liberal-dtmf" value="true"/> so, FS will 
accept both types of DTMF standard, while FS always send rfc2833. Why FS 
always send in rfc2833 and not the agreed with client like is the case 
of audio codecs? Is rfc2833 in FS a rfc4733 in the code, and is named 
rfc2833 in wiki?

-- 
Mimiko desu.


From anthony.minessale at gmail.com  Thu Jan  5 20:01:42 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Jan 2017 11:01:42 -0600
Subject: [Freeswitch-users] DTMF SIP Info or rfc2833 (rfc 4733)
In-Reply-To: <7339cecd-431a-ff69-49f1-0bb8e5e30c6d@gmail.com>
References: <7339cecd-431a-ff69-49f1-0bb8e5e30c6d@gmail.com>
Message-ID: <CAKbxfG-zv0VtNtmaNjEwcSLygBWXb_COOisAAPMpZKBHwASFLA@mail.gmail.com>

The 4733 is just an update to 2833 and its basically the same in terms of
our usage so you can consider the names interchangeable.
There is no way to negotiate SIP INFO DTMF so you can enable it as needed.
If you want to send INFO DTMF you can set the variable dtmf_type=info in
the outbound dial string.

INFO is the most reliable when used with TCP, 2833 is more reliable over
UDP.
INFO traverses the signaling patch where 2833 traverses the media so there
are considerations to make.  There is no best.


On Thu, Jan 5, 2017 at 10:49 AM, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
>
> What is better to use as standard for DTMF: SIP Info or rfc2833 ? In sip
> profile I can set <param name="liberal-dtmf" value="true"/> so, FS will
> accept both types of DTMF standard, while FS always send rfc2833. Why FS
> always send in rfc2833 and not the agreed with client like is the case
> of audio codecs? Is rfc2833 in FS a rfc4733 in the code, and is named
> rfc2833 in wiki?
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mike at jerris.com  Thu Jan  5 20:01:45 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Jan 2017 12:01:45 -0500
Subject: [Freeswitch-users] Fs_path
In-Reply-To: <CAFGRPVoBOJf5nAjinyy_GTmYm66+hLNo5XpNTwAhJUjX=HWVRg@mail.gmail.com>
References: <CAFGRPVpdD-KxFd4DNxkZqZKZyG51ONu2GXwp4vtqJLZ5G22a6Q@mail.gmail.com>
	<CANsY4DKiYOh=fyduUeASHSPO3X-OonrqiznV4hnkrtcvxJXZEg@mail.gmail.com>
	<CAEwTsAjV_d0Jm9LCZJ0g1sJPga74a-v5Z_wOsQ7dwD=2OaEYgQ@mail.gmail.com>
	<CAFGRPVoBOJf5nAjinyy_GTmYm66+hLNo5XpNTwAhJUjX=HWVRg@mail.gmail.com>
Message-ID: <818944FD-E893-41C5-8584-0DC092CE18A2@jerris.com>

fs_path has to do with WHERE it sends the INVITE.  Not found would happen if it can?t find the registration.


> On Jan 5, 2017, at 11:44 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> So if i understand correctly, even if fs_path is not there, there is no reason for fs not FINDING the registered user, i.e. Not even sending an invite request to said user and stating in the cli the user was not found?
> 
> On Thu, Jan 5, 2017 at 10:46 AM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> only partially correct here.  The main place fs_path is inserted is based on the nat detection settings you configure in freeswitch itself.  When we get a registration that's detected behind nat we write down the ip/port the request came from in fs_path before we store the registration, the. when calling that ref we will send the invite to that ip/port
> 
> On Thu, Jan 5, 2017 at 7:16 AM Vladyslav Zakhozhai <v.zakhozhai at gmail.com <mailto:v.zakhozhai at gmail.com>> wrote:
> Hi David,
> 
> there is no magick with fs_path. fs_path contains value from Path header of a request which comes from sip proxy, i.e. kamailio. It means that replies and SIP requests from freeswitch to UAC will traverse SIP proxy which should be on the path.
> 
> For example,
> Request: UAC ==> SIP proxy/load balancer ==> freeswitch box
> Reply: freeswitch box ==> SIP proxy/load balancer ==> UAC
> Request (INVITE): freeswitch box ==> SIP proxy/load balancer ==> UAC
> 
> Without path specified freeswitch will send direct messages to UAC:
> 
> Request: UAC ==> SIP proxy ==> freeswitch box
> Reply: freeswitch box ==> UAC
> 
> This behaviour may be undesirable.
> 
> So SIP proxy inserts itself to Path header and freeswitch recognizes it and add fs_path tag to contact.
> 
> 2017-01-05 6:14 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>>:
> Hello guys,
> 
> I've been wondering where the "fs_path" parameter comes from? How is it built, etc? I've been seeing cases where when the client has no "fs_path", fs can't find the user and returns temp unavailable...
> 
> Thanks all,
> 
> David
> 
> 
> 
> _________________________________________________________________________
> 
> 
> 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>
> 
> 
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> 
> 
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> ? ?????????,
> ????????? ???????
> 
> 
> 
> 
> 
> _________________________________________________________________________
> 
> Professional FreeSWITCH Consulting Services:
> 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> 
> http://www.freeswitchsolutions.com <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
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> 
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <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/>
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
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From vbvbrj at gmail.com  Thu Jan  5 20:11:42 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 5 Jan 2017 19:11:42 +0200
Subject: [Freeswitch-users] DTMF SIP Info or rfc2833 (rfc 4733)
In-Reply-To: <CAKbxfG-zv0VtNtmaNjEwcSLygBWXb_COOisAAPMpZKBHwASFLA@mail.gmail.com>
References: <7339cecd-431a-ff69-49f1-0bb8e5e30c6d@gmail.com>
	<CAKbxfG-zv0VtNtmaNjEwcSLygBWXb_COOisAAPMpZKBHwASFLA@mail.gmail.com>
Message-ID: <9935153c-6a9b-9220-b11a-aec2f75ab7e7@gmail.com>

On 05.01.2017 19:01, Anthony Minessale wrote:
> The 4733 is just an update to 2833 and its basically the same in terms
> of our usage so you can consider the names interchangeable.
> There is no way to negotiate SIP INFO DTMF so you can enable it as needed.
> If you want to send INFO DTMF you can set the variable dtmf_type=info in
> the outbound dial string.
>
> INFO is the most reliable when used with TCP, 2833 is more reliable over
> UDP.
> INFO traverses the signaling patch where 2833 traverses the media so
> there are considerations to make.  There is no best.

Oh, my project does not allow to flow rtp directly between clients, so 
SIP info is ok, also I use UDP so rfc2833 is needed. As I understand FS 
will catch SIP info and rfc2833 in same leg using that option.

Well, found some interesting info:

http://sip-implementors.cs.columbia.narkive.com/0VSY8gy9/is-rfc-2833-a-must-in-sending-dtmf

> Remember that SIP and RTP sometimes travels different paths. For many
> applications, it is important that DTMF is timed with the rest of the
> audio. Sending DTMF without timestamps and in a different signalling
> path totally removes the relation to the audio stream.
>
> My experience is that very few carriers now support SIP Info.

> Well, arguably not much relation to audio stream is really needed for
> DTMF applications. Relative events ordering, not absolute timestamps, is
> what important here and it is provided reliably by the CSeq in SIP INFO.
>
> I think technically SIP INFO delivery method is superior to RFC2833
> (DTMF is a signaling, not media after all, so that it should take
> signaling path), however, as it has been correctly mentioned here, there
> is no common standard and much weaker support across different
> implementations.

> The only upside I see is that it's easier to debug DTMF in the SIP
> channel than deep down in RTP.
> Otherwise, RFC2833 is the way to go.

I think info from the post should be reflected at 
https://freeswitch.org/confluence/display/FREESWITCH/DTMF

-- 
Mimiko desu.


From jose.lopes at itcenter.com.pt  Thu Jan  5 20:58:53 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Thu, 5 Jan 2017 17:58:53 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
Message-ID: <CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>

Hello Brian,

Thanks for your reply.

I tried the dialplan bellow with silence_stream://2000, and i have that
issue.
I tried with silence_stream://3000 and the audio cut off is greater.
Without the playback, there is no audio cut off, but FreeSwitch doesn't
send any rtp packets to verto client before the bridge.

There is any thing more that i can do?


<include>
  <context name="default">
    <extension name="call_debug" continue="true">
      <condition field="${call_debug}" expression="^true$" break="never">
        <action application="info"/>
      </condition>
    </extension>
    <extension name="itsp_send_call">
      <condition field="destination_number" expression="^.*$">
        <action application="answer"/>
        <action application="playback" data="silence_stream://2000"/>
        <action application="bridge"
data="{absolute_codec_string='PCMU'}sofia/gateway/1002/${destination_number}"/>
      </condition>
    </extension>
  </context>
</include>


Best Regards,
Jose Lopes


Os melhores cumprimentos / Best regards,

Jos? Lopes
Research and Development
Phone: +351 256 370 980
Email: jose.lopes at itcenter.com.pt

<http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/>
<http://www.itcenter.com.pt/>
www.itcenter.com.pt [image: ITCENTER Store]
<http://store.itcenter.com.pt/> [image:
ITCENTER Helpdesk] <http://support.itcenter.com.pt/otrs/customer.pl>
[image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image:
ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image:
ITCENTER Twitter] <https://twitter.com/ITCenter_PT>


2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:

> Prefix them with silence_stream://2000 or 3000 and it should go away.
>
> /b
>
>
> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> i have the same issue, i think its related to slow audio setup during the
>> call
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
>> to sip external voicemail
>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>> <jose.lopes at itcenter.com.pt>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 1/5/2017, 6:35:45 PM
>>
>> Hello Guys,
>>
>> I have audio cut off at the begin of the verto call to  FreeSwitch that
>> redirect to sip external voicemail (Access voicemail mailbox) .
>>
>> This happen when I use PCMU at verto codecs and sip codecs (if i use opus
>> at verto codecs, there is no issue, but this causes audio transcoding) .
>>
>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>> PSTN/ITSPs" from https://freeswitch.org/conflue
>> nce/display/FREESWITCH/mod_verto.
>> I notice if i remove the playback action, there is no issue. But I need
>> the playback action to send rtp packets to verto client.
>>
>> I simulate this using another FreeSwitch as external voicemail server and
>> I only listen "id followed by pound" from the initial message of voicemail
>> ("Please enter your id followed by pound").
>> The log of this call is at https://pastebin.freeswitch.org/view/507fa115
>>
>> What I can do to use PCMU at verto codecs and sip codecs on type of call?
>> Should i open a issue on FreeSwitch JIRA ?
>>
>>
>> Best regards,
>> Jose Lopes
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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From anthony.minessale at gmail.com  Thu Jan  5 21:13:47 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Jan 2017 12:13:47 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
Message-ID: <CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>

Try making the call with

fsctl debug_level 10

and observe the logs, answer followed by silence_stream should send audio
to the client.
Also try playing an audio file instead of silence stream to see if you hear
it.





On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
wrote:

> Hello Brian,
>
> Thanks for your reply.
>
> I tried the dialplan bellow with silence_stream://2000, and i have that
> issue.
> I tried with silence_stream://3000 and the audio cut off is greater.
> Without the playback, there is no audio cut off, but FreeSwitch doesn't
> send any rtp packets to verto client before the bridge.
>
> There is any thing more that i can do?
>
>
> <include>
>   <context name="default">
>     <extension name="call_debug" continue="true">
>       <condition field="${call_debug}" expression="^true$" break="never">
>         <action application="info"/>
>       </condition>
>     </extension>
>     <extension name="itsp_send_call">
>       <condition field="destination_number" expression="^.*$">
>         <action application="answer"/>
>         <action application="playback" data="silence_stream://2000"/>
>         <action application="bridge" data="{absolute_codec_string='
> PCMU'}sofia/gateway/1002/${destination_number}"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
>
> Best Regards,
> Jose Lopes
>
>
> Os melhores cumprimentos / Best regards,
>
> Jos? Lopes
> Research and Development
> Phone: +351 256 370 980
> Email: jose.lopes at itcenter.com.pt
>
> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/>
> <http://www.itcenter.com.pt/>
> www.itcenter.com.pt [image: ITCENTER Store]
> <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk]
> <http://support.itcenter.com.pt/otrs/customer.pl>
> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image:
> ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image:
> ITCENTER Twitter] <https://twitter.com/ITCenter_PT>
>
>
> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>
>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>
>> /b
>>
>>
>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>
>>> hi,
>>>
>>> i have the same issue, i think its related to slow audio setup during
>>> the call
>>>
>>>
>>> Regards,
>>> Bipin
>>>
>>>
>>> ------------------------------
>>> -------- Original Message --------
>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
>>> to sip external voicemail
>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>> <jose.lopes at itcenter.com.pt>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: 1/5/2017, 6:35:45 PM
>>>
>>> Hello Guys,
>>>
>>> I have audio cut off at the begin of the verto call to  FreeSwitch that
>>> redirect to sip external voicemail (Access voicemail mailbox) .
>>>
>>> This happen when I use PCMU at verto codecs and sip codecs (if i use
>>> opus at verto codecs, there is no issue, but this causes audio transcoding)
>>> .
>>>
>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>> nce/display/FREESWITCH/mod_verto.
>>> I notice if i remove the playback action, there is no issue. But I need
>>> the playback action to send rtp packets to verto client.
>>>
>>> I simulate this using another FreeSwitch as external voicemail server
>>> and I only listen "id followed by pound" from the initial message of
>>> voicemail ("Please enter your id followed by pound").
>>> The log of this call is at https://pastebin.freeswitch.org/view/507fa115
>>>
>>> What I can do to use PCMU at verto codecs and sip codecs on type of call?
>>> Should i open a issue on FreeSwitch JIRA ?
>>>
>>>
>>> Best regards,
>>> Jose Lopes
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>> https://www.gofundme.com/freeswitch_ubuntu
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *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
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From anthony.minessale at gmail.com  Thu Jan  5 21:14:30 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Jan 2017 12:14:30 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
Message-ID: <CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>

Also make sure you don't have answer_delay set in your vars.xml


On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Try making the call with
>
> fsctl debug_level 10
>
> and observe the logs, answer followed by silence_stream should send audio
> to the client.
> Also try playing an audio file instead of silence stream to see if you
> hear it.
>
>
>
>
>
> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
> wrote:
>
>> Hello Brian,
>>
>> Thanks for your reply.
>>
>> I tried the dialplan bellow with silence_stream://2000, and i have that
>> issue.
>> I tried with silence_stream://3000 and the audio cut off is greater.
>> Without the playback, there is no audio cut off, but FreeSwitch doesn't
>> send any rtp packets to verto client before the bridge.
>>
>> There is any thing more that i can do?
>>
>>
>> <include>
>>   <context name="default">
>>     <extension name="call_debug" continue="true">
>>       <condition field="${call_debug}" expression="^true$" break="never">
>>         <action application="info"/>
>>       </condition>
>>     </extension>
>>     <extension name="itsp_send_call">
>>       <condition field="destination_number" expression="^.*$">
>>         <action application="answer"/>
>>         <action application="playback" data="silence_stream://2000"/>
>>         <action application="bridge" data="{absolute_codec_string='
>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>       </condition>
>>     </extension>
>>   </context>
>> </include>
>>
>>
>> Best Regards,
>> Jose Lopes
>>
>>
>> Os melhores cumprimentos / Best regards,
>>
>> Jos? Lopes
>> Research and Development
>> Phone: +351 256 370 980
>> Email: jose.lopes at itcenter.com.pt
>>
>> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/>
>> <http://www.itcenter.com.pt/>
>> www.itcenter.com.pt [image: ITCENTER Store]
>> <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk]
>> <http://support.itcenter.com.pt/otrs/customer.pl>
>> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image:
>> ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image:
>> ITCENTER Twitter] <https://twitter.com/ITCenter_PT>
>>
>>
>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>
>>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>>
>>> /b
>>>
>>>
>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>>
>>>> hi,
>>>>
>>>> i have the same issue, i think its related to slow audio setup during
>>>> the call
>>>>
>>>>
>>>> Regards,
>>>> Bipin
>>>>
>>>>
>>>> ------------------------------
>>>> -------- Original Message --------
>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto
>>>> call to sip external voicemail
>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>> <jose.lopes at itcenter.com.pt>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> <freeswitch-users at lists.freeswitch.org>
>>>> Date: 1/5/2017, 6:35:45 PM
>>>>
>>>> Hello Guys,
>>>>
>>>> I have audio cut off at the begin of the verto call to  FreeSwitch that
>>>> redirect to sip external voicemail (Access voicemail mailbox) .
>>>>
>>>> This happen when I use PCMU at verto codecs and sip codecs (if i use
>>>> opus at verto codecs, there is no issue, but this causes audio transcoding)
>>>> .
>>>>
>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>>> nce/display/FREESWITCH/mod_verto.
>>>> I notice if i remove the playback action, there is no issue. But I need
>>>> the playback action to send rtp packets to verto client.
>>>>
>>>> I simulate this using another FreeSwitch as external voicemail server
>>>> and I only listen "id followed by pound" from the initial message of
>>>> voicemail ("Please enter your id followed by pound").
>>>> The log of this call is at https://pastebin.freeswitch
>>>> .org/view/507fa115
>>>>
>>>> What I can do to use PCMU at verto codecs and sip codecs on type of
>>>> call?
>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>
>>>>
>>>> Best regards,
>>>> Jose Lopes
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>> https://www.gofundme.com/freeswitch_ubuntu
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>> *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
>>
>
>
>
> --
> 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 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From info at itwrx.org  Thu Jan  5 23:55:02 2017
From: info at itwrx.org (ITwrx.org)
Date: Thu, 5 Jan 2017 14:55:02 -0600
Subject: [Freeswitch-users] tls with letsencrypt
Message-ID: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>

hi,

i'm trying to use a letsencrypt generated cert with freeswitch but am
not sure how to proceed. I've read the old and new wiki posts concerning
tls but they don't seem to cover my exact scenario. It seems to me that
freeswitch is looking into the configured "tls-cert-dir" for the
hardcoded filename tls.pem and is expecting that a self generated ca has
signed it. i have placed the fullchain.pem in that directory (generated
with certbot) and have renamed it tls.pem but i guess it's not finding
the CA sig it expects(?) as i'm getting:

tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup
failed (error:00000005:lib(0):func(0):DH lib)

when trying to connect with csipsimple from phone. I would like to avoid
generating client certs signed by a custom CA where users have to copy
the client cert and ca cert to their device as it adds complexity and
problems. Is there a workaround or suggested method for using a
letsencrypt cert with freeswitch so that clients like csipsimple can
just validate against their built-in CA store?

thanks in advance,
ITwrx

-- 
Information Technology Works
https://ITwrx.org
@ITwrxorg



From brian at freeswitch.org  Fri Jan  6 01:36:17 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Jan 2017 16:36:17 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
Message-ID: <CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>

How did you format the cert? and in what files did you put them in? and are
your permissions correct on those files?

On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org> wrote:

> hi,
>
> i'm trying to use a letsencrypt generated cert with freeswitch but am
> not sure how to proceed. I've read the old and new wiki posts concerning
> tls but they don't seem to cover my exact scenario. It seems to me that
> freeswitch is looking into the configured "tls-cert-dir" for the
> hardcoded filename tls.pem and is expecting that a self generated ca has
> signed it. i have placed the fullchain.pem in that directory (generated
> with certbot) and have renamed it tls.pem but i guess it's not finding
> the CA sig it expects(?) as i'm getting:
>
> tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup
> failed (error:00000005:lib(0):func(0):DH lib)
>
> when trying to connect with csipsimple from phone. I would like to avoid
> generating client certs signed by a custom CA where users have to copy
> the client cert and ca cert to their device as it adds complexity and
> problems. Is there a workaround or suggested method for using a
> letsencrypt cert with freeswitch so that clients like csipsimple can
> just validate against their built-in CA store?
>
> thanks in advance,
> ITwrx
>
> --
> Information Technology Works
> https://ITwrx.org
> @ITwrxorg
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From info at itwrx.org  Fri Jan  6 01:53:11 2017
From: info at itwrx.org (ITwrx.org)
Date: Thu, 5 Jan 2017 16:53:11 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
	<CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
Message-ID: <ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>

i just copied the pem formatted cert that certbot generated to
/etc/freeswitch/tls and named it tls.pem. it's freeswitch:freeswitch 660
for perms. freeswitch seems capable of reading it, as the tls enabled
profile starts up. i only get an error in fs_cli when the csipsimple
client tries to connect using tls.

thanks

On 01/05/2017 04:36 PM, Brian West wrote:
> How did you format the cert? and in what files did you put them in?
> and are your permissions correct on those files?
>
> On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org
> <mailto:info at itwrx.org>> wrote:
>
>     hi,
>
>     i'm trying to use a letsencrypt generated cert with freeswitch but am
>     not sure how to proceed. I've read the old and new wiki posts
>     concerning
>     tls but they don't seem to cover my exact scenario. It seems to me
>     that
>     freeswitch is looking into the configured "tls-cert-dir" for the
>     hardcoded filename tls.pem and is expecting that a self generated
>     ca has
>     signed it. i have placed the fullchain.pem in that directory
>     (generated
>     with certbot) and have renamed it tls.pem but i guess it's not finding
>     the CA sig it expects(?) as i'm getting:
>
>     tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup
>     failed (error:00000005:lib(0):func(0):DH lib)
>
>     when trying to connect with csipsimple from phone. I would like to
>     avoid
>     generating client certs signed by a custom CA where users have to copy
>     the client cert and ca cert to their device as it adds complexity and
>     problems. Is there a workaround or suggested method for using a
>     letsencrypt cert with freeswitch so that clients like csipsimple can
>     just validate against their built-in CA store?
>
>     thanks in advance,
>     ITwrx
>
>     --
>     Information Technology Works
>     https://ITwrx.org
>     @ITwrxorg
>
>
>     _________________________________________________________________________
>     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://confluence.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
>     <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
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
> http://www.freeswitchbook.com 
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! |
> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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


-- 
Information Technology Works
https://ITwrx.org
@ITwrxorg

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From aaron at tfdm.us  Fri Jan  6 02:13:24 2017
From: aaron at tfdm.us (Aaron S)
Date: Thu, 05 Jan 2017 15:13:24 -0800
Subject: [Freeswitch-users] TLS/SRTP performance numbers and bottlenecks?
Message-ID: <f1dcc628197d731bba321f90aba93cc7@tfdm.us>

I have an application that requires a large number (500-1000 to start) 
of concurrent g.711u streams/calls as seen in the Hybrid Encryption 
section at https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS. 
  Parallelizing this is something that's certainly possible (and would be 
used with two identical systems for redundancy's sake to start), but I'm 
wondering if two systems on bare metal would be sufficient, or more is 
required.

Looking for stats via google, I'm not seeing much out there.  I'm not 
sure where the bottlenecks of this system are.  I'm not doing any 
transcoding, just stripping out the encryption on one leg of the call.

Any tips, real-world numbers/experience, or what have you would be very 
appreciated.

Thanks,

-Aaron


From brian at freeswitch.org  Fri Jan  6 03:43:47 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Jan 2017 18:43:47 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
	<CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
	<ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>
Message-ID: <CAEJMVkDN8khPRvkx1PrXzyi9+zf2Mkbvd6dV3xq2mkPLyS9d4w@mail.gmail.com>

There is a lot more to it than that, what files are in that tls folder?

On Thu, Jan 5, 2017 at 4:53 PM, ITwrx.org <info at itwrx.org> wrote:

> i just copied the pem formatted cert that certbot generated to
> /etc/freeswitch/tls and named it tls.pem. it's freeswitch:freeswitch 660
> for perms. freeswitch seems capable of reading it, as the tls enabled
> profile starts up. i only get an error in fs_cli when the csipsimple client
> tries to connect using tls.
>
> thanks
>
>
> On 01/05/2017 04:36 PM, Brian West wrote:
>
> How did you format the cert? and in what files did you put them in? and
> are your permissions correct on those files?
>
> On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org> wrote:
>
>> hi,
>>
>> i'm trying to use a letsencrypt generated cert with freeswitch but am
>> not sure how to proceed. I've read the old and new wiki posts concerning
>> tls but they don't seem to cover my exact scenario. It seems to me that
>> freeswitch is looking into the configured "tls-cert-dir" for the
>> hardcoded filename tls.pem and is expecting that a self generated ca has
>> signed it. i have placed the fullchain.pem in that directory (generated
>> with certbot) and have renamed it tls.pem but i guess it's not finding
>> the CA sig it expects(?) as i'm getting:
>>
>> tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup
>> failed (error:00000005:lib(0):func(0):DH lib)
>>
>> when trying to connect with csipsimple from phone. I would like to avoid
>> generating client certs signed by a custom CA where users have to copy
>> the client cert and ca cert to their device as it adds complexity and
>> problems. Is there a workaround or suggested method for using a
>> letsencrypt cert with freeswitch so that clients like csipsimple can
>> just validate against their built-in CA store?
>>
>> thanks in advance,
>> ITwrx
>>
>> --
>> Information Technology Works
>> https://ITwrx.org
>> @ITwrxorg
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Information Technology Workshttps://ITwrx.org
> @ITwrxorg
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From info at itwrx.org  Fri Jan  6 05:24:29 2017
From: info at itwrx.org (ITwrx.org)
Date: Thu, 5 Jan 2017 20:24:29 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <CAEJMVkDN8khPRvkx1PrXzyi9+zf2Mkbvd6dV3xq2mkPLyS9d4w@mail.gmail.com>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
	<CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
	<ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>
	<CAEJMVkDN8khPRvkx1PrXzyi9+zf2Mkbvd6dV3xq2mkPLyS9d4w@mail.gmail.com>
Message-ID: <35824367-2e60-e2b4-61d8-ee2dc6428c76@itwrx.org>

dtls-srtp.pem,
tls.pem(the "stand in" i previously described),
and the original (could be from my old server where i set up tls
following the freeswitch wiki) tls.pem which has been renamed to
tls.pem.orig.

On 01/05/2017 06:43 PM, Brian West wrote:
> There is a lot more to it than that, what files are in that tls folder?
>
> On Thu, Jan 5, 2017 at 4:53 PM, ITwrx.org <info at itwrx.org
> <mailto:info at itwrx.org>> wrote:
>
>     i just copied the pem formatted cert that certbot generated to
>     /etc/freeswitch/tls and named it tls.pem. it's
>     freeswitch:freeswitch 660 for perms. freeswitch seems capable of
>     reading it, as the tls enabled profile starts up. i only get an
>     error in fs_cli when the csipsimple client tries to connect using tls.
>
>     thanks
>
>
>     On 01/05/2017 04:36 PM, Brian West wrote:
>>     How did you format the cert? and in what files did you put them
>>     in? and are your permissions correct on those files?
>>
>>     On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org
>>     <mailto:info at itwrx.org>> wrote:
>>
>>         hi,
>>
>>         i'm trying to use a letsencrypt generated cert with
>>         freeswitch but am
>>         not sure how to proceed. I've read the old and new wiki posts
>>         concerning
>>         tls but they don't seem to cover my exact scenario. It seems
>>         to me that
>>         freeswitch is looking into the configured "tls-cert-dir" for the
>>         hardcoded filename tls.pem and is expecting that a self
>>         generated ca has
>>         signed it. i have placed the fullchain.pem in that directory
>>         (generated
>>         with certbot) and have renamed it tls.pem but i guess it's
>>         not finding
>>         the CA sig it expects(?) as i'm getting:
>>
>>         tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0):
>>         TLS setup
>>         failed (error:00000005:lib(0):func(0):DH lib)
>>
>>         when trying to connect with csipsimple from phone. I would
>>         like to avoid
>>         generating client certs signed by a custom CA where users
>>         have to copy
>>         the client cert and ca cert to their device as it adds
>>         complexity and
>>         problems. Is there a workaround or suggested method for using a
>>         letsencrypt cert with freeswitch so that clients like
>>         csipsimple can
>>         just validate against their built-in CA store?
>>
>>         thanks in advance,
>>         ITwrx
>>
>>         --
>>         Information Technology Works
>>         https://ITwrx.org
>>         @ITwrxorg
>>
>>
>>         _________________________________________________________________________
>>         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://confluence.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
>>         <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
>>
>>
>>
>>
>>     -- 
>>
>>     */Brian West/*
>>     brian at freeswitch.org <mailto:brian at freeswitch.org>
>>
>>
>>     */Twitter: @FreeSWITCH , @briankwest/*
>>     http://www.freeswitchbook.com 
>>     http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com>
>>     https://www.gofundme.com/freeswitch_ubuntu
>>     <https://www.gofundme.com/freeswitch_ubuntu>
>>
>>     Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>     Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>>     *T:*+19184209001 <tel:%28918%29%20420-9001> | *F:*+19184209002
>>     <tel:%28918%29%20420-9002> | *M:*+1918424WEST (9378)
>>     *Skype:*briankwest
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services: 
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.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
>>     <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
>
>     -- 
>     Information Technology Works
>     https://ITwrx.org
>     @ITwrxorg
>
>     _________________________________________________________________________
>     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://confluence.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
>     <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 
>
> -- 
>
> */Brian West/* brian at freeswitch.org <mailto:brian at freeswitch.org>
>
> */Twitter: @FreeSWITCH , @briankwest/* http://www.freeswitchbook.com 
> http://www.freeswitchcookbook.comhttps://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! |
> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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

-- 
Information Technology Works
https://ITwrx.org
@ITwrxorg

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From devang.nathwani31589 at gmail.com  Fri Jan  6 11:43:09 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Fri, 6 Jan 2017 14:13:09 +0530
Subject: [Freeswitch-users] Replace disposition cause
Message-ID: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>

Hello,

I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with '503
SERVICE_UNAVAILABLE';

tried;
<action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
<action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
before bridge application
But its not working
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From v.zakhozhai at gmail.com  Fri Jan  6 12:07:00 2017
From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai)
Date: Fri, 6 Jan 2017 11:07:00 +0200
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
Message-ID: <CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>

Devang, I'm no sure but this config should not work.
I think that you need to handle failure_causes after the bridge and use
respond command to respond with cause you need.

Maybe I am wrong but this approach will do the work.

2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.com>:

> Hello,
>
> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
> '503 SERVICE_UNAVAILABLE';
>
> tried;
> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
> before bridge application
> But its not working
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
? ?????????,
????????? ???????
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From gascagonzalo at gmail.com  Fri Jan  6 12:48:47 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Fri, 6 Jan 2017 01:48:47 -0800
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
Message-ID: <CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>

Can you share more details about the call flow and debug if possible? Are
you planning to replace an incoming disconnect code or you want to send an
specific disconnect code?

<action application="set" data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>

*continue_on_fail* "Controls what happens when the called party can not be
reached (busy/offline). If "true" the dialplan continues to be processed.
If "false" the dialplan will stop processing. Can contain the return
messages that will continue on fail also."

<action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>

*failure_causes*  Controls which failure causes will be considered as a
failure to the bridge(s). This will change the values for which
continue_on_fail will fail by default unless continue_on_fail is set to
true.

Depending of your flow you can use:

<action application="hangup" data="SERVICE_UNAVAILABLE"/>


On Fri, Jan 6, 2017 at 1:07 AM, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
wrote:

> Devang, I'm no sure but this config should not work.
> I think that you need to handle failure_causes after the bridge and use
> respond command to respond with cause you need.
>
> Maybe I am wrong but this approach will do the work.
>
> 2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.com
> >:
>
>> Hello,
>>
>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>> '503 SERVICE_UNAVAILABLE';
>>
>> tried;
>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>> before bridge application
>> But its not working
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ? ?????????,
> ????????? ???????
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From devang.nathwani31589 at gmail.com  Fri Jan  6 13:01:53 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Fri, 6 Jan 2017 15:31:53 +0530
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
Message-ID: <CAGOm0PKb3B=FbrYTyDkUeANTzuDB3t8ewLb1JCOSFDZ+5fNY1A@mail.gmail.com>

tried adding,
<action application="set\" data="failure_causes=503">
<action application="respond" data="503 Service Unavailable">

after bridge application
but not working.

On Fri, Jan 6, 2017 at 2:37 PM, Vladyslav Zakhozhai <v.zakhozhai at gmail.com>
wrote:

> Devang, I'm no sure but this config should not work.
> I think that you need to handle failure_causes after the bridge and use
> respond command to respond with cause you need.
>
> Maybe I am wrong but this approach will do the work.
>
> 2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.com
> >:
>
>> Hello,
>>
>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>> '503 SERVICE_UNAVAILABLE';
>>
>> tried;
>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>> before bridge application
>> But its not working
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ? ?????????,
> ????????? ???????
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gascagonzalo at gmail.com  Fri Jan  6 13:03:51 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Fri, 6 Jan 2017 02:03:51 -0800
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAGOm0PKb3B=FbrYTyDkUeANTzuDB3t8ewLb1JCOSFDZ+5fNY1A@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
	<CAGOm0PKb3B=FbrYTyDkUeANTzuDB3t8ewLb1JCOSFDZ+5fNY1A@mail.gmail.com>
Message-ID: <CAL+DTgLKDYaenwvUYaCqqobz+HMtEHzNtSAY81zdKB-+MNUO5A@mail.gmail.com>

Did you read the previous email at all??


On Fri, Jan 6, 2017 at 2:01 AM, devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> tried adding,
> <action application="set\" data="failure_causes=503">
> <action application="respond" data="503 Service Unavailable">
>
> after bridge application
> but not working.
>
> On Fri, Jan 6, 2017 at 2:37 PM, Vladyslav Zakhozhai <v.zakhozhai at gmail.com
> > wrote:
>
>> Devang, I'm no sure but this config should not work.
>> I think that you need to handle failure_causes after the bridge and use
>> respond command to respond with cause you need.
>>
>> Maybe I am wrong but this approach will do the work.
>>
>> 2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.co
>> m>:
>>
>>> Hello,
>>>
>>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>>> '503 SERVICE_UNAVAILABLE';
>>>
>>> tried;
>>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>> before bridge application
>>> But its not working
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ? ?????????,
>> ????????? ???????
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From devang.nathwani31589 at gmail.com  Fri Jan  6 13:08:12 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Fri, 6 Jan 2017 15:38:12 +0530
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
	<CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>
Message-ID: <CAGOm0P+Ld6kDthxmYbcnPmPjnXfQbNQtdnwqoMreZDTeUF9V1A@mail.gmail.com>

Hello,

no i haven't read when i drafted the previous mail!

The scenario is something, gateway is completely down suppose because of
hardware failure or electricity issue.

so freeswitch trying to send caller's request to gateway, as gateway is
completely down freeswitch continuously trying to send 'INVITE' request to
gateway, getting nothing in response so freeswitch sending 480
TEMPORARILY_UNAVAILABLE to caller but i want to change with '503
SERVICE_UNAVAILABLE'

and fscli showing, [CS_EXECUTE] [ALLOTTED_TIMEOUT] and [CS_CONSUME_MEDIA]
[ORIGINATOR_CANCEL]

On Fri, Jan 6, 2017 at 3:18 PM, Gonzalo Gasca Meza <gascagonzalo at gmail.com>
wrote:

>
> Can you share more details about the call flow and debug if possible? Are
> you planning to replace an incoming disconnect code or you want to send an
> specific disconnect code?
>
> <action application="set" data="continue_on_fail=
> TEMPORARILY_UNAVAILABLE"/>
>
> *continue_on_fail* "Controls what happens when the called party can not
> be reached (busy/offline). If "true" the dialplan continues to be
> processed. If "false" the dialplan will stop processing. Can contain the
> return messages that will continue on fail also."
>
> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>
> *failure_causes*  Controls which failure causes will be considered as a
> failure to the bridge(s). This will change the values for which
> continue_on_fail will fail by default unless continue_on_fail is set to
> true.
>
> Depending of your flow you can use:
>
> <action application="hangup" data="SERVICE_UNAVAILABLE"/>
>
>
> On Fri, Jan 6, 2017 at 1:07 AM, Vladyslav Zakhozhai <v.zakhozhai at gmail.com
> > wrote:
>
>> Devang, I'm no sure but this config should not work.
>> I think that you need to handle failure_causes after the bridge and use
>> respond command to respond with cause you need.
>>
>> Maybe I am wrong but this approach will do the work.
>>
>> 2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.co
>> m>:
>>
>>> Hello,
>>>
>>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>>> '503 SERVICE_UNAVAILABLE';
>>>
>>> tried;
>>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>> before bridge application
>>> But its not working
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ? ?????????,
>> ????????? ???????
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From devang.nathwani31589 at gmail.com  Fri Jan  6 13:15:39 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Fri, 6 Jan 2017 15:45:39 +0530
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAGOm0P+Ld6kDthxmYbcnPmPjnXfQbNQtdnwqoMreZDTeUF9V1A@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
	<CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>
	<CAGOm0P+Ld6kDthxmYbcnPmPjnXfQbNQtdnwqoMreZDTeUF9V1A@mail.gmail.com>
Message-ID: <CAGOm0PKeJSJpqKvcQVYY_C0WNneofdiUBuiVbvJRzekrZdqFdQ@mail.gmail.com>

yeah, i am getting reference from here,
https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables

On Fri, Jan 6, 2017 at 3:38 PM, devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> Hello,
>
> no i haven't read when i drafted the previous mail!
>
> The scenario is something, gateway is completely down suppose because of
> hardware failure or electricity issue.
>
> so freeswitch trying to send caller's request to gateway, as gateway is
> completely down freeswitch continuously trying to send 'INVITE' request to
> gateway, getting nothing in response so freeswitch sending 480
> TEMPORARILY_UNAVAILABLE to caller but i want to change with '503
> SERVICE_UNAVAILABLE'
>
> and fscli showing, [CS_EXECUTE] [ALLOTTED_TIMEOUT] and [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
>
> On Fri, Jan 6, 2017 at 3:18 PM, Gonzalo Gasca Meza <gascagonzalo at gmail.com
> > wrote:
>
>>
>> Can you share more details about the call flow and debug if possible? Are
>> you planning to replace an incoming disconnect code or you want to send an
>> specific disconnect code?
>>
>> <action application="set" data="continue_on_fail=TEMPORA
>> RILY_UNAVAILABLE"/>
>>
>> *continue_on_fail* "Controls what happens when the called party can not
>> be reached (busy/offline). If "true" the dialplan continues to be
>> processed. If "false" the dialplan will stop processing. Can contain the
>> return messages that will continue on fail also."
>>
>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>
>> *failure_causes*  Controls which failure causes will be considered as a
>> failure to the bridge(s). This will change the values for which
>> continue_on_fail will fail by default unless continue_on_fail is set to
>> true.
>>
>> Depending of your flow you can use:
>>
>> <action application="hangup" data="SERVICE_UNAVAILABLE"/>
>>
>>
>> On Fri, Jan 6, 2017 at 1:07 AM, Vladyslav Zakhozhai <
>> v.zakhozhai at gmail.com> wrote:
>>
>>> Devang, I'm no sure but this config should not work.
>>> I think that you need to handle failure_causes after the bridge and use
>>> respond command to respond with cause you need.
>>>
>>> Maybe I am wrong but this approach will do the work.
>>>
>>> 2017-01-06 10:43 GMT+02:00 devang nathwani <
>>> devang.nathwani31589 at gmail.com>:
>>>
>>>> Hello,
>>>>
>>>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>>>> '503 SERVICE_UNAVAILABLE';
>>>>
>>>> tried;
>>>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>>>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>>> before bridge application
>>>> But its not working
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ? ?????????,
>>> ????????? ???????
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mirkobrankovic at gmail.com  Fri Jan  6 13:58:48 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Fri, 6 Jan 2017 11:58:48 +0100
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <35824367-2e60-e2b4-61d8-ee2dc6428c76@itwrx.org>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
	<CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
	<ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>
	<CAEJMVkDN8khPRvkx1PrXzyi9+zf2Mkbvd6dV3xq2mkPLyS9d4w@mail.gmail.com>
	<35824367-2e60-e2b4-61d8-ee2dc6428c76@itwrx.org>
Message-ID: <CAND18T2ZiBF2PYWP8+imvpA9a3p=srtfaBSmY-KKqCx9Ob-yqA@mail.gmail.com>

Hey,
All I had to do to get it work is to place cert and key in one pem file for
FS, so like:
cat /etc/letsencrypt/live/${domain}/cert.pem
/etc/letsencrypt/live/${domain}/privkey.pem >
/usr/local/freeswitch/certs/wss.pem

On Fri, Jan 6, 2017 at 3:24 AM, ITwrx.org <info at itwrx.org> wrote:

> dtls-srtp.pem,
> tls.pem(the "stand in" i previously described),
> and the original (could be from my old server where i set up tls following
> the freeswitch wiki) tls.pem which has been renamed to tls.pem.orig.
>
>
> On 01/05/2017 06:43 PM, Brian West wrote:
>
> There is a lot more to it than that, what files are in that tls folder?
>
> On Thu, Jan 5, 2017 at 4:53 PM, ITwrx.org <info at itwrx.org> wrote:
>
>> i just copied the pem formatted cert that certbot generated to
>> /etc/freeswitch/tls and named it tls.pem. it's freeswitch:freeswitch 660
>> for perms. freeswitch seems capable of reading it, as the tls enabled
>> profile starts up. i only get an error in fs_cli when the csipsimple client
>> tries to connect using tls.
>>
>> thanks
>>
>>
>> On 01/05/2017 04:36 PM, Brian West wrote:
>>
>> How did you format the cert? and in what files did you put them in? and
>> are your permissions correct on those files?
>>
>> On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org> wrote:
>>
>>> hi,
>>>
>>> i'm trying to use a letsencrypt generated cert with freeswitch but am
>>> not sure how to proceed. I've read the old and new wiki posts concerning
>>> tls but they don't seem to cover my exact scenario. It seems to me that
>>> freeswitch is looking into the configured "tls-cert-dir" for the
>>> hardcoded filename tls.pem and is expecting that a self generated ca has
>>> signed it. i have placed the fullchain.pem in that directory (generated
>>> with certbot) and have renamed it tls.pem but i guess it's not finding
>>> the CA sig it expects(?) as i'm getting:
>>>
>>> tport_tls.c:1044 tls_connect() tls_connect(0x373c000e8d0): TLS setup
>>> failed (error:00000005:lib(0):func(0):DH lib)
>>>
>>> when trying to connect with csipsimple from phone. I would like to avoid
>>> generating client certs signed by a custom CA where users have to copy
>>> the client cert and ca cert to their device as it adds complexity and
>>> problems. Is there a workaround or suggested method for using a
>>> letsencrypt cert with freeswitch so that clients like csipsimple can
>>> just validate against their built-in CA store?
>>>
>>> thanks in advance,
>>> ITwrx
>>>
>>> --
>>> Information Technology Works
>>> https://ITwrx.org
>>> @ITwrxorg
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> https://www.gofundme.com/freeswitch_ubuntu
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <%28918%29%20420-9001> | *F:*+19184209002
>> <%28918%29%20420-9002> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>> --
>> Information Technology Workshttps://ITwrx.org
>> @ITwrxorg
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
>> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
>> http://www.freeswitch.org http://confluence.freeswitch.org
>> http://www.cluecon.com FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/ma
>> ilman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.frees
>> witch.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 https://www.gofundme.com/
> freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
> <+1%20918-420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
> --
> Information Technology Workshttps://ITwrx.org
> @ITwrxorg
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Mirko
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From jose.lopes at itcenter.com.pt  Fri Jan  6 14:08:01 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Fri, 6 Jan 2017 11:08:01 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
Message-ID: <CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>

Hello Anthony,

Thanks for your reply.

I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
with ~2 seconds) instead of silence_stream.
When i make the call from verto client, i ear the audio file, then no audio
for ~2/3 seconds and then i ear "id followed by pound" (audio cut off from
voicemail initial message "Please enter your id followed by pound").

I checked if i have the variable answer_delay and i don't have it.

The log of this call is at https://pastebin.freeswitch.org/view/e130e172 .

There is any thing more that i can do?


Best Regards,
Jose Lopes

2017-01-05 18:14 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Also make sure you don't have answer_delay set in your vars.xml
>
>
> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Try making the call with
>>
>> fsctl debug_level 10
>>
>> and observe the logs, answer followed by silence_stream should send audio
>> to the client.
>> Also try playing an audio file instead of silence stream to see if you
>> hear it.
>>
>>
>>
>>
>>
>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>> wrote:
>>
>>> Hello Brian,
>>>
>>> Thanks for your reply.
>>>
>>> I tried the dialplan bellow with silence_stream://2000, and i have that
>>> issue.
>>> I tried with silence_stream://3000 and the audio cut off is greater.
>>> Without the playback, there is no audio cut off, but FreeSwitch doesn't
>>> send any rtp packets to verto client before the bridge.
>>>
>>> There is any thing more that i can do?
>>>
>>>
>>> <include>
>>>   <context name="default">
>>>     <extension name="call_debug" continue="true">
>>>       <condition field="${call_debug}" expression="^true$" break="never">
>>>         <action application="info"/>
>>>       </condition>
>>>     </extension>
>>>     <extension name="itsp_send_call">
>>>       <condition field="destination_number" expression="^.*$">
>>>         <action application="answer"/>
>>>         <action application="playback" data="silence_stream://2000"/>
>>>         <action application="bridge" data="{absolute_codec_string='
>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>       </condition>
>>>     </extension>
>>>   </context>
>>> </include>
>>>
>>>
>>> Best Regards,
>>> Jose Lopes
>>>
>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>
>>>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>>>
>>>> /b
>>>>
>>>>
>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>>>
>>>>> hi,
>>>>>
>>>>> i have the same issue, i think its related to slow audio setup during
>>>>> the call
>>>>>
>>>>>
>>>>> Regards,
>>>>> Bipin
>>>>>
>>>>>
>>>>> ------------------------------
>>>>> -------- Original Message --------
>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto
>>>>> call to sip external voicemail
>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>> <jose.lopes at itcenter.com.pt>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>
>>>>> Hello Guys,
>>>>>
>>>>> I have audio cut off at the begin of the verto call to  FreeSwitch
>>>>> that redirect to sip external voicemail (Access voicemail mailbox) .
>>>>>
>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i use
>>>>> opus at verto codecs, there is no issue, but this causes audio transcoding)
>>>>> .
>>>>>
>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>> nce/display/FREESWITCH/mod_verto.
>>>>> I notice if i remove the playback action, there is no issue. But I
>>>>> need the playback action to send rtp packets to verto client.
>>>>>
>>>>> I simulate this using another FreeSwitch as external voicemail server
>>>>> and I only listen "id followed by pound" from the initial message of
>>>>> voicemail ("Please enter your id followed by pound").
>>>>> The log of this call is at https://pastebin.freeswitch
>>>>> .org/view/507fa115
>>>>>
>>>>> What I can do to use PCMU at verto codecs and sip codecs on type of
>>>>> call?
>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>
>>>>>
>>>>> Best regards,
>>>>> Jose Lopes
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>
>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>> *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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 <(919)%20386-9900>
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>
>
>
> --
> 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 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From w8hdkim at gmail.com  Fri Jan  6 15:23:37 2017
From: w8hdkim at gmail.com (Kim Culhan)
Date: Fri, 6 Jan 2017 07:23:37 -0500
Subject: [Freeswitch-users] tls with letsencrypt
Message-ID: <CAKZxVQXG_dhMmEQ2+chwMxBex2i86a=xMS9wxXEB6KDYPizfvw@mail.gmail.com>

On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org
> <https://webmail.kim.net/src/compose.php?send_to=info at itwrx.org>> wrote:
>
>> hi,
>>
>> i'm trying to use a letsencrypt generated cert with freeswitch but am
>> not sure how to proceed.
>
>  Suggest you proceed to take a look at the docs here:

https://wiki.freeswitch.org/wiki/SIP_TLS

Steps 1 & 2 involve making your own certificate; you don't have to use
letsencrypt certs.

Start with Step 3 and proceed from there, as Bryan mentioned
there is a lot more to it than '..just copied the pem formatted cert that
certbot generated to /etc/freeswitch/tls and named it tls.pem.'

regards
-kim
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From brian at freeswitch.org  Fri Jan  6 19:07:24 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 6 Jan 2017 10:07:24 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <CAKZxVQXG_dhMmEQ2+chwMxBex2i86a=xMS9wxXEB6KDYPizfvw@mail.gmail.com>
References: <CAKZxVQXG_dhMmEQ2+chwMxBex2i86a=xMS9wxXEB6KDYPizfvw@mail.gmail.com>
Message-ID: <CAEJMVkBqby+VGWa_d63zDwYjPe_r3UC=99gFHamu4W7nNdpwDA@mail.gmail.com>

You probably need to put in the chain cert there somewhere too.

On Fri, Jan 6, 2017 at 6:23 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

> On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org
>> <https://webmail.kim.net/src/compose.php?send_to=info at itwrx.org>> wrote:
>>
>>> hi,
>>>
>>> i'm trying to use a letsencrypt generated cert with freeswitch but am
>>> not sure how to proceed.
>>
>>  Suggest you proceed to take a look at the docs here:
>
> https://wiki.freeswitch.org/wiki/SIP_TLS
>
> Steps 1 & 2 involve making your own certificate; you don't have to use
> letsencrypt certs.
>
> Start with Step 3 and proceed from there, as Bryan mentioned
> there is a lot more to it than '..just copied the pem formatted cert that
> certbot generated to /etc/freeswitch/tls and named it tls.pem.'
>
> regards
> -kim
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From info at itwrx.org  Fri Jan  6 21:19:05 2017
From: info at itwrx.org (ITwrx.org)
Date: Fri, 6 Jan 2017 12:19:05 -0600
Subject: [Freeswitch-users] tls with letsencrypt
In-Reply-To: <CAND18T2ZiBF2PYWP8+imvpA9a3p=srtfaBSmY-KKqCx9Ob-yqA@mail.gmail.com>
References: <342ae9b6-43a2-f21b-0793-703ad5e4a004@itwrx.org>
	<CAEJMVkAFOMLK9ykA0n0B0S9SiA7Cbmq5vb3F8H_EFvOtWW4AEg@mail.gmail.com>
	<ba1e2e19-800f-08ad-d10a-31a2bd68d7dd@itwrx.org>
	<CAEJMVkDN8khPRvkx1PrXzyi9+zf2Mkbvd6dV3xq2mkPLyS9d4w@mail.gmail.com>
	<35824367-2e60-e2b4-61d8-ee2dc6428c76@itwrx.org>
	<CAND18T2ZiBF2PYWP8+imvpA9a3p=srtfaBSmY-KKqCx9Ob-yqA@mail.gmail.com>
Message-ID: <f8f8b27a-e97c-d084-633f-c5ef937d7b6a@itwrx.org>

Thanks for everyone's input. i ended up concatenating the cert, the
intermediate cert and the key from letsencrypt as tls.pem and i can
register and make calls with linphone desktop client over tls. Before, i
had the cert and the intermediate cert concatenated as tls.pem. :)

csipsimple still causes the "dh_lib" error, however. Is this caused by a
cipher suite mismatch between freeswitch and csipsimple? or something else?

thanks.

On 01/06/2017 04:58 AM, Mirko Brankovic wrote:
> Hey,
> All I had to do to get it work is to place cert and key in one pem
> file for FS, so like:
> cat /etc/letsencrypt/live/${domain}/cert.pem
> /etc/letsencrypt/live/${domain}/privkey.pem >
> /usr/local/freeswitch/certs/wss.pem
>
> On Fri, Jan 6, 2017 at 3:24 AM, ITwrx.org <info at itwrx.org
> <mailto:info at itwrx.org>> wrote:
>
>     dtls-srtp.pem,
>     tls.pem(the "stand in" i previously described),
>     and the original (could be from my old server where i set up tls
>     following the freeswitch wiki) tls.pem which has been renamed to
>     tls.pem.orig.
>
>
>     On 01/05/2017 06:43 PM, Brian West wrote:
>>     There is a lot more to it than that, what files are in that tls
>>     folder?
>>
>>     On Thu, Jan 5, 2017 at 4:53 PM, ITwrx.org <info at itwrx.org
>>     <mailto:info at itwrx.org>> wrote:
>>
>>         i just copied the pem formatted cert that certbot generated
>>         to /etc/freeswitch/tls and named it tls.pem. it's
>>         freeswitch:freeswitch 660 for perms. freeswitch seems capable
>>         of reading it, as the tls enabled profile starts up. i only
>>         get an error in fs_cli when the csipsimple client tries to
>>         connect using tls.
>>
>>         thanks
>>
>>
>>         On 01/05/2017 04:36 PM, Brian West wrote:
>>>         How did you format the cert? and in what files did you put
>>>         them in? and are your permissions correct on those files?
>>>
>>>         On Thu, Jan 5, 2017 at 2:55 PM, ITwrx.org <info at itwrx.org
>>>         <mailto:info at itwrx.org>> wrote:
>>>
>>>             hi,
>>>
>>>             i'm trying to use a letsencrypt generated cert with
>>>             freeswitch but am
>>>             not sure how to proceed. I've read the old and new wiki
>>>             posts concerning
>>>             tls but they don't seem to cover my exact scenario. It
>>>             seems to me that
>>>             freeswitch is looking into the configured "tls-cert-dir"
>>>             for the
>>>             hardcoded filename tls.pem and is expecting that a self
>>>             generated ca has
>>>             signed it. i have placed the fullchain.pem in that
>>>             directory (generated
>>>             with certbot) and have renamed it tls.pem but i guess
>>>             it's not finding
>>>             the CA sig it expects(?) as i'm getting:
>>>
>>>             tport_tls.c:1044 tls_connect()
>>>             tls_connect(0x373c000e8d0): TLS setup
>>>             failed (error:00000005:lib(0):func(0):DH lib)
>>>
>>>             when trying to connect with csipsimple from phone. I
>>>             would like to avoid
>>>             generating client certs signed by a custom CA where
>>>             users have to copy
>>>             the client cert and ca cert to their device as it adds
>>>             complexity and
>>>             problems. Is there a workaround or suggested method for
>>>             using a
>>>             letsencrypt cert with freeswitch so that clients like
>>>             csipsimple can
>>>             just validate against their built-in CA store?
>>>
>>>             thanks in advance,
>>>             ITwrx
>>>
>>>             --
>>>             Information Technology Works
>>>             https://ITwrx.org
>>>             @ITwrxorg
>>>
>>>
>>>             _________________________________________________________________________
>>>             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://confluence.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
>>>             <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
>>>
>>>
>>>
>>>
>>>         -- 
>>>
>>>         */Brian West/*
>>>         brian at freeswitch.org <mailto:brian at freeswitch.org>
>>>
>>>
>>>         */Twitter: @FreeSWITCH , @briankwest/*
>>>         http://www.freeswitchbook.com 
>>>         http://www.freeswitchcookbook.com
>>>         <http://www.freeswitchcookbook.com>
>>>         https://www.gofundme.com/freeswitch_ubuntu
>>>         <https://www.gofundme.com/freeswitch_ubuntu>
>>>
>>>         Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>>         Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>>         *T:*+19184209001 <tel:%28918%29%20420-9001> |
>>>         *F:*+19184209002 <tel:%28918%29%20420-9002> |
>>>         *M:*+1918424WEST (9378)
>>>         *Skype:*briankwest
>>>
>>>
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services: 
>>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>         http://www.freeswitchsolutions.com
>>>         <http://www.freeswitchsolutions.com>
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://confluence.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
>>>         <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
>>
>>         -- 
>>         Information Technology Works
>>         https://ITwrx.org
>>         @ITwrxorg
>>
>>         _________________________________________________________________________
>>         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://confluence.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
>>         <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 
>>
>>     -- 
>>
>>     */Brian West/* brian at freeswitch.org <mailto:brian at freeswitch.org>
>>
>>     */Twitter: @FreeSWITCH , @briankwest/*
>>     http://www.freeswitchbook.com  http://www.freeswitchcookbook.com
>>     <http://www.freeswitchcookbook.com>https://www.gofundme.com/freeswitch_ubuntu
>>     <https://www.gofundme.com/freeswitch_ubuntu>
>>
>>     Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>     Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>>     *T:*+19184209001 <tel:+1%20918-420-9001> | *F:*+19184209002
>>     <tel:+1%20918-420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services: 
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.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
>>     <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
>
>     -- 
>     Information Technology Works
>     https://ITwrx.org
>     @ITwrxorg
>
>     _________________________________________________________________________
>     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://confluence.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
>     <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 
>
> -- 
> Regards,
> Mirko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Information Technology Works
https://ITwrx.org
@ITwrxorg

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From john.nash778 at gmail.com  Mon Jan  9 10:57:35 2017
From: john.nash778 at gmail.com (John Nash)
Date: Mon, 9 Jan 2017 13:27:35 +0530
Subject: [Freeswitch-users] Freeswitch install from yum
Message-ID: <CAA4rGPxc7099C8ScALsxaqxnkUfDySo1OshoVERhbtS4O9obRQ@mail.gmail.com>

I installed freeswitch using yum commands as per given
https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7

All seems to be OK but I have two questions ..

1- I do not find ulimit settings in any startup file as recommend . I see
files /etc/sysconfig/freeswitch,
/etc/systemd/system/multi-user.target.wants/freeswitch.service
 am I supposed to create some kind of init script and mention ulimit
settings there?

2- In my box I see lua version as Lua 5.1.4  Copyright (C) 1994-2008
Lua.org, PUC-Rio . Is higher version of lua recommended? Any memory related
issue with 5.1.4 version?
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From tony at intelecenter.com  Mon Jan  9 12:30:43 2017
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Mon, 9 Jan 2017 09:30:43 +0000
Subject: [Freeswitch-users] Freeswitch install from yum
In-Reply-To: <CAA4rGPxc7099C8ScALsxaqxnkUfDySo1OshoVERhbtS4O9obRQ@mail.gmail.com>
References: <CAA4rGPxc7099C8ScALsxaqxnkUfDySo1OshoVERhbtS4O9obRQ@mail.gmail.com>
Message-ID: <CAANhKX+ebFO2bJgiMZyHffXBXw0XxYz5LqfYYa-ueXrb6xxvtw@mail.gmail.com>

John-


you can add these to your file (see below):
/etc/systemd/system/multi-user.target.wants/freeswitch.service


a symlink is created to this at /usr/lib/systemd/system/freeswitch.service
 when you enable the service with systemd (systemctl enable freeswitch)



[Unit]

Description=FreeSWITCH

Wants=network-online.target

After=syslog.target network.target network-online.target

After=mariadb.service httpd.service



[Service]

Type=forking

User=freeswitch

WorkingDirectory= /var/run/freeswitch

PIDFile=/var/run/freeswitch/freeswitch.pid

EnvironmentFile=-/etc/sysconfig/freeswitch

ExecStart=/usr/bin/freeswitch -ncwait -nonat $FREESWITCH_PARAMS

ExecReload=/usr/bin/kill -HUP $MAINPID

LimitFSIZE= unlimited

LimitDATA= unlimited

LimitSTACK= 245760

LimitCORE= unlimited

LimitNOFILE=999999

LimitAS= unlimited

LimitNPROC= unlimited

LimitMEMLOCK= unlimited

LimitLOCKS= unlimited

LimitSIGPENDING= unlimited

LimitMSGQUEUE= unlimited

[Install]

WantedBy=multi-user.target

On Mon, Jan 9, 2017 at 7:57 AM, John Nash <john.nash778 at gmail.com> wrote:

> I installed freeswitch using yum commands as per given
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7
>
> All seems to be OK but I have two questions ..
>
> 1- I do not find ulimit settings in any startup file as recommend . I see
> files /etc/sysconfig/freeswitch, /etc/systemd/system/multi-user.target.wants/freeswitch.service
>  am I supposed to create some kind of init script and mention ulimit
> settings there?
>
> 2- In my box I see lua version as Lua 5.1.4  Copyright (C) 1994-2008
> Lua.org, PUC-Rio . Is higher version of lua recommended? Any memory related
> issue with 5.1.4 version?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From manpower13.cse at gmail.com  Mon Jan  9 14:50:56 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 9 Jan 2017 17:20:56 +0530
Subject: [Freeswitch-users] Gateway outbound configuration
Message-ID: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>

Hai,

       I try to use reliance sip trunking directly VOIP client(zoiper) its
work fine ,but when i try to use with freeswitch gateway i am getting
always fails bellow is my siptrunking details.


Details:
   Username:+914146380100
  password:12345
  proxy: 10.237.246.225
  domain: bangalore.relianceims.in
  Auth username:+914146380100 at bangalore.relianceims.in

*My Gateway XML*


<include>
  <gateway name="reliance">
  <param name="username" value="+914146380100"/>

      <param name="password" value="123456"/>

  <param name="register" value="true"/>
  <param name="caller-id-in-from" value="true"/>
  <param name="realm" value="bangalore.relianceims.in"/>
  <param name="proxy" value="10.21.254.231"/>
  <param name="ping" value="25"/>
</gateway>
</include>
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From bipin at xbipin.com  Mon Jan  9 15:22:15 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 9 Jan 2017 16:22:15 +0400
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
Message-ID: <b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>

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From manpower13.cse at gmail.com  Mon Jan  9 15:49:38 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 9 Jan 2017 18:19:38 +0530
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
Message-ID: <CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>

Thanks for your quick reply i tried with  extension-in-contact param but
still no luck ,I attached my wiresharke trace(make sure use filter for SIP)


Thanks
Murugan Pandian

On Mon, Jan 9, 2017 at 5:52 PM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> can u try with the below line as well
> <param name="extension-in-contact" value="true"/>
>
> also does the reliance gateway send a proxy authentication required 407
> when u try to register first time?
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: [Freeswitch-users] Gateway outbound configuration
> From: Murugan Pandian <manpower13.cse at gmail.com>
> <manpower13.cse at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/9/2017, 3:50:56 PM
>
> Hai,
>
>        I try to use reliance sip trunking directly VOIP client(zoiper) its
> work fine ,but when i try to use with freeswitch gateway i am getting
> always fails bellow is my siptrunking details.
>
>
> Details:
>    Username:+914146380100
>   password:12345
>   proxy: 10.237.246.225
>   domain: bangalore.relianceims.in
>   Auth username:+914146380100 at bangalore.relianceims.in
>
> *My Gateway XML*
>
>
> <include>
>   <gateway name="reliance">
>   <param name="username" value="+914146380100"/>
>
>       <param name="password" value="123456"/>
>
>   <param name="register" value="true"/>
>   <param name="caller-id-in-from" value="true"/>
>   <param name="realm" value="bangalore.relianceims.in"/>
>   <param name="proxy" value="10.21.254.231"/>
>   <param name="ping" value="25"/>
> </gateway>
> </include>
>
>
>
> _________________________________________________________________________
> 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 bipin at xbipin.com  Mon Jan  9 17:07:09 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 9 Jan 2017 18:07:09 +0400
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
Message-ID: <19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>

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From bipin at xbipin.com  Mon Jan  9 17:41:43 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 9 Jan 2017 18:41:43 +0400
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
Message-ID: <28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>

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From manpower13.cse at gmail.com  Mon Jan  9 18:03:32 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 9 Jan 2017 20:33:32 +0530
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
	<28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
Message-ID: <CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>

HI when i try to use bangalore.relianceims.in in proxy i am getting  DNS
error,i attached zoiper trace

On Mon, Jan 9, 2017 at 8:11 PM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> also in ur proxy field use the domain name which u use in realm rather
> than the ip address
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Gateway outbound configuration
> From: Bipin Patel <bipin at xbipin.com> <bipin at xbipin.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/9/2017, 6:07:09 PM
>
> hi,
>
> can u send a similar trace with zoiper where it works?
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Gateway outbound configuration
> From: Murugan Pandian <manpower13.cse at gmail.com>
> <manpower13.cse at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/9/2017, 4:49:38 PM
>
> Thanks for your quick reply i tried with  extension-in-contact param but
> still no luck ,I attached my wiresharke trace(make sure use filter for SIP)
>
>
> Thanks
> Murugan Pandian
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vivek at advaitamtech.com  Mon Jan  9 13:28:22 2017
From: vivek at advaitamtech.com (vivek at advaitamtech.com)
Date: Mon, 9 Jan 2017 15:58:22 +0530 (IST)
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
	Authentication Required"
Message-ID: <1483957702.686832416@apps.rackspace.com>


Hi All,
 
Am trying to proxy the "407 Proxy Authentication Required" through freeswitch,
 
My Objective is to proxy the "407 Proxy Authentication Required" received from the third party server to another server which is connected to freeswitch.
 
Is it possible to do this?. If possible how do I do it. Please help me out.
 
Thanks,
Vivek.
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From mike at jerris.com  Mon Jan  9 19:04:14 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 9 Jan 2017 11:04:14 -0500
Subject: [Freeswitch-users] Freeswitch install from yum
In-Reply-To: <CAA4rGPxc7099C8ScALsxaqxnkUfDySo1OshoVERhbtS4O9obRQ@mail.gmail.com>
References: <CAA4rGPxc7099C8ScALsxaqxnkUfDySo1OshoVERhbtS4O9obRQ@mail.gmail.com>
Message-ID: <31DB5163-0434-4B6A-92E5-01FDB038BF5A@jerris.com>

mod_lua will work with lua 5.1 and lua 5.2.  I recommend lua 5.2 as thats what is used on debian and is better tested, but to my knowledge both work well.


> On Jan 9, 2017, at 2:57 AM, John Nash <john.nash778 at gmail.com> wrote:
> 
> I installed freeswitch using yum commands as per given https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 <https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7>
> 
> All seems to be OK but I have two questions ..
> 
> 1- I do not find ulimit settings in any startup file as recommend . I see files /etc/sysconfig/freeswitch, /etc/systemd/system/multi-user.target.wants/freeswitch.service  am I supposed to create some kind of init script and mention ulimit settings there?
> 
> 2- In my box I see lua version as Lua 5.1.4  Copyright (C) 1994-2008 Lua.org, PUC-Rio . Is higher version of lua recommended? Any memory related issue with 5.1.4 version?

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From bipin at xbipin.com  Mon Jan  9 19:28:53 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 9 Jan 2017 20:28:53 +0400
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
	<28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
	<CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>
Message-ID: <d01bc4ba-beaf-a9fc-ed2e-bbfd38b88e85@xbipin.com>

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From paul.mateer at outlook.com  Mon Jan  9 19:58:53 2017
From: paul.mateer at outlook.com (Paul Mateer)
Date: Mon, 9 Jan 2017 16:58:53 +0000
Subject: [Freeswitch-users] No input audio with FSClient
Message-ID: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone

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From mitch.capper at gmail.com  Mon Jan  9 20:08:13 2017
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 9 Jan 2017 09:08:13 -0800
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>

Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch
itself see if the audio is on the client but not FS.  Enabling logging on
the client/server using fs_cli and see anything interesting there.  Finally
make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com> wrote:

> I seem to have a problem with audio input when using FSClient.
>
>
>
> I have one box running FreeSWITCH and another running FSClient. I can call
> the server no problem and get audio back (I dialled 9198 to get the Tetris
> tune) but when I provide an audio stream using the mic and dial 9196 I get
> nothing back.
>
>
>
> I know the mic is providing sound and the FreeSWITCH server is operating
> OK because I can use X-Lite to perform the same test and I get the audio
> feed played back to me.
>
>
>
> I'm not sure if there is something amiss in the configuration of FSClient
> (although it should be the default config) or if something else is amiss
> (there doesn't appear to be anything odd in the FreeSWITCH log for the
> client).
>
>
>
> Does anyone have any thoughts on what might be wrong, or what i should
> look at?
>
>
>
> Thanks,
>
>
>
> Paul
>
>
>
> Sent from my Windows 10 phone
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manpower13.cse at gmail.com  Mon Jan  9 20:09:13 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 9 Jan 2017 22:39:13 +0530
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <d01bc4ba-beaf-a9fc-ed2e-bbfd38b88e85@xbipin.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
	<28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
	<CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>
	<d01bc4ba-beaf-a9fc-ed2e-bbfd38b88e85@xbipin.com>
Message-ID: <CAMRuXGhHRjA7ax0wGFKLvXqwPhiqiUm+EB_5VVbGqytiX6J2uw@mail.gmail.com>

Still no luck for my understanding from trace Zoiper request URI and
freeswitch gateway request URI different

>From Zoiper: sip:bangalore.relianceims.in
>From fs gateway: sip: 10.237.246.255

I am getting 404 error

On Mon, Jan 9, 2017 at 9:58 PM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> ok so it seems the reliance server needs the domain name in the from
> domain, put the below in the gateway settings and try again, it should work
>
> <param name="from-domain" value="bangalore.relianceims.in"/>
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Gateway outbound configuration
> From: Murugan Pandian <manpower13.cse at gmail.com>
> <manpower13.cse at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 1/9/2017, 7:03:32 PM
>
> HI when i try to use bangalore.relianceims.in in proxy i am getting  DNS
> error,i attached zoiper trace
>
> On Mon, Jan 9, 2017 at 8:11 PM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> also in ur proxy field use the domain name which u use in realm rather
>> than the ip address
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: Re: [Freeswitch-users] Gateway outbound configuration
>> From: Bipin Patel <bipin at xbipin.com> <bipin at xbipin.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 1/9/2017, 6:07:09 PM
>>
>> hi,
>>
>> can u send a similar trace with zoiper where it works?
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: Re: [Freeswitch-users] Gateway outbound configuration
>> From: Murugan Pandian <manpower13.cse at gmail.com>
>> <manpower13.cse at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 1/9/2017, 4:49:38 PM
>>
>> Thanks for your quick reply i tried with  extension-in-contact param but
>> still no luck ,I attached my wiresharke trace(make sure use filter for SIP)
>>
>>
>> Thanks
>> Murugan Pandian
>>
>>
>>
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manpower13.cse at gmail.com  Mon Jan  9 21:06:06 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 9 Jan 2017 23:36:06 +0530
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <CAMRuXGhHRjA7ax0wGFKLvXqwPhiqiUm+EB_5VVbGqytiX6J2uw@mail.gmail.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
	<28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
	<CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>
	<d01bc4ba-beaf-a9fc-ed2e-bbfd38b88e85@xbipin.com>
	<CAMRuXGhHRjA7ax0wGFKLvXqwPhiqiUm+EB_5VVbGqytiX6J2uw@mail.gmail.com>
Message-ID: <CAMRuXGhoxNDg1MnkHoYkDZ_LKruZ6_6RCSCX-ai8dhxfkPJyNA@mail.gmail.com>

Thanks for your help finally its work bellow is my gateway config


<include>
  <gateway name="reliance">
  <param name="username" value="+91414655252"/>

  <param name="auth-username" value="+91414655252 at bangalore.relianceims.in"/>

 <param name="password" value="123456"/>

<param name="register" value="true"/>

<param name="realm" value="bangalore.relianceims.in"/>

<param name="proxy" value="bangalore.relianceims.in"/>

<param name="regitser-proxy" value="10.237.246.255"/>

<param name="from-domain" value="bangalore.relianceims.in"/>

<param name="extension-in-contact" value="true"/>

  <param name="caller-id-in-from" value="true"/>

</gateway></include>


Regard's

Murugan Pandian


On Mon, Jan 9, 2017 at 10:39 PM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> Still no luck for my understanding from trace Zoiper request URI and
> freeswitch gateway request URI different
>
> From Zoiper: sip:bangalore.relianceims.in
> From fs gateway: sip: 10.237.246.255
>
> I am getting 404 error
>
> On Mon, Jan 9, 2017 at 9:58 PM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> ok so it seems the reliance server needs the domain name in the from
>> domain, put the below in the gateway settings and try again, it should work
>>
>> <param name="from-domain" value="bangalore.relianceims.in"/>
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: Re: [Freeswitch-users] Gateway outbound configuration
>> From: Murugan Pandian <manpower13.cse at gmail.com>
>> <manpower13.cse at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 1/9/2017, 7:03:32 PM
>>
>> HI when i try to use bangalore.relianceims.in in proxy i am getting  DNS
>> error,i attached zoiper trace
>>
>> On Mon, Jan 9, 2017 at 8:11 PM, Bipin Patel <bipin at xbipin.com> wrote:
>>
>>> hi,
>>>
>>> also in ur proxy field use the domain name which u use in realm rather
>>> than the ip address
>>>
>>>
>>> Regards,
>>> Bipin
>>>
>>>
>>> ------------------------------
>>> -------- Original Message --------
>>> Subject: Re: [Freeswitch-users] Gateway outbound configuration
>>> From: Bipin Patel <bipin at xbipin.com> <bipin at xbipin.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: 1/9/2017, 6:07:09 PM
>>>
>>> hi,
>>>
>>> can u send a similar trace with zoiper where it works?
>>>
>>>
>>> Regards,
>>> Bipin
>>>
>>>
>>> ------------------------------
>>> -------- Original Message --------
>>> Subject: Re: [Freeswitch-users] Gateway outbound configuration
>>> From: Murugan Pandian <manpower13.cse at gmail.com>
>>> <manpower13.cse at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: 1/9/2017, 4:49:38 PM
>>>
>>> Thanks for your quick reply i tried with  extension-in-contact param
>>> but still no luck ,I attached my wiresharke trace(make sure use filter for
>>> SIP)
>>>
>>>
>>> Thanks
>>> Murugan Pandian
>>>
>>>
>>>
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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.mateer at outlook.com  Mon Jan  9 21:13:20 2017
From: paul.mateer at outlook.com (Paul Mateer)
Date: Mon, 9 Jan 2017 18:13:20 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>,
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
Message-ID: <AM5PR10MB07086D5C9DE6957EF3172FBDF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

Hi Mitch. Thanks for getting back so quickly.

I have the both the Main Input and Speakerphone Input set to ?Microphone (High Definition Audio Device)? in the Sofia Settings, although switching them to the alternative (Primary Sound Capture Driver) makes no difference.

I know that the problem is at the server end of things because when I monitor communications between the client and the server I don't see any outgoing RTP packet traffic after the INVITE request is accepted by the server even though FSClient indicates the call has been answered.

I didn't know that you could use fs_cli to configure logging in FreeSWITCH that is embedded in FSClient ? presumably you need to specify particular values for some of the parameters to do that?

Sent from my Windows 10 phone

From: Mitch Capper<mailto:mitch.capper at gmail.com>
Sent: 09 January 2017 17:11
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] No input audio with FSClient

Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch itself see if the audio is on the client but not FS.  Enabling logging on the client/server using fs_cli and see anything interesting there.  Finally make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com<mailto:paul.mateer at outlook.com>> wrote:
I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone


_________________________________________________________________________
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 anthony.minessale at gmail.com  Mon Jan  9 21:25:46 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 9 Jan 2017 12:25:46 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
Message-ID: <CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>

So that concludes that media is already established on the webrtc end and
there is no problem with that.
The box you are calling on SIP is also FS, you may want to add a sleep 2000
in that dialplan before the voicemail.
Also since webrtc has no ringing indication you may want to set the
variable ringback to get some audible feedback when making calls.


On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
wrote:

> Hello Anthony,
>
> Thanks for your reply.
>
> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
> with ~2 seconds) instead of silence_stream.
> When i make the call from verto client, i ear the audio file, then no
> audio for ~2/3 seconds and then i ear "id followed by pound" (audio cut
> off from voicemail initial message "Please enter your id followed by
> pound").
>
> I checked if i have the variable answer_delay and i don't have it.
>
> The log of this call is at https://pastebin.freeswitch.org/view/e130e172 .
>
> There is any thing more that i can do?
>
>
> Best Regards,
> Jose Lopes
>
> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> Also make sure you don't have answer_delay set in your vars.xml
>>
>>
>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Try making the call with
>>>
>>> fsctl debug_level 10
>>>
>>> and observe the logs, answer followed by silence_stream should send
>>> audio to the client.
>>> Also try playing an audio file instead of silence stream to see if you
>>> hear it.
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>>> wrote:
>>>
>>>> Hello Brian,
>>>>
>>>> Thanks for your reply.
>>>>
>>>> I tried the dialplan bellow with silence_stream://2000, and i have that
>>>> issue.
>>>> I tried with silence_stream://3000 and the audio cut off is greater.
>>>> Without the playback, there is no audio cut off, but FreeSwitch doesn't
>>>> send any rtp packets to verto client before the bridge.
>>>>
>>>> There is any thing more that i can do?
>>>>
>>>>
>>>> <include>
>>>>   <context name="default">
>>>>     <extension name="call_debug" continue="true">
>>>>       <condition field="${call_debug}" expression="^true$"
>>>> break="never">
>>>>         <action application="info"/>
>>>>       </condition>
>>>>     </extension>
>>>>     <extension name="itsp_send_call">
>>>>       <condition field="destination_number" expression="^.*$">
>>>>         <action application="answer"/>
>>>>         <action application="playback" data="silence_stream://2000"/>
>>>>         <action application="bridge" data="{absolute_codec_string='
>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>       </condition>
>>>>     </extension>
>>>>   </context>
>>>> </include>
>>>>
>>>>
>>>> Best Regards,
>>>> Jose Lopes
>>>>
>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>
>>>>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>>>>
>>>>> /b
>>>>>
>>>>>
>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>>>>
>>>>>> hi,
>>>>>>
>>>>>> i have the same issue, i think its related to slow audio setup during
>>>>>> the call
>>>>>>
>>>>>>
>>>>>> Regards,
>>>>>> Bipin
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>> -------- Original Message --------
>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto
>>>>>> call to sip external voicemail
>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>
>>>>>> Hello Guys,
>>>>>>
>>>>>> I have audio cut off at the begin of the verto call to  FreeSwitch
>>>>>> that redirect to sip external voicemail (Access voicemail mailbox) .
>>>>>>
>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i use
>>>>>> opus at verto codecs, there is no issue, but this causes audio transcoding)
>>>>>> .
>>>>>>
>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>>>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>> I notice if i remove the playback action, there is no issue. But I
>>>>>> need the playback action to send rtp packets to verto client.
>>>>>>
>>>>>> I simulate this using another FreeSwitch as external voicemail server
>>>>>> and I only listen "id followed by pound" from the initial message of
>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>> .org/view/507fa115
>>>>>>
>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type of
>>>>>> call?
>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>
>>>>>>
>>>>>> Best regards,
>>>>>> Jose Lopes
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>
>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>
>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>> *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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-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 <(919)%20386-9900>
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>
>>
>>
>> --
>> 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 <(919)%20386-9900>
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From paul.mateer at outlook.com  Mon Jan  9 22:37:24 2017
From: paul.mateer at outlook.com (Paul Mateer)
Date: Mon, 9 Jan 2017 19:37:24 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <AM5PR10MB07086D5C9DE6957EF3172FBDF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>,
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>,
	<AM5PR10MB07086D5C9DE6957EF3172FBDF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <AM5PR10MB07087336FB7495543271C1E1F0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

Sorry. Just re-rest my last email and realised there was a typo ? it should have read ?I know that the problem is NOT at the server end of things?

Sorry about that.

Paul

Sent from my Windows 10 phone

From: Paul Mateer<mailto:paul.mateer at outlook.com>
Sent: 09 January 2017 18:16
To: Mitch Capper<mailto:mitch.capper at gmail.com>; FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] No input audio with FSClient

Hi Mitch. Thanks for getting back so quickly.

I have the both the Main Input and Speakerphone Input set to ?Microphone (High Definition Audio Device)? in the Sofia Settings, although switching them to the alternative (Primary Sound Capture Driver) makes no difference.

I know that the problem is at the server end of things because when I monitor communications between the client and the server I don't see any outgoing RTP packet traffic after the INVITE request is accepted by the server even though FSClient indicates the call has been answered.

I didn't know that you could use fs_cli to configure logging in FreeSWITCH that is embedded in FSClient ? presumably you need to specify particular values for some of the parameters to do that?

Sent from my Windows 10 phone

From: Mitch Capper<mailto:mitch.capper at gmail.com>
Sent: 09 January 2017 17:11
To: FreeSWITCH Users Help<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] No input audio with FSClient

Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch itself see if the audio is on the client but not FS.  Enabling logging on the client/server using fs_cli and see anything interesting there.  Finally make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com<mailto:paul.mateer at outlook.com>> wrote:
I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone


_________________________________________________________________________
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 mitch.capper at gmail.com  Mon Jan  9 23:44:53 2017
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 9 Jan 2017 12:44:53 -0800
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
Message-ID: <CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>

Under options in FSClient for the event socket you can change them but
otherwise just connect like normal debugging and sip_trace might be a good
start,  you can use fs_logger as well to collect the logs for you if
desired.  Let us know the result of the recording test as well.


~mitch

On Mon, Jan 9, 2017 at 9:08 AM, Mitch Capper <mitch.capper at gmail.com> wrote:

> Hi Paul,
> One place to start would be to record the call with FSClient and
> Freeswitch itself see if the audio is on the client but not FS.  Enabling
> logging on the client/server using fs_cli and see anything interesting
> there.  Finally make sure you have the right mic input selected in the
> FSClient options.
>
>
>
>
> ~mitch
>
> On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com>
> wrote:
>
>> I seem to have a problem with audio input when using FSClient.
>>
>>
>>
>> I have one box running FreeSWITCH and another running FSClient. I can
>> call the server no problem and get audio back (I dialled 9198 to get the
>> Tetris tune) but when I provide an audio stream using the mic and dial 9196
>> I get nothing back.
>>
>>
>>
>> I know the mic is providing sound and the FreeSWITCH server is operating
>> OK because I can use X-Lite to perform the same test and I get the audio
>> feed played back to me.
>>
>>
>>
>> I'm not sure if there is something amiss in the configuration of FSClient
>> (although it should be the default config) or if something else is amiss
>> (there doesn't appear to be anything odd in the FreeSWITCH log for the
>> client).
>>
>>
>>
>> Does anyone have any thoughts on what might be wrong, or what i should
>> look at?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Paul
>>
>>
>>
>> Sent from my Windows 10 phone
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From gonzalo at parzee.com  Tue Jan 10 11:26:01 2017
From: gonzalo at parzee.com (Gonzalo Gasca Meza)
Date: Tue, 10 Jan 2017 02:26:01 -0600
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
Message-ID: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>

Hi all,

I'm trying to install Speech Recognition engine.

Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)

Following this example:

https://wiki.freeswitch.org/wiki/Mod_pocketsphinx

<include>
   <extension name="pizza_demo">
     <condition field="destination_number" expression="^(pizza|74992)$"/>
     <condition field="${module_exists(mod_v8)}" expression="true"/>
     <condition field="${module_exists(mod_pocketsphinx)}" 
expression="true">
      <action application="javascript" data="ps_pizza.js"/>
     </condition>
   </extension>
</include>

I'm replacing mod_spidermonkey with mod_v8 based on this document and 
when I call this number I get:

EXECUTE sofia/external/14088053951 at 104.236.190.206 
javascript(ps_pizza.js)
2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception: 
SyntaxError: Unexpected identifier (near: "					if 
(result.interpretation. at confidence >= grammar_object.min_score) {")

If I change: result.interpretation. at confidence to 
result.interpretation.['@confidence']

EXECUTE sofia/external/14088053951 at 104.236.190.206 
export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT 
(export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
EXECUTE sofia/external/14088053951 at 104.236.190.206 
javascript(ps_pizza.js)
2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception: 
SyntaxError: Unexpected string (near: "					if 
(result.interpretation.['@confidence'] >= grammar_object.min_score) {")

SpeechTools.jm from:
https://raw.githubusercontent.com/lordnull/FreeSWITCH/master/scripts/javascript/js_modules/SpeechTools.jm

Similar discussion:
http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105818.html

Any pointers?
I get the mod_v8 module compiled from Freeswitch using this: 
https://freeswitch.org/confluence/display/FREESWITCH/mod_v8

Thanks





From findmeinwland at gmail.com  Tue Jan 10 11:32:21 2017
From: findmeinwland at gmail.com (=?UTF-8?B?0JDRgNGC0YPRgA==?=)
Date: Tue, 10 Jan 2017 13:32:21 +0500
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
In-Reply-To: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
References: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
Message-ID: <bc37c753-0953-567f-57f7-077e94478df0@gmail.com>

But it's a syntax error, no? I think you need to verify your code


10.01.2017 13:26, Gonzalo Gasca Meza ?????:
> Hi all,
>
> I'm trying to install Speech Recognition engine.
>
> Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)
>
> Following this example:
>
> https://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>
> <include>
>     <extension name="pizza_demo">
>       <condition field="destination_number" expression="^(pizza|74992)$"/>
>       <condition field="${module_exists(mod_v8)}" expression="true"/>
>       <condition field="${module_exists(mod_pocketsphinx)}"
> expression="true">
>        <action application="javascript" data="ps_pizza.js"/>
>       </condition>
>     </extension>
> </include>
>
> I'm replacing mod_spidermonkey with mod_v8 based on this document and
> when I call this number I get:
>
> EXECUTE sofia/external/14088053951 at 104.236.190.206
> javascript(ps_pizza.js)
> 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception:
> SyntaxError: Unexpected identifier (near: "					if
> (result.interpretation. at confidence >= grammar_object.min_score) {")
>
> If I change: result.interpretation. at confidence to
> result.interpretation.['@confidence']
>
> EXECUTE sofia/external/14088053951 at 104.236.190.206
> export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
> 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
> EXECUTE sofia/external/14088053951 at 104.236.190.206
> javascript(ps_pizza.js)
> 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception:
> SyntaxError: Unexpected string (near: "					if
> (result.interpretation.['@confidence'] >= grammar_object.min_score) {")
>
> SpeechTools.jm from:
> https://raw.githubusercontent.com/lordnull/FreeSWITCH/master/scripts/javascript/js_modules/SpeechTools.jm
>
> Similar discussion:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105818.html
>
> Any pointers?
> I get the mod_v8 module compiled from Freeswitch using this:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>
> 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 gascagonzalo at gmail.com  Tue Jan 10 11:37:50 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Tue, 10 Jan 2017 00:37:50 -0800
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
In-Reply-To: <bc37c753-0953-567f-57f7-077e94478df0@gmail.com>
References: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
	<bc37c753-0953-567f-57f7-077e94478df0@gmail.com>
Message-ID: <CAL+DTgLaExbYvomM+W-TQyjf7u5fnO=wjx22-H_QKKUyqZ7rmQ@mail.gmail.com>

Code is from sample Grammar provided by Wiki, not my original code...

On Tue, Jan 10, 2017 at 12:32 AM, ????? <findmeinwland at gmail.com> wrote:

> But it's a syntax error, no? I think you need to verify your code
>
>
> 10.01.2017 13:26, Gonzalo Gasca Meza ?????:
> > Hi all,
> >
> > I'm trying to install Speech Recognition engine.
> >
> > Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)
> >
> > Following this example:
> >
> > https://wiki.freeswitch.org/wiki/Mod_pocketsphinx
> >
> > <include>
> >     <extension name="pizza_demo">
> >       <condition field="destination_number"
> expression="^(pizza|74992)$"/>
> >       <condition field="${module_exists(mod_v8)}" expression="true"/>
> >       <condition field="${module_exists(mod_pocketsphinx)}"
> > expression="true">
> >        <action application="javascript" data="ps_pizza.js"/>
> >       </condition>
> >     </extension>
> > </include>
> >
> > I'm replacing mod_spidermonkey with mod_v8 based on this document and
> > when I call this number I get:
> >
> > EXECUTE sofia/external/14088053951 at 104.236.190.206
> > javascript(ps_pizza.js)
> > 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception:
> > SyntaxError: Unexpected identifier (near: "
>      if
> > (result.interpretation. at confidence >= grammar_object.min_score) {")
> >
> > If I change: result.interpretation. at confidence to
> > result.interpretation.['@confidence']
> >
> > EXECUTE sofia/external/14088053951 at 104.236.190.206
> > export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
> > 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT
> > (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
> > EXECUTE sofia/external/14088053951 at 104.236.190.206
> > javascript(ps_pizza.js)
> > 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception:
> > SyntaxError: Unexpected string (near: "
>      if
> > (result.interpretation.['@confidence'] >= grammar_object.min_score) {")
> >
> > SpeechTools.jm from:
> > https://raw.githubusercontent.com/lordnull/FreeSWITCH/
> master/scripts/javascript/js_modules/SpeechTools.jm
> >
> > Similar discussion:
> > http://lists.freeswitch.org/pipermail/freeswitch-users/
> 2014-June/105818.html
> >
> > Any pointers?
> > I get the mod_v8 module compiled from Freeswitch using this:
> > https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
> >
> > 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 jose.lopes at itcenter.com.pt  Tue Jan 10 12:58:36 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Tue, 10 Jan 2017 09:58:36 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
	<CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
Message-ID: <CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>

Hello Anthony,

At this replicated scenario, the box I am calling on SIP is FS.
But on real scenario, the box I am calling on SIP is not Freeswitch, it is
an external voicemail server and the initial message have audio cut off.

Thanks for the information about variable ringback, I am already using on
real scenario.

One strange thing is if I use the codec OPUS at verto, this issue doesn't
happen.
But I need to use codec PCMU to avoid audio transcoding.

Let me know if you need more information to debug this issue.

Best Regards,
Jose Lopes




2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> So that concludes that media is already established on the webrtc end and
> there is no problem with that.
> The box you are calling on SIP is also FS, you may want to add a sleep
> 2000 in that dialplan before the voicemail.
> Also since webrtc has no ringing indication you may want to set the
> variable ringback to get some audible feedback when making calls.
>
>
> On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
> wrote:
>
>> Hello Anthony,
>>
>> Thanks for your reply.
>>
>> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
>> with ~2 seconds) instead of silence_stream.
>> When i make the call from verto client, i ear the audio file, then no
>> audio for ~2/3 seconds and then i ear "id followed by pound" (audio cut
>> off from voicemail initial message "Please enter your id followed by
>> pound").
>>
>> I checked if i have the variable answer_delay and i don't have it.
>>
>> The log of this call is at https://pastebin.freeswitch.org/view/e130e172
>> .
>>
>> There is any thing more that i can do?
>>
>>
>> Best Regards,
>> Jose Lopes
>>
>> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> Also make sure you don't have answer_delay set in your vars.xml
>>>
>>>
>>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Try making the call with
>>>>
>>>> fsctl debug_level 10
>>>>
>>>> and observe the logs, answer followed by silence_stream should send
>>>> audio to the client.
>>>> Also try playing an audio file instead of silence stream to see if you
>>>> hear it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt
>>>> > wrote:
>>>>
>>>>> Hello Brian,
>>>>>
>>>>> Thanks for your reply.
>>>>>
>>>>> I tried the dialplan bellow with silence_stream://2000, and i have
>>>>> that issue.
>>>>> I tried with silence_stream://3000 and the audio cut off is greater.
>>>>> Without the playback, there is no audio cut off, but FreeSwitch
>>>>> doesn't send any rtp packets to verto client before the bridge.
>>>>>
>>>>> There is any thing more that i can do?
>>>>>
>>>>>
>>>>> <include>
>>>>>   <context name="default">
>>>>>     <extension name="call_debug" continue="true">
>>>>>       <condition field="${call_debug}" expression="^true$"
>>>>> break="never">
>>>>>         <action application="info"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>     <extension name="itsp_send_call">
>>>>>       <condition field="destination_number" expression="^.*$">
>>>>>         <action application="answer"/>
>>>>>         <action application="playback" data="silence_stream://2000"/>
>>>>>         <action application="bridge" data="{absolute_codec_string='
>>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>   </context>
>>>>> </include>
>>>>>
>>>>>
>>>>> Best Regards,
>>>>> Jose Lopes
>>>>>
>>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>>
>>>>>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>>
>>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>>>>>
>>>>>>> hi,
>>>>>>>
>>>>>>> i have the same issue, i think its related to slow audio setup
>>>>>>> during the call
>>>>>>>
>>>>>>>
>>>>>>> Regards,
>>>>>>> Bipin
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------
>>>>>>> -------- Original Message --------
>>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto
>>>>>>> call to sip external voicemail
>>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>>
>>>>>>> Hello Guys,
>>>>>>>
>>>>>>> I have audio cut off at the begin of the verto call to  FreeSwitch
>>>>>>> that redirect to sip external voicemail (Access voicemail mailbox) .
>>>>>>>
>>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i use
>>>>>>> opus at verto codecs, there is no issue, but this causes audio transcoding)
>>>>>>> .
>>>>>>>
>>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>>>>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>>> I notice if i remove the playback action, there is no issue. But I
>>>>>>> need the playback action to send rtp packets to verto client.
>>>>>>>
>>>>>>> I simulate this using another FreeSwitch as external voicemail
>>>>>>> server and I only listen "id followed by pound" from the initial message of
>>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>>> .org/view/507fa115
>>>>>>>
>>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type of
>>>>>>> call?
>>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>>
>>>>>>>
>>>>>>> Best regards,
>>>>>>> Jose Lopes
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Brian West*
>>>>>> brian at freeswitch.org
>>>>>>
>>>>>>
>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>> http://www.freeswitchbook.com
>>>>>> http://www.freeswitchcookbook.com
>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>
>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>
>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>> *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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-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 <(919)%20386-9900>
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>
>>>
>>>
>>> --
>>> 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 <(919)%20386-9900>
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.Mateer at outlook.com  Tue Jan 10 14:18:27 2017
From: Paul.Mateer at outlook.com (Paul Mateer)
Date: Tue, 10 Jan 2017 11:18:27 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>,
	<CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>
Message-ID: <AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

OK, so I amended the FreeSWITCH.xml for FSClient to add a record_session action before the bridge action of the "number" extension in the default context of the dialplan section, and the client records the input audio stream fine.

I did the same thing in the "echo" extension of the default.xml in the FreeSWITCH dialplan folder and I get a recording of silence. This would seem to fit in with the fact that I don't see any RTP traffic data from the client to the server.


I have logs from both the server and the client (Debug Level set to 9 with SIP Trace active) but there is nothing notable logged between the start of the call and it's termination (I can provide the logs if that's of any help - and a WireShark trace).


Paul


PS. Sorry about the mailing issue yesterday - I think I hit reply to all instead of just reply.


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Mitch Capper <mitch.capper at gmail.com>
Sent: 09 January 2017 20:44:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] No input audio with FSClient

Under options in FSClient for the event socket you can change them but otherwise just connect like normal debugging and sip_trace might be a good start,  you can use fs_logger as well to collect the logs for you if desired.  Let us know the result of the recording test as well.


~mitch

On Mon, Jan 9, 2017 at 9:08 AM, Mitch Capper <mitch.capper at gmail.com<mailto:mitch.capper at gmail.com>> wrote:
Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch itself see if the audio is on the client but not FS.  Enabling logging on the client/server using fs_cli and see anything interesting there.  Finally make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com<mailto:paul.mateer at outlook.com>> wrote:
I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone


_________________________________________________________________________
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 manpower13.cse at gmail.com  Tue Jan 10 16:57:59 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 10 Jan 2017 19:27:59 +0530
Subject: [Freeswitch-users] Regex Nothing to repeat Dialplan issue
Message-ID: <CAMRuXGisQh+FPO81XeW26qQeqNW=Gi3vdvnZ-f1NbkL5k7yGMw@mail.gmail.com>

HI,

  When i receive inboung call from , iam getting dialplan condition error ,

Compile error: 1 [nothing to repeat] inbound number contain + like +91414.
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From brian at freeswitch.org  Tue Jan 10 18:18:01 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Jan 2017 09:18:01 -0600
Subject: [Freeswitch-users] Regex Nothing to repeat Dialplan issue
In-Reply-To: <CAMRuXGisQh+FPO81XeW26qQeqNW=Gi3vdvnZ-f1NbkL5k7yGMw@mail.gmail.com>
References: <CAMRuXGisQh+FPO81XeW26qQeqNW=Gi3vdvnZ-f1NbkL5k7yGMw@mail.gmail.com>
Message-ID: <CAEJMVkD3WFsq5b8ZgzM=oiKD+UGqf+pjy8T8Nr07D5qMHsAHfA@mail.gmail.com>

Show us your condition and the log of the compare.

/b

On Tue, Jan 10, 2017 at 7:57 AM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> HI,
>
>   When i receive inboung call from , iam getting dialplan condition error ,
>
> Compile error: 1 [nothing to repeat] inbound number contain + like +91414.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From tayeb.meftah at gmail.com  Tue Jan 10 13:26:06 2017
From: tayeb.meftah at gmail.com (Tayeb Meftah)
Date: Tue, 10 Jan 2017 11:26:06 +0100
Subject: [Freeswitch-users] Freeswitch and JSSip, getting 488
Message-ID: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>

Hey guys
it have been a long time;)
i am having an issue with JSSip and Freeswitch (WSS)
lovely no signed certificate... ;)
the issue is i am getting 488
sip traces:
http://paste.debian.net/908067/
thanks!


From Paul.Mateer at outlook.com  Tue Jan 10 18:31:34 2017
From: Paul.Mateer at outlook.com (Paul Mateer)
Date: Tue, 10 Jan 2017 15:31:34 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>,
	<CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>,
	<AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <AM5PR10MB0708B4C6BE2116FC8CEA354AF0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

OK. After a bit of further digging I've found that the CF_AUDIO flag is never set on one of the legs (I'm assuming it's leg A working on the principle that the thread always passes data from channel A to channel B).  As the flag is not set the function calls switch_ivr_sleep() for that leg and that causes the code to write a comfort noise frame.


I've tried changing the flag value in the debugger and I do get the audio stream transmitted to the server (and echoed back) although it's a bit choppy.  Now I need to try and determine where this would normally get set and why it isn't in this case.


Paul

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Paul Mateer <Paul.Mateer at outlook.com>
Sent: 10 January 2017 11:18:27
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] No input audio with FSClient


OK, so I amended the FreeSWITCH.xml for FSClient to add a record_session action before the bridge action of the "number" extension in the default context of the dialplan section, and the client records the input audio stream fine.

I did the same thing in the "echo" extension of the default.xml in the FreeSWITCH dialplan folder and I get a recording of silence. This would seem to fit in with the fact that I don't see any RTP traffic data from the client to the server.


I have logs from both the server and the client (Debug Level set to 9 with SIP Trace active) but there is nothing notable logged between the start of the call and it's termination (I can provide the logs if that's of any help - and a WireShark trace).


Paul


PS. Sorry about the mailing issue yesterday - I think I hit reply to all instead of just reply.


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Mitch Capper <mitch.capper at gmail.com>
Sent: 09 January 2017 20:44:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] No input audio with FSClient

Under options in FSClient for the event socket you can change them but otherwise just connect like normal debugging and sip_trace might be a good start,  you can use fs_logger as well to collect the logs for you if desired.  Let us know the result of the recording test as well.


~mitch

On Mon, Jan 9, 2017 at 9:08 AM, Mitch Capper <mitch.capper at gmail.com<mailto:mitch.capper at gmail.com>> wrote:
Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch itself see if the audio is on the client but not FS.  Enabling logging on the client/server using fs_cli and see anything interesting there.  Finally make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com<mailto:paul.mateer at outlook.com>> wrote:
I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone


_________________________________________________________________________
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>
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From brian at freeswitch.org  Tue Jan 10 18:40:01 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Jan 2017 09:40:01 -0600
Subject: [Freeswitch-users] Freeswitch and JSSip, getting 488
In-Reply-To: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>
References: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>
Message-ID: <CAEJMVkAzCm4E_duoUKG2HBdeR7Xq7=zLtbZPmHC=L8E-7tgbaA@mail.gmail.com>

Without the FreeSWITCH debug log I highly doubt we can answer your question.

/b


On Tue, Jan 10, 2017 at 4:26 AM, Tayeb Meftah <tayeb.meftah at gmail.com>
wrote:

> Hey guys
> it have been a long time;)
> i am having an issue with JSSip and Freeswitch (WSS)
> lovely no signed certificate... ;)
> the issue is i am getting 488
> sip traces:
> http://paste.debian.net/908067/
> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From anthony.minessale at gmail.com  Tue Jan 10 20:22:30 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 10 Jan 2017 11:22:30 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
	<CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
	<CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>
Message-ID: <CAKbxfG9HizcGj4MiFiXU3enO4KdYgUtZVsZj1n1aDdJufhtR_g@mail.gmail.com>

The minute you call it an Issue you should be filing it on JIRA.
We get countless emails a day to the list so I don't always read them all
so you are lucky I have managed to follow this thread.

https://freeswitch.org/jira

We have a small team and dealing with the mailing list is a volunteer
effort.

Here is also a tip.  Just provide the info to questions asked.  I asked you
to do a diagnostic test for me by adding sleep to the other FS.
Regardless if you can change the production or not, its still relevant to
me what happens when you change it.




On Tue, Jan 10, 2017 at 3:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
wrote:

> Hello Anthony,
>
> At this replicated scenario, the box I am calling on SIP is FS.
> But on real scenario, the box I am calling on SIP is not Freeswitch, it is
> an external voicemail server and the initial message have audio cut off.
>
> Thanks for the information about variable ringback, I am already using on
> real scenario.
>
> One strange thing is if I use the codec OPUS at verto, this issue doesn't
> happen.
> But I need to use codec PCMU to avoid audio transcoding.
>
> Let me know if you need more information to debug this issue.
>
> Best Regards,
> Jose Lopes
>
>
>
>
> 2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> So that concludes that media is already established on the webrtc end and
>> there is no problem with that.
>> The box you are calling on SIP is also FS, you may want to add a sleep
>> 2000 in that dialplan before the voicemail.
>> Also since webrtc has no ringing indication you may want to set the
>> variable ringback to get some audible feedback when making calls.
>>
>>
>> On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>> wrote:
>>
>>> Hello Anthony,
>>>
>>> Thanks for your reply.
>>>
>>> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
>>> with ~2 seconds) instead of silence_stream.
>>> When i make the call from verto client, i ear the audio file, then no
>>> audio for ~2/3 seconds and then i ear "id followed by pound" (audio cut
>>> off from voicemail initial message "Please enter your id followed by
>>> pound").
>>>
>>> I checked if i have the variable answer_delay and i don't have it.
>>>
>>> The log of this call is at https://pastebin.freeswitch.org/view/e130e172
>>> .
>>>
>>> There is any thing more that i can do?
>>>
>>>
>>> Best Regards,
>>> Jose Lopes
>>>
>>> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <
>>> anthony.minessale at gmail.com>:
>>>
>>>> Also make sure you don't have answer_delay set in your vars.xml
>>>>
>>>>
>>>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Try making the call with
>>>>>
>>>>> fsctl debug_level 10
>>>>>
>>>>> and observe the logs, answer followed by silence_stream should send
>>>>> audio to the client.
>>>>> Also try playing an audio file instead of silence stream to see if you
>>>>> hear it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <
>>>>> jose.lopes at itcenter.com.pt> wrote:
>>>>>
>>>>>> Hello Brian,
>>>>>>
>>>>>> Thanks for your reply.
>>>>>>
>>>>>> I tried the dialplan bellow with silence_stream://2000, and i have
>>>>>> that issue.
>>>>>> I tried with silence_stream://3000 and the audio cut off is greater.
>>>>>> Without the playback, there is no audio cut off, but FreeSwitch
>>>>>> doesn't send any rtp packets to verto client before the bridge.
>>>>>>
>>>>>> There is any thing more that i can do?
>>>>>>
>>>>>>
>>>>>> <include>
>>>>>>   <context name="default">
>>>>>>     <extension name="call_debug" continue="true">
>>>>>>       <condition field="${call_debug}" expression="^true$"
>>>>>> break="never">
>>>>>>         <action application="info"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>     <extension name="itsp_send_call">
>>>>>>       <condition field="destination_number" expression="^.*$">
>>>>>>         <action application="answer"/>
>>>>>>         <action application="playback" data="silence_stream://2000"/>
>>>>>>         <action application="bridge" data="{absolute_codec_string='
>>>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>   </context>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>> Best Regards,
>>>>>> Jose Lopes
>>>>>>
>>>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>>>
>>>>>>> Prefix them with silence_stream://2000 or 3000 and it should go away.
>>>>>>>
>>>>>>> /b
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> hi,
>>>>>>>>
>>>>>>>> i have the same issue, i think its related to slow audio setup
>>>>>>>> during the call
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Bipin
>>>>>>>>
>>>>>>>>
>>>>>>>> ------------------------------
>>>>>>>> -------- Original Message --------
>>>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto
>>>>>>>> call to sip external voicemail
>>>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>>>
>>>>>>>> Hello Guys,
>>>>>>>>
>>>>>>>> I have audio cut off at the begin of the verto call to  FreeSwitch
>>>>>>>> that redirect to sip external voicemail (Access voicemail mailbox) .
>>>>>>>>
>>>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i
>>>>>>>> use opus at verto codecs, there is no issue, but this causes audio
>>>>>>>> transcoding) .
>>>>>>>>
>>>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>>>>>>>> PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>>>> I notice if i remove the playback action, there is no issue. But I
>>>>>>>> need the playback action to send rtp packets to verto client.
>>>>>>>>
>>>>>>>> I simulate this using another FreeSwitch as external voicemail
>>>>>>>> server and I only listen "id followed by pound" from the initial message of
>>>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>>>> .org/view/507fa115
>>>>>>>>
>>>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type of
>>>>>>>> call?
>>>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>>>
>>>>>>>>
>>>>>>>> Best regards,
>>>>>>>> Jose Lopes
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> 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/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> *Brian West*
>>>>>>> brian at freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>> http://www.freeswitchbook.com
>>>>>>> http://www.freeswitchcookbook.com
>>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>>
>>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>
>>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>>> *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/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-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 <(919)%20386-9900>
>>>>>
>>>>>
>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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 <(919)%20386-9900>
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 <(919)%20386-9900>
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From jose.lopes at itcenter.com.pt  Tue Jan 10 21:27:48 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Tue, 10 Jan 2017 18:27:48 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAKbxfG9HizcGj4MiFiXU3enO4KdYgUtZVsZj1n1aDdJufhtR_g@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
	<CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
	<CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>
	<CAKbxfG9HizcGj4MiFiXU3enO4KdYgUtZVsZj1n1aDdJufhtR_g@mail.gmail.com>
Message-ID: <CAK5rbQs_M5TyTeerb5iZ+RfCKubstdWfh1=gGEHtQWqPbZCqig@mail.gmail.com>

Hello Anthony

Thanks for your reply. I am sorry, I didn't notice that it was a diagnose
test.
I am available to make the tests that you need to analyse this situation
and i put bellow information about the test you ask.
If you see it is better to create a issue on JIRA, i will do it.
Thanks for your and FreeSwitch Team effort.


If I add a sleep of 2000 at voicemail server, after the answer there is no
audio cut off (I put bellow the change on dialplan).
But there is silence of 3/4 seconds between the ivr-say_name and the
initial message from voicemail without audio cut off.

Let me know if you need more tests or information.


Dialplan on FreeSwitch Test:
<include>
  <context name="default">
    <extension name="itsp_send_call">
      <condition field="destination_number" expression="^.*$">
        <action application="answer"/>
        <action application="playback"
data="/usr/share/freeswitch/sounds/en/us/callie/ivr/8000/ivr-say_name.wav"/>
        <action application="set" data="ringback=$${us-ring}"/>
        <action application="bridge"
data="{absolute_codec_string='$${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${destination_number}"/>
      </condition>
    </extension>
  </context>
</include>

Extract of Dialplan on FreeSwitch External Voicemail Server

    <!-- voicemail main extension -->
    <extension name="vmain">
      <condition field="destination_number"
expression="^vmain$|^4000$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="sleep" data="2000"/> <!-- line added -->
        <action application="voicemail" data="check default
${domain_name}"/>
      </condition>
    </extension>


Best Regards,
Jose Lopes




2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> The minute you call it an Issue you should be filing it on JIRA.
> We get countless emails a day to the list so I don't always read them all
> so you are lucky I have managed to follow this thread.
>
> https://freeswitch.org/jira
>
> We have a small team and dealing with the mailing list is a volunteer
> effort.
>
> Here is also a tip.  Just provide the info to questions asked.  I asked
> you to do a diagnostic test for me by adding sleep to the other FS.
> Regardless if you can change the production or not, its still relevant to
> me what happens when you change it.
>
>
>
>
> On Tue, Jan 10, 2017 at 3:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
> wrote:
>
>> Hello Anthony,
>>
>> At this replicated scenario, the box I am calling on SIP is FS.
>> But on real scenario, the box I am calling on SIP is not Freeswitch, it
>> is an external voicemail server and the initial message have audio cut off.
>>
>> Thanks for the information about variable ringback, I am already using on
>> real scenario.
>>
>> One strange thing is if I use the codec OPUS at verto, this issue doesn't
>> happen.
>> But I need to use codec PCMU to avoid audio transcoding.
>>
>> Let me know if you need more information to debug this issue.
>>
>> Best Regards,
>> Jose Lopes
>>
>>
>>
>>
>> 2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> So that concludes that media is already established on the webrtc end
>>> and there is no problem with that.
>>> The box you are calling on SIP is also FS, you may want to add a sleep
>>> 2000 in that dialplan before the voicemail.
>>> Also since webrtc has no ringing indication you may want to set the
>>> variable ringback to get some audible feedback when making calls.
>>>
>>>
>>> On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>>> wrote:
>>>
>>>> Hello Anthony,
>>>>
>>>> Thanks for your reply.
>>>>
>>>> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
>>>> with ~2 seconds) instead of silence_stream.
>>>> When i make the call from verto client, i ear the audio file, then no
>>>> audio for ~2/3 seconds and then i ear "id followed by pound" (audio
>>>> cut off from voicemail initial message "Please enter your id followed by
>>>> pound").
>>>>
>>>> I checked if i have the variable answer_delay and i don't have it.
>>>>
>>>> The log of this call is at https://pastebin.freeswitch
>>>> .org/view/e130e172 .
>>>>
>>>> There is any thing more that i can do?
>>>>
>>>>
>>>> Best Regards,
>>>> Jose Lopes
>>>>
>>>> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <
>>>> anthony.minessale at gmail.com>:
>>>>
>>>>> Also make sure you don't have answer_delay set in your vars.xml
>>>>>
>>>>>
>>>>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> Try making the call with
>>>>>>
>>>>>> fsctl debug_level 10
>>>>>>
>>>>>> and observe the logs, answer followed by silence_stream should send
>>>>>> audio to the client.
>>>>>> Also try playing an audio file instead of silence stream to see if
>>>>>> you hear it.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <
>>>>>> jose.lopes at itcenter.com.pt> wrote:
>>>>>>
>>>>>>> Hello Brian,
>>>>>>>
>>>>>>> Thanks for your reply.
>>>>>>>
>>>>>>> I tried the dialplan bellow with silence_stream://2000, and i have
>>>>>>> that issue.
>>>>>>> I tried with silence_stream://3000 and the audio cut off is greater.
>>>>>>> Without the playback, there is no audio cut off, but FreeSwitch
>>>>>>> doesn't send any rtp packets to verto client before the bridge.
>>>>>>>
>>>>>>> There is any thing more that i can do?
>>>>>>>
>>>>>>>
>>>>>>> <include>
>>>>>>>   <context name="default">
>>>>>>>     <extension name="call_debug" continue="true">
>>>>>>>       <condition field="${call_debug}" expression="^true$"
>>>>>>> break="never">
>>>>>>>         <action application="info"/>
>>>>>>>       </condition>
>>>>>>>     </extension>
>>>>>>>     <extension name="itsp_send_call">
>>>>>>>       <condition field="destination_number" expression="^.*$">
>>>>>>>         <action application="answer"/>
>>>>>>>         <action application="playback" data="silence_stream://2000"/>
>>>>>>>         <action application="bridge" data="{absolute_codec_string='
>>>>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>>>>       </condition>
>>>>>>>     </extension>
>>>>>>>   </context>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>> Best Regards,
>>>>>>> Jose Lopes
>>>>>>>
>>>>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>>>>
>>>>>>>> Prefix them with silence_stream://2000 or 3000 and it should go
>>>>>>>> away.
>>>>>>>>
>>>>>>>> /b
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> hi,
>>>>>>>>>
>>>>>>>>> i have the same issue, i think its related to slow audio setup
>>>>>>>>> during the call
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>> Bipin
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ------------------------------
>>>>>>>>> -------- Original Message --------
>>>>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the
>>>>>>>>> verto call to sip external voicemail
>>>>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>>>>
>>>>>>>>> Hello Guys,
>>>>>>>>>
>>>>>>>>> I have audio cut off at the begin of the verto call to  FreeSwitch
>>>>>>>>> that redirect to sip external voicemail (Access voicemail mailbox) .
>>>>>>>>>
>>>>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i
>>>>>>>>> use opus at verto codecs, there is no issue, but this causes audio
>>>>>>>>> transcoding) .
>>>>>>>>>
>>>>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto)
>>>>>>>>> to PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>>>>> I notice if i remove the playback action, there is no issue. But I
>>>>>>>>> need the playback action to send rtp packets to verto client.
>>>>>>>>>
>>>>>>>>> I simulate this using another FreeSwitch as external voicemail
>>>>>>>>> server and I only listen "id followed by pound" from the initial message of
>>>>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>>>>> .org/view/507fa115
>>>>>>>>>
>>>>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type
>>>>>>>>> of call?
>>>>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Best regards,
>>>>>>>>> Jose Lopes
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> 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/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> *Brian West*
>>>>>>>> brian at freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>>> http://www.freeswitchbook.com
>>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>>>
>>>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>>>>>>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>>
>>>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>>>> *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/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-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
>>>>>> <(919)%20386-9900>
>>>>>>
>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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 <(919)%20386-9900>
>>>>>
>>>>>
>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-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 <(919)%20386-9900>
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tayeb.meftah at gmail.com  Tue Jan 10 21:12:48 2017
From: tayeb.meftah at gmail.com (Tayeb Meftah)
Date: Tue, 10 Jan 2017 19:12:48 +0100
Subject: [Freeswitch-users] Freeswitch and JSSip, getting 488
In-Reply-To: <CAEJMVkAzCm4E_duoUKG2HBdeR7Xq7=zLtbZPmHC=L8E-7tgbaA@mail.gmail.com>
References: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>
	<CAEJMVkAzCm4E_duoUKG2HBdeR7Xq7=zLtbZPmHC=L8E-7tgbaA@mail.gmail.com>
Message-ID: <ED5C663E-375D-423E-A688-14B2B19BEDC7@gmail.com>

Hello Brian
I fixed the issue by adding aply candidate all to my sip profile
I did aply rfc1918.auto
But my Opus codec is OPUS/48000/0
That mean Zero Channel...
Any clue?

Envoy? de mon iPad

> Le 10 janv. 2017 ? 16:40, Brian West <brian at freeswitch.org> a ?crit :
> 
> Without the FreeSWITCH debug log I highly doubt we can answer your question.
> 
> /b
> 
> 
>> On Tue, Jan 10, 2017 at 4:26 AM, Tayeb Meftah <tayeb.meftah at gmail.com> wrote:
>> Hey guys
>> it have been a long time;)
>> i am having an issue with JSSip and Freeswitch (WSS)
>> lovely no signed certificate... ;)
>> the issue is i am getting 488
>> sip traces:
>> http://paste.debian.net/908067/
>> 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
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org
> 
> 
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com 
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
> 
> Got Bugs? Report them here! | Reddit: /r/freeswitch
> 
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> 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 sebastian_ml at gmx.net  Tue Jan 10 21:52:10 2017
From: sebastian_ml at gmx.net (Sebastian Kemper)
Date: Tue, 10 Jan 2017 19:52:10 +0100
Subject: [Freeswitch-users] Gateway outbound configuration
In-Reply-To: <CAMRuXGhoxNDg1MnkHoYkDZ_LKruZ6_6RCSCX-ai8dhxfkPJyNA@mail.gmail.com>
References: <CAMRuXGgvvVtZypyisu0gG6LVAoyUUYfHNjyA2dk-bhjWDQ3EvA@mail.gmail.com>
	<b500757d-b383-2b6d-c43e-e8bd114c75a8@xbipin.com>
	<CAMRuXGiYf00fzrAjgG_z4Utg-5vVYfQyYpELGCSkco=FdTt3og@mail.gmail.com>
	<19116ebc-324a-bcb3-09f7-55ffbc47a5bc@xbipin.com>
	<28899ec6-43d5-ce83-08ef-e8b4571aa224@xbipin.com>
	<CAMRuXGjVe-OXNdFrhJsv8iF+F4gcw5_WokqrsUBOEcfZjt5gkA@mail.gmail.com>
	<d01bc4ba-beaf-a9fc-ed2e-bbfd38b88e85@xbipin.com>
	<CAMRuXGhHRjA7ax0wGFKLvXqwPhiqiUm+EB_5VVbGqytiX6J2uw@mail.gmail.com>
	<CAMRuXGhoxNDg1MnkHoYkDZ_LKruZ6_6RCSCX-ai8dhxfkPJyNA@mail.gmail.com>
Message-ID: <20170110185210.GA3262@wolfgang.lan>

On Mon, Jan 09, 2017 at 11:36:06PM +0530, Murugan Pandian wrote:
> Thanks for your help finally its work bellow is my gateway config
> 
> 
> <include>
>   <gateway name="reliance">
>   <param name="username" value="+91414655252"/>
> 
>   <param name="auth-username" value="+91414655252 at bangalore.relianceims.in"/>
> 
>  <param name="password" value="123456"/>
> 
> <param name="register" value="true"/>
> 
> <param name="realm" value="bangalore.relianceims.in"/>
> 
> <param name="proxy" value="bangalore.relianceims.in"/>
> 
> <param name="regitser-proxy" value="10.237.246.255"/>

Hi Murugan,

There's a typo: regitser-proxy

Regards,
Sebastian


From anthony.minessale at gmail.com  Tue Jan 10 21:59:25 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 10 Jan 2017 12:59:25 -0600
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAK5rbQs_M5TyTeerb5iZ+RfCKubstdWfh1=gGEHtQWqPbZCqig@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
	<CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
	<CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>
	<CAKbxfG9HizcGj4MiFiXU3enO4KdYgUtZVsZj1n1aDdJufhtR_g@mail.gmail.com>
	<CAK5rbQs_M5TyTeerb5iZ+RfCKubstdWfh1=gGEHtQWqPbZCqig@mail.gmail.com>
Message-ID: <CAKbxfG8_4QqXZqjsMFF5nPHGcdpcFzBdSsNqjVO+0eTwAGULFQ@mail.gmail.com>

Yes, please file a JIRA because that was my main concern.  Its very easy
for me to forget about this thread or not notice it was updated.
Once it is in JIRA it serves a place holder for all the info.


On Tue, Jan 10, 2017 at 12:27 PM, Jos? Lopes <jose.lopes at itcenter.com.pt>
wrote:

> Hello Anthony
>
> Thanks for your reply. I am sorry, I didn't notice that it was a diagnose
> test.
> I am available to make the tests that you need to analyse this situation
> and i put bellow information about the test you ask.
> If you see it is better to create a issue on JIRA, i will do it.
> Thanks for your and FreeSwitch Team effort.
>
>
> If I add a sleep of 2000 at voicemail server, after the answer there is no
> audio cut off (I put bellow the change on dialplan).
> But there is silence of 3/4 seconds between the ivr-say_name and the
> initial message from voicemail without audio cut off.
>
> Let me know if you need more tests or information.
>
>
> Dialplan on FreeSwitch Test:
> <include>
>   <context name="default">
>     <extension name="itsp_send_call">
>       <condition field="destination_number" expression="^.*$">
>         <action application="answer"/>
>         <action application="playback" data="/usr/share/freeswitch/
> sounds/en/us/callie/ivr/8000/ivr-say_name.wav"/>
>         <action application="set" data="ringback=$${us-ring}"/>
>         <action application="bridge" data="{absolute_codec_string='
> $${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${
> destination_number}"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Extract of Dialplan on FreeSwitch External Voicemail Server
>
>     <!-- voicemail main extension -->
>     <extension name="vmain">
>       <condition field="destination_number" expression="^vmain$|^4000$|^\*
> 98$">
>         <action application="answer"/>
>         <action application="sleep" data="1000"/>
>         <action application="sleep" data="2000"/> <!-- line added -->
>         <action application="voicemail" data="check default
> ${domain_name}"/>
>       </condition>
>     </extension>
>
>
> Best Regards,
> Jose Lopes
>
>
>
>
> 2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> The minute you call it an Issue you should be filing it on JIRA.
>> We get countless emails a day to the list so I don't always read them all
>> so you are lucky I have managed to follow this thread.
>>
>> https://freeswitch.org/jira
>>
>> We have a small team and dealing with the mailing list is a volunteer
>> effort.
>>
>> Here is also a tip.  Just provide the info to questions asked.  I asked
>> you to do a diagnostic test for me by adding sleep to the other FS.
>> Regardless if you can change the production or not, its still relevant to
>> me what happens when you change it.
>>
>>
>>
>>
>> On Tue, Jan 10, 2017 at 3:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>> wrote:
>>
>>> Hello Anthony,
>>>
>>> At this replicated scenario, the box I am calling on SIP is FS.
>>> But on real scenario, the box I am calling on SIP is not Freeswitch, it
>>> is an external voicemail server and the initial message have audio cut off.
>>>
>>> Thanks for the information about variable ringback, I am already using
>>> on real scenario.
>>>
>>> One strange thing is if I use the codec OPUS at verto, this issue
>>> doesn't happen.
>>> But I need to use codec PCMU to avoid audio transcoding.
>>>
>>> Let me know if you need more information to debug this issue.
>>>
>>> Best Regards,
>>> Jose Lopes
>>>
>>>
>>>
>>>
>>> 2017-01-09 18:25 GMT+00:00 Anthony Minessale <
>>> anthony.minessale at gmail.com>:
>>>
>>>> So that concludes that media is already established on the webrtc end
>>>> and there is no problem with that.
>>>> The box you are calling on SIP is also FS, you may want to add a sleep
>>>> 2000 in that dialplan before the voicemail.
>>>> Also since webrtc has no ringing indication you may want to set the
>>>> variable ringback to get some audible feedback when making calls.
>>>>
>>>>
>>>> On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>> wrote:
>>>>
>>>>> Hello Anthony,
>>>>>
>>>>> Thanks for your reply.
>>>>>
>>>>> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
>>>>> with ~2 seconds) instead of silence_stream.
>>>>> When i make the call from verto client, i ear the audio file, then no
>>>>> audio for ~2/3 seconds and then i ear "id followed by pound" (audio
>>>>> cut off from voicemail initial message "Please enter your id followed by
>>>>> pound").
>>>>>
>>>>> I checked if i have the variable answer_delay and i don't have it.
>>>>>
>>>>> The log of this call is at https://pastebin.freeswitch
>>>>> .org/view/e130e172 .
>>>>>
>>>>> There is any thing more that i can do?
>>>>>
>>>>>
>>>>> Best Regards,
>>>>> Jose Lopes
>>>>>
>>>>> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <
>>>>> anthony.minessale at gmail.com>:
>>>>>
>>>>>> Also make sure you don't have answer_delay set in your vars.xml
>>>>>>
>>>>>>
>>>>>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> Try making the call with
>>>>>>>
>>>>>>> fsctl debug_level 10
>>>>>>>
>>>>>>> and observe the logs, answer followed by silence_stream should send
>>>>>>> audio to the client.
>>>>>>> Also try playing an audio file instead of silence stream to see if
>>>>>>> you hear it.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <
>>>>>>> jose.lopes at itcenter.com.pt> wrote:
>>>>>>>
>>>>>>>> Hello Brian,
>>>>>>>>
>>>>>>>> Thanks for your reply.
>>>>>>>>
>>>>>>>> I tried the dialplan bellow with silence_stream://2000, and i have
>>>>>>>> that issue.
>>>>>>>> I tried with silence_stream://3000 and the audio cut off is
>>>>>>>> greater.
>>>>>>>> Without the playback, there is no audio cut off, but FreeSwitch
>>>>>>>> doesn't send any rtp packets to verto client before the bridge.
>>>>>>>>
>>>>>>>> There is any thing more that i can do?
>>>>>>>>
>>>>>>>>
>>>>>>>> <include>
>>>>>>>>   <context name="default">
>>>>>>>>     <extension name="call_debug" continue="true">
>>>>>>>>       <condition field="${call_debug}" expression="^true$"
>>>>>>>> break="never">
>>>>>>>>         <action application="info"/>
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>     <extension name="itsp_send_call">
>>>>>>>>       <condition field="destination_number" expression="^.*$">
>>>>>>>>         <action application="answer"/>
>>>>>>>>         <action application="playback"
>>>>>>>> data="silence_stream://2000"/>
>>>>>>>>         <action application="bridge" data="{absolute_codec_string='
>>>>>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>>>>>       </condition>
>>>>>>>>     </extension>
>>>>>>>>   </context>
>>>>>>>> </include>
>>>>>>>>
>>>>>>>>
>>>>>>>> Best Regards,
>>>>>>>> Jose Lopes
>>>>>>>>
>>>>>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>>>>>
>>>>>>>>> Prefix them with silence_stream://2000 or 3000 and it should go
>>>>>>>>> away.
>>>>>>>>>
>>>>>>>>> /b
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> hi,
>>>>>>>>>>
>>>>>>>>>> i have the same issue, i think its related to slow audio setup
>>>>>>>>>> during the call
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Regards,
>>>>>>>>>> Bipin
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> ------------------------------
>>>>>>>>>> -------- Original Message --------
>>>>>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the
>>>>>>>>>> verto call to sip external voicemail
>>>>>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>>>>> <freeswitch-users at lists.freeswitch.org>
>>>>>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>>>>>
>>>>>>>>>> Hello Guys,
>>>>>>>>>>
>>>>>>>>>> I have audio cut off at the begin of the verto call to
>>>>>>>>>>  FreeSwitch that redirect to sip external voicemail (Access voicemail
>>>>>>>>>> mailbox) .
>>>>>>>>>>
>>>>>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i
>>>>>>>>>> use opus at verto codecs, there is no issue, but this causes audio
>>>>>>>>>> transcoding) .
>>>>>>>>>>
>>>>>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto)
>>>>>>>>>> to PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>>>>>> I notice if i remove the playback action, there is no issue. But
>>>>>>>>>> I need the playback action to send rtp packets to verto client.
>>>>>>>>>>
>>>>>>>>>> I simulate this using another FreeSwitch as external voicemail
>>>>>>>>>> server and I only listen "id followed by pound" from the initial message of
>>>>>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>>>>>> .org/view/507fa115
>>>>>>>>>>
>>>>>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type
>>>>>>>>>> of call?
>>>>>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Best regards,
>>>>>>>>>> Jose Lopes
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> 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/free
>>>>>>>>>> switch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>>
>>>>>>>>> *Brian West*
>>>>>>>>> brian at freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>>>> http://www.freeswitchbook.com
>>>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>>>>
>>>>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>>>>>>>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>>>
>>>>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>>>>> *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/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-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
>>>>>>> <(919)%20386-9900>
>>>>>>>
>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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
>>>>>> <(919)%20386-9900>
>>>>>>
>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-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 <(919)%20386-9900>
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 <(919)%20386-9900>
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From brian at freeswitch.org  Tue Jan 10 22:44:38 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 10 Jan 2017 13:44:38 -0600
Subject: [Freeswitch-users] Freeswitch and JSSip, getting 488
In-Reply-To: <ED5C663E-375D-423E-A688-14B2B19BEDC7@gmail.com>
References: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>
	<CAEJMVkAzCm4E_duoUKG2HBdeR7Xq7=zLtbZPmHC=L8E-7tgbaA@mail.gmail.com>
	<ED5C663E-375D-423E-A688-14B2B19BEDC7@gmail.com>
Message-ID: <CAEJMVkB=4g3nHBCB6OWXx+kx+2aRf5EmCYryJp8RPocbJyoH9w@mail.gmail.com>

it MUST be /2, even when its not stereo.

/b


On Tue, Jan 10, 2017 at 12:12 PM, Tayeb Meftah <tayeb.meftah at gmail.com>
wrote:

> Hello Brian
> I fixed the issue by adding aply candidate all to my sip profile
> I did aply rfc1918.auto
> But my Opus codec is OPUS/48000/0
> That mean Zero Channel...
> Any clue?
>
> Envoy? de mon iPad
>
> Le 10 janv. 2017 ? 16:40, Brian West <brian at freeswitch.org> a ?crit :
>
> Without the FreeSWITCH debug log I highly doubt we can answer your
> question.
>
> /b
>
>
> On Tue, Jan 10, 2017 at 4:26 AM, Tayeb Meftah <tayeb.meftah at gmail.com>
> wrote:
>
>> Hey guys
>> it have been a long time;)
>> i am having an issue with JSSip and Freeswitch (WSS)
>> lovely no signed certificate... ;)
>> the issue is i am getting 488
>> sip traces:
>> http://paste.debian.net/908067/
>> 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From mike at jerris.com  Tue Jan 10 22:55:10 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 10 Jan 2017 14:55:10 -0500
Subject: [Freeswitch-users] Freeswitch and JSSip, getting 488
In-Reply-To: <CAEJMVkB=4g3nHBCB6OWXx+kx+2aRf5EmCYryJp8RPocbJyoH9w@mail.gmail.com>
References: <CAJySFWL+M1rd=v7f9xadJARuhMrsAfTQP65Z2otvzxZaQNk1XQ@mail.gmail.com>
	<CAEJMVkAzCm4E_duoUKG2HBdeR7Xq7=zLtbZPmHC=L8E-7tgbaA@mail.gmail.com>
	<ED5C663E-375D-423E-A688-14B2B19BEDC7@gmail.com>
	<CAEJMVkB=4g3nHBCB6OWXx+kx+2aRf5EmCYryJp8RPocbJyoH9w@mail.gmail.com>
Message-ID: <707552D5-2E32-4D49-B837-157FBAF7CA19@jerris.com>

try using sip.js.  Every person i?ve seen try to use jssip has failed due to outstanding bugs in jssip.  Even better, unless there is a compelling reason to use sip in the browser, use mod_verto.


> On Jan 10, 2017, at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
> 
> it MUST be /2, even when its not stereo.
> 
> /b
> 
> 
> On Tue, Jan 10, 2017 at 12:12 PM, Tayeb Meftah <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
> Hello Brian
> I fixed the issue by adding aply candidate all to my sip profile
> I did aply rfc1918.auto
> But my Opus codec is OPUS/48000/0
> That mean Zero Channel...
> Any clue?
> 
> Envoy? de mon iPad
> 
> Le 10 janv. 2017 ? 16:40, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> a ?crit :
> 
>> Without the FreeSWITCH debug log I highly doubt we can answer your question.
>> 
>> /b
>> 
>> 
>> On Tue, Jan 10, 2017 at 4:26 AM, Tayeb Meftah <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
>> Hey guys
>> it have been a long time;)
>> i am having an issue with JSSip and Freeswitch (WSS)
>> lovely no signed certificate... ;)
>> the issue is i am getting 488
>> sip traces:
>> http://paste.debian.net/908067/ <http://paste.debian.net/908067/>
>> thanks!
>> 
>> _________________________________________________________________________
>> 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/>
>> https://www.gofundme.com/freeswitch_ubuntu <https://www.gofundme.com/freeswitch_ubuntu>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>> T:+19184209001 <tel:(918)%20420-9001> | F:+19184209002 <tel:(918)%20420-9002> | M:+1918424WEST (9378)
>> Skype:briankwest
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <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/>
> https://www.gofundme.com/freeswitch_ubuntu <https://www.gofundme.com/freeswitch_ubuntu>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> 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 david.villasmil.work at gmail.com  Wed Jan 11 05:45:44 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 10 Jan 2017 21:45:44 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
Message-ID: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>

hello guys,

i have 2 profiles for two different ip addresses (IP1 and IP2), both
profiles have force-register-domain and force-register-db-domain to the
IP1, i can register on IP2 but when calling this user like user at IP1, fs
returns USER_NOT_REGISTERED... i thought by forcing the register domain i
could call the user to any of the 2 ips... any ideas?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
?
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From gascagonzalo at gmail.com  Wed Jan 11 09:07:19 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Tue, 10 Jan 2017 22:07:19 -0800
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
In-Reply-To: <CAL+DTgLaExbYvomM+W-TQyjf7u5fnO=wjx22-H_QKKUyqZ7rmQ@mail.gmail.com>
References: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
	<bc37c753-0953-567f-57f7-077e94478df0@gmail.com>
	<CAL+DTgLaExbYvomM+W-TQyjf7u5fnO=wjx22-H_QKKUyqZ7rmQ@mail.gmail.com>
Message-ID: <CAL+DTg+4eBpfjPyyqKOiMymg5KMXoE5a0uvq8S8CUKoOvPPtmg@mail.gmail.com>

I renamed @confidence to confidence helped solving this specific issue but
still getting errors when reading result.interpretation.

2017-01-11 00:52:52.739773 [DEBUG] mod_pocketsphinx.c:464 Recognized:
takeout, Confidence: 99, Confidence-Threshold: 0
2017-01-11 00:52:52.759764 [DEBUG] SpeechTools.jm:188 ----XML:

<result grammar="pizza_order">
  <interpretation grammar="pizza_order" confidence="99">
    <input mode="speech">takeout</input>
  </interpretation>
</result>

2017-01-11 00:52:52.759764 [CRIT] SpeechTools.jm:214 ----ERROR:
TypeError: Cannot read property 'interpretation' of undefined
2017-01-11 00:52:55.559759 [NOTICE] sofia.c:1011 Hangup sofia/external/
14088053951 at 104.236.190.206 [CS_EXECUTE] [NORMAL_CLEARING]



On Tue, Jan 10, 2017 at 12:37 AM, Gonzalo Gasca Meza <gascagonzalo at gmail.com
> wrote:

> Code is from sample Grammar provided by Wiki, not my original code...
>
> On Tue, Jan 10, 2017 at 12:32 AM, ????? <findmeinwland at gmail.com> wrote:
>
>> But it's a syntax error, no? I think you need to verify your code
>>
>>
>> 10.01.2017 13:26, Gonzalo Gasca Meza ?????:
>> > Hi all,
>> >
>> > I'm trying to install Speech Recognition engine.
>> >
>> > Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)
>> >
>> > Following this example:
>> >
>> > https://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>> >
>> > <include>
>> >     <extension name="pizza_demo">
>> >       <condition field="destination_number"
>> expression="^(pizza|74992)$"/>
>> >       <condition field="${module_exists(mod_v8)}" expression="true"/>
>> >       <condition field="${module_exists(mod_pocketsphinx)}"
>> > expression="true">
>> >        <action application="javascript" data="ps_pizza.js"/>
>> >       </condition>
>> >     </extension>
>> > </include>
>> >
>> > I'm replacing mod_spidermonkey with mod_v8 based on this document and
>> > when I call this number I get:
>> >
>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>> > javascript(ps_pizza.js)
>> > 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception:
>> > SyntaxError: Unexpected identifier (near: "
>>        if
>> > (result.interpretation. at confidence >= grammar_object.min_score) {")
>> >
>> > If I change: result.interpretation. at confidence to
>> > result.interpretation.['@confidence']
>> >
>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>> > export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
>> > 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT
>> > (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>> > javascript(ps_pizza.js)
>> > 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception:
>> > SyntaxError: Unexpected string (near: "
>>        if
>> > (result.interpretation.['@confidence'] >= grammar_object.min_score) {")
>> >
>> > SpeechTools.jm from:
>> > https://raw.githubusercontent.com/lordnull/FreeSWITCH/master
>> /scripts/javascript/js_modules/SpeechTools.jm
>> >
>> > Similar discussion:
>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2014-
>> June/105818.html
>> >
>> > Any pointers?
>> > I get the mod_v8 module compiled from Freeswitch using this:
>> > https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>> >
>> > 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 findmeinwland at gmail.com  Wed Jan 11 09:10:59 2017
From: findmeinwland at gmail.com (=?UTF-8?B?0JDRgNGC0YPRgA==?=)
Date: Wed, 11 Jan 2017 11:10:59 +0500
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
In-Reply-To: <CAL+DTg+4eBpfjPyyqKOiMymg5KMXoE5a0uvq8S8CUKoOvPPtmg@mail.gmail.com>
References: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
	<bc37c753-0953-567f-57f7-077e94478df0@gmail.com>
	<CAL+DTgLaExbYvomM+W-TQyjf7u5fnO=wjx22-H_QKKUyqZ7rmQ@mail.gmail.com>
	<CAL+DTg+4eBpfjPyyqKOiMymg5KMXoE5a0uvq8S8CUKoOvPPtmg@mail.gmail.com>
Message-ID: <64c940e3-c77b-d2f6-e105-ff2cf91384ce@gmail.com>

You can debug it by yourself, print to console different values. Explore 
the code.


11.01.2017 11:07, Gonzalo Gasca Meza ?????:
> I renamed @confidence to confidence helped solving this specific issue 
> but still getting errors when reading result.interpretation.
>
> 2017-01-11 00:52:52.739773 [DEBUG] mod_pocketsphinx.c:464 Recognized: 
> takeout, Confidence: 99, Confidence-Threshold: 0
> 2017-01-11 00:52:52.759764 [DEBUG] SpeechTools.jm:188 ----XML:
>
> <result grammar="pizza_order">
>   <interpretation grammar="pizza_order" confidence="99">
>     <input mode="speech">takeout</input>
>   </interpretation>
> </result>
>
> 2017-01-11 00:52:52.759764 [CRIT] SpeechTools.jm:214 ----ERROR:
> TypeError: Cannot read property 'interpretation' of undefined
> 2017-01-11 00:52:55.559759 [NOTICE] sofia.c:1011 Hangup 
> sofia/external/14088053951 at 104.236.190.206 
> <mailto:14088053951 at 104.236.190.206> [CS_EXECUTE] [NORMAL_CLEARING]
>
>
>
> On Tue, Jan 10, 2017 at 12:37 AM, Gonzalo Gasca Meza 
> <gascagonzalo at gmail.com <mailto:gascagonzalo at gmail.com>> wrote:
>
>     Code is from sample Grammar provided by Wiki, not my original code...
>
>     On Tue, Jan 10, 2017 at 12:32 AM, ????? <findmeinwland at gmail.com
>     <mailto:findmeinwland at gmail.com>> wrote:
>
>         But it's a syntax error, no? I think you need to verify your code
>
>
>         10.01.2017 13:26, Gonzalo Gasca Meza ?????:
>         > Hi all,
>         >
>         > I'm trying to install Speech Recognition engine.
>         >
>         > Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)
>         >
>         > Following this example:
>         >
>         > https://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>         <https://wiki.freeswitch.org/wiki/Mod_pocketsphinx>
>         >
>         > <include>
>         >     <extension name="pizza_demo">
>         >       <condition field="destination_number"
>         expression="^(pizza|74992)$"/>
>         >       <condition field="${module_exists(mod_v8)}"
>         expression="true"/>
>         >       <condition field="${module_exists(mod_pocketsphinx)}"
>         > expression="true">
>         >        <action application="javascript" data="ps_pizza.js"/>
>         >       </condition>
>         >     </extension>
>         > </include>
>         >
>         > I'm replacing mod_spidermonkey with mod_v8 based on this
>         document and
>         > when I call this number I get:
>         >
>         > EXECUTE sofia/external/14088053951 at 104.236.190.206
>         <mailto:14088053951 at 104.236.190.206>
>         > javascript(ps_pizza.js)
>         > 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception:
>         > SyntaxError: Unexpected identifier (near: "                 
>                          if
>         > (result.interpretation. at confidence >=
>         grammar_object.min_score) {")
>         >
>         > If I change: result.interpretation. at confidence to
>         > result.interpretation.['@confidence']
>         >
>         > EXECUTE sofia/external/14088053951 at 104.236.190.206
>         <mailto:14088053951 at 104.236.190.206>
>         > export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
>         > 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT
>         > (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
>         > EXECUTE sofia/external/14088053951 at 104.236.190.206
>         <mailto:14088053951 at 104.236.190.206>
>         > javascript(ps_pizza.js)
>         > 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception:
>         > SyntaxError: Unexpected string (near: "                    
>                          if
>         > (result.interpretation.['@confidence'] >=
>         grammar_object.min_score) {")
>         >
>         > SpeechTools.jm from:
>         >
>         https://raw.githubusercontent.com/lordnull/FreeSWITCH/master/scripts/javascript/js_modules/SpeechTools.jm
>         <https://raw.githubusercontent.com/lordnull/FreeSWITCH/master/scripts/javascript/js_modules/SpeechTools.jm>
>         >
>         > Similar discussion:
>         >
>         http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105818.html
>         <http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105818.html>
>         >
>         > Any pointers?
>         > I get the mod_v8 module compiled from Freeswitch using this:
>         > https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>         <https://freeswitch.org/confluence/display/FREESWITCH/mod_v8>
>         >
>         > Thanks
>         >
>         >
>         >
>         >
>         >
>         _________________________________________________________________________
>         > 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://confluence.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
>         <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
>
>
>         _________________________________________________________________________
>         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://confluence.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
>         <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
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gascagonzalo at gmail.com  Wed Jan 11 10:48:02 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Tue, 10 Jan 2017 23:48:02 -0800
Subject: [Freeswitch-users] Speech recognition SpeechTools.jm Error
In-Reply-To: <64c940e3-c77b-d2f6-e105-ff2cf91384ce@gmail.com>
References: <c4e68da1a4f48272b4aa1dbe239ce136@parzee.com>
	<bc37c753-0953-567f-57f7-077e94478df0@gmail.com>
	<CAL+DTgLaExbYvomM+W-TQyjf7u5fnO=wjx22-H_QKKUyqZ7rmQ@mail.gmail.com>
	<CAL+DTg+4eBpfjPyyqKOiMymg5KMXoE5a0uvq8S8CUKoOvPPtmg@mail.gmail.com>
	<64c940e3-c77b-d2f6-e105-ff2cf91384ce@gmail.com>
Message-ID: <CAL+DTg+mQq60Ay+fe2Z4frO6paS2-AOOKtQO3G896m+DAQersQ@mail.gmail.com>

Sigh...


Need to confirm if this SpeechRecognition.jm script needs to be migrated to
XML handling based on this note:

"The new V8 module relies on XML support built into FS, since Google V8
doesn't have this built-in like Spidermonkey."
https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+example+-+XML
http://stackoverflow.com/questions/38846047/freeswitch-speech-recognition-speechtools-jm-error?rq=1

Seems to be that handling XML

<result grammar="pizza_order">
  <interpretation grammar="pizza_order" confidence="100">
    <input mode="speech">pickup</input>
  </interpretation>
</result>

Require use of XML - use('XML'); as is not parsed by default, and original
.jm file may not work just using the original code.

Thanks

-Gonzalo

On Tue, Jan 10, 2017 at 10:10 PM, ????? <findmeinwland at gmail.com> wrote:

> You can debug it by yourself, print to console different values. Explore
> the code.
>
>
> 11.01.2017 11:07, Gonzalo Gasca Meza ?????:
>
> I renamed @confidence to confidence helped solving this specific issue
> but still getting errors when reading result.interpretation.
>
> 2017-01-11 00:52:52.739773 [DEBUG] mod_pocketsphinx.c:464 Recognized:
> takeout, Confidence: 99, Confidence-Threshold: 0
> 2017-01-11 00:52:52.759764 [DEBUG] SpeechTools.jm:188 ----XML:
>
> <result grammar="pizza_order">
>   <interpretation grammar="pizza_order" confidence="99">
>     <input mode="speech">takeout</input>
>   </interpretation>
> </result>
>
> 2017-01-11 00:52:52.759764 [CRIT] SpeechTools.jm:214 ----ERROR:
> TypeError: Cannot read property 'interpretation' of undefined
> 2017-01-11 00:52:55.559759 [NOTICE] sofia.c:1011 Hangup sofia/external/
> 14088053951 at 104.236.190.206 [CS_EXECUTE] [NORMAL_CLEARING]
>
>
>
> On Tue, Jan 10, 2017 at 12:37 AM, Gonzalo Gasca Meza <
> gascagonzalo at gmail.com> wrote:
>
>> Code is from sample Grammar provided by Wiki, not my original code...
>>
>> On Tue, Jan 10, 2017 at 12:32 AM, ????? <findmeinwland at gmail.com> wrote:
>>
>>> But it's a syntax error, no? I think you need to verify your code
>>>
>>>
>>> 10.01.2017 13:26, Gonzalo Gasca Meza ?????:
>>> > Hi all,
>>> >
>>> > I'm trying to install Speech Recognition engine.
>>> >
>>> > Version: FreeSWITCH (Version 1.6.13 -21-e755b43 64bit)
>>> >
>>> > Following this example:
>>> >
>>> > https://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>>> >
>>> > <include>
>>> >     <extension name="pizza_demo">
>>> >       <condition field="destination_number"
>>> expression="^(pizza|74992)$"/>
>>> >       <condition field="${module_exists(mod_v8)}" expression="true"/>
>>> >       <condition field="${module_exists(mod_pocketsphinx)}"
>>> > expression="true">
>>> >        <action application="javascript" data="ps_pizza.js"/>
>>> >       </condition>
>>> >     </extension>
>>> > </include>
>>> >
>>> > I'm replacing mod_spidermonkey with mod_v8 based on this document and
>>> > when I call this number I get:
>>> >
>>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>>> > javascript(ps_pizza.js)
>>> > 2017-01-10 03:04:39.099748 [ERR] SpeechTools.jm:191 Exception:
>>> > SyntaxError: Unexpected identifier (near: "
>>>        if
>>> > (result.interpretation. at confidence >= grammar_object.min_score) {")
>>> >
>>> > If I change: result.interpretation. at confidence to
>>> > result.interpretation.['@confidence']
>>> >
>>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>>> > export(RFC2822_DATE=Tue, 10 Jan 2017 03:13:10 -0500)
>>> > 2017-01-10 03:13:10.139816 [DEBUG] switch_channel.c:1296 EXPORT
>>> > (export_vars) [RFC2822_DATE]=[Tue, 10 Jan 2017 03:13:10 -0500]
>>> > EXECUTE sofia/external/14088053951 at 104.236.190.206
>>> > javascript(ps_pizza.js)
>>> > 2017-01-10 03:13:10.159763 [ERR] SpeechTools.jm:191 Exception:
>>> > SyntaxError: Unexpected string (near: "
>>>        if
>>> > (result.interpretation.['@confidence'] >= grammar_object.min_score)
>>> {")
>>> >
>>> > SpeechTools.jm from:
>>> > https://raw.githubusercontent.com/lordnull/FreeSWITCH/master
>>> /scripts/javascript/js_modules/SpeechTools.jm
>>> >
>>> > Similar discussion:
>>> > http://lists.freeswitch.org/pipermail/freeswitch-users/2014-
>>> June/105818.html
>>> >
>>> > Any pointers?
>>> > I get the mod_v8 module compiled from Freeswitch using this:
>>> > https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>>> >
>>> > 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/free
>>> switch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asilva at wirelessmundi.com  Wed Jan 11 13:38:00 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Wed, 11 Jan 2017 11:38:00 +0100
Subject: [Freeswitch-users] Postgresql in fs core: recommendations
Message-ID: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>

Hi,

On debian jessie, i've switch from using sqlite in the core with db 
located  at /dev/shm to postgresql server running on the same box, but 
now for the first time, i see SIP messages in FS "503 Server Busy" and 
losing endpoint registration with expire, i think the delay added by pg 
to process the queries affects fs main process to handle huge number of 
sip messages per second.

In terms of fs i have:
- call average 6 cps
- avg 100 registers / sec (the expire value configure in the accounts is 
of 300 seconds)
- avg 150 options / sec (max of 400 pkt/sec)
- avg 50 notify / sec (max of 100 pkt/sec)


In fs configurations i've:
max-db-handles = 100
max-sessions = 2000
sessions-per-second = 50


My server i think is quite power to handle this traffic:

2 x Intel(R) Xeon(R) CPU E5-2620 v4 @ 2.10GHz
32GB of ram
sata disk 7200RPM


I was wonder if any of you have this issues and if you could share your 
pg configuration?

For now i i've try for pg:

max_connections = 100
ssl = false
shared_buffers = 5GB
dynamic_shared_memory_type = posix


Thanks,

-- 

Saludos / Regards / Cumprimentos,
Ant?nio silva



From Paul.Mateer at outlook.com  Wed Jan 11 16:39:06 2017
From: Paul.Mateer at outlook.com (Paul Mateer)
Date: Wed, 11 Jan 2017 13:39:06 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <AM5PR10MB07088392B8B5A7EB68F37B09F0660@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

OK. I'm wondering if this is a possible bug in the FreeSWITCH code?

I see two threads running - each with it's own session and channel. One is an outbound channel called "sofia/softphone/9196", whilst the other is the inbound channel called "portaudio/sofia/gateway/2/9196".


Now the CF_AUDIO flag is set on the inbound channel during a call to switch_core_media_negotiate_sdp(), but as far as I can tell the outbound channel never gets its CF_AUDIO flag set, which is why there's no audio transmitted. I can correct the problem by adding the following call to the if(channel_caller) statement block at the end of switch_ivr_originate() method:


    switch_channel_set_flag(caller_channel, CF_AUDIO);


but as I only started working with FreeSWITCH before Christmas, I'm not sure if this is a definite issue and whether this is an appropriate correction.


Does anyone have any thoughts on this?


Paul

________________________________
From: Paul Mateer <paul.mateer at outlook.com>
Sent: 09 January 2017 16:58:53
To: freeswitch-users at lists.freeswitch.org
Subject: No input audio with FSClient

I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone

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From asilva at wirelessmundi.com  Wed Jan 11 17:30:35 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Wed, 11 Jan 2017 15:30:35 +0100
Subject: [Freeswitch-users] Postgresql in fs core: recommendations
In-Reply-To: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>
References: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>
Message-ID: <c7bcd72b-b6e2-637a-f545-d7d0141e4ac4@wirelessmundi.com>

Just hit this link, where it explains how to put pg in ram:

http://www.manniwood.com/postgresql_94_in_ram/index.html

I guess i could try this to have similar behavior as sqlite.

On 01/11/2017 11:38 AM, Antonio Silva wrote:
> Hi,
>
> On debian jessie, i've switch from using sqlite in the core with db
> located  at /dev/shm to postgresql server running on the same box, but
> now for the first time, i see SIP messages in FS "503 Server Busy" and
> losing endpoint registration with expire, i think the delay added by pg
> to process the queries affects fs main process to handle huge number of
> sip messages per second.
>
> In terms of fs i have:
> - call average 6 cps
> - avg 100 registers / sec (the expire value configure in the accounts is
> of 300 seconds)
> - avg 150 options / sec (max of 400 pkt/sec)
> - avg 50 notify / sec (max of 100 pkt/sec)
>
>
> In fs configurations i've:
> max-db-handles = 100
> max-sessions = 2000
> sessions-per-second = 50
>
>
> My server i think is quite power to handle this traffic:
>
> 2 x Intel(R) Xeon(R) CPU E5-2620 v4 @ 2.10GHz
> 32GB of ram
> sata disk 7200RPM
>
>
> I was wonder if any of you have this issues and if you could share your
> pg configuration?
>
> For now i i've try for pg:
>
> max_connections = 100
> ssl = false
> shared_buffers = 5GB
> dynamic_shared_memory_type = posix
>
>
> Thanks,
>

-- 

Saludos / Regards / Cumprimentos,
Ant?nio silva



From mike at jerris.com  Wed Jan 11 18:17:23 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 11 Jan 2017 10:17:23 -0500
Subject: [Freeswitch-users] Postgresql in fs core: recommendations
In-Reply-To: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>
References: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>
Message-ID: <009DED95-6EAA-4117-B58E-62C7756FD49C@jerris.com>

might need to play w/ doing register in threads.  There is a profile param for this.

> On Jan 11, 2017, at 5:38 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:
> 
> Hi,
> 
> On debian jessie, i've switch from using sqlite in the core with db 
> located  at /dev/shm to postgresql server running on the same box, but 
> now for the first time, i see SIP messages in FS "503 Server Busy" and 
> losing endpoint registration with expire, i think the delay added by pg 
> to process the queries affects fs main process to handle huge number of 
> sip messages per second.
> 
> In terms of fs i have:
> - call average 6 cps
> - avg 100 registers / sec (the expire value configure in the accounts is 
> of 300 seconds)
> - avg 150 options / sec (max of 400 pkt/sec)
> - avg 50 notify / sec (max of 100 pkt/sec)
> 
> 
> In fs configurations i've:
> max-db-handles = 100
> max-sessions = 2000
> sessions-per-second = 50
> 
> 
> My server i think is quite power to handle this traffic:
> 
> 2 x Intel(R) Xeon(R) CPU E5-2620 v4 @ 2.10GHz
> 32GB of ram
> sata disk 7200RPM
> 
> 
> I was wonder if any of you have this issues and if you could share your 
> pg configuration?
> 
> For now i i've try for pg:
> 
> max_connections = 100
> ssl = false
> shared_buffers = 5GB
> dynamic_shared_memory_type = posix
> 
> 
> Thanks,
> 
> -- 
> 
> Saludos / Regards / Cumprimentos,
> Ant?nio silva
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From asilva at wirelessmundi.com  Wed Jan 11 18:47:50 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Wed, 11 Jan 2017 16:47:50 +0100
Subject: [Freeswitch-users] Postgresql in fs core: recommendations
In-Reply-To: <009DED95-6EAA-4117-B58E-62C7756FD49C@jerris.com>
References: <7a447c0d-e1ba-1c9e-92c3-5bb2a01ee111@wirelessmundi.com>
	<009DED95-6EAA-4117-B58E-62C7756FD49C@jerris.com>
Message-ID: <3031c25b-5a97-05cb-7735-be7bd6bf8c38@wirelessmundi.com>

Hi, yes already set it:

<param name="max-reg-threads" value="8"/>
<param name="inbound-reg-in-new-thread" value="true"/>


The only change here is from sqlite to postgresql, for sure that is some 
tunning in pg to handle this volume of traffic... i'm still trying to 
figure it out.


On 01/11/2017 04:17 PM, Michael Jerris wrote:
> might need to play w/ doing register in threads.  There is a profile param for this.
>
>> On Jan 11, 2017, at 5:38 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:
>>
>> Hi,
>>
>> On debian jessie, i've switch from using sqlite in the core with db
>> located  at /dev/shm to postgresql server running on the same box, but
>> now for the first time, i see SIP messages in FS "503 Server Busy" and
>> losing endpoint registration with expire, i think the delay added by pg
>> to process the queries affects fs main process to handle huge number of
>> sip messages per second.
>>
>> In terms of fs i have:
>> - call average 6 cps
>> - avg 100 registers / sec (the expire value configure in the accounts is
>> of 300 seconds)
>> - avg 150 options / sec (max of 400 pkt/sec)
>> - avg 50 notify / sec (max of 100 pkt/sec)
>>
>>
>> In fs configurations i've:
>> max-db-handles = 100
>> max-sessions = 2000
>> sessions-per-second = 50
>>
>>
>> My server i think is quite power to handle this traffic:
>>
>> 2 x Intel(R) Xeon(R) CPU E5-2620 v4 @ 2.10GHz
>> 32GB of ram
>> sata disk 7200RPM
>>
>>
>> I was wonder if any of you have this issues and if you could share your
>> pg configuration?
>>
>> For now i i've try for pg:
>>
>> max_connections = 100
>> ssl = false
>> shared_buffers = 5GB
>> dynamic_shared_memory_type = posix
>>
>>
>> Thanks,
>>
>> -- 
>>
>> Saludos / Regards / Cumprimentos,
>> Ant?nio silva
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 

Saludos / Regards / Cumprimentos,
Ant?nio silva



From mitch.capper at gmail.com  Wed Jan 11 21:09:14 2017
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 11 Jan 2017 10:09:14 -0800
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
	<CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>
	<AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CALn233qy3H4Jmx=e3Vemb4n70UN5ZosB_jOnfBKUVHjDCtDjJA@mail.gmail.com>

Can you post the logs to the FS pastbin that may be the easiest way to look
at them.  fs_logger can anonymize the logs if needed.  Also we should
probably get build versions and the logs should give us the codec settings.


~mitch

On Tue, Jan 10, 2017 at 3:18 AM, Paul Mateer <Paul.Mateer at outlook.com>
wrote:

> OK, so I amended the FreeSWITCH.xml for FSClient to add a record_session
> action before the bridge action of the "number" extension in the default
> context of the dialplan section, and the client records the input audio
> stream fine.
>
> I did the same thing in the "echo" extension of the default.xml in the
> FreeSWITCH dialplan folder and I get a recording of silence. This would
> seem to fit in with the fact that I don't see any RTP traffic data from the
> client to the server.
>
>
> I have logs from both the server and the client (Debug Level set to 9 with
> SIP Trace active) but there is nothing notable logged between the start of
> the call and it's termination (I can provide the logs if that's of any help
> - and a WireShark trace).
>
>
> Paul
>
>
> PS. Sorry about the mailing issue yesterday - I think I hit reply to all
> instead of just reply.
>
>
> ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Mitch Capper <
> mitch.capper at gmail.com>
> *Sent:* 09 January 2017 20:44:53
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] No input audio with FSClient
>
> Under options in FSClient for the event socket you can change them but
> otherwise just connect like normal debugging and sip_trace might be a good
> start,  you can use fs_logger as well to collect the logs for you if
> desired.  Let us know the result of the recording test as well.
>
>
> ~mitch
>
> On Mon, Jan 9, 2017 at 9:08 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
>
>> Hi Paul,
>> One place to start would be to record the call with FSClient and
>> Freeswitch itself see if the audio is on the client but not FS.  Enabling
>> logging on the client/server using fs_cli and see anything interesting
>> there.  Finally make sure you have the right mic input selected in the
>> FSClient options.
>>
>>
>>
>>
>> ~mitch
>>
>> On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com>
>> wrote:
>>
>>> I seem to have a problem with audio input when using FSClient.
>>>
>>>
>>>
>>> I have one box running FreeSWITCH and another running FSClient. I can
>>> call the server no problem and get audio back (I dialled 9198 to get the
>>> Tetris tune) but when I provide an audio stream using the mic and dial 9196
>>> I get nothing back.
>>>
>>>
>>>
>>> I know the mic is providing sound and the FreeSWITCH server is operating
>>> OK because I can use X-Lite to perform the same test and I get the audio
>>> feed played back to me.
>>>
>>>
>>>
>>> I'm not sure if there is something amiss in the configuration of
>>> FSClient (although it should be the default config) or if something else is
>>> amiss (there doesn't appear to be anything odd in the FreeSWITCH log for
>>> the client).
>>>
>>>
>>>
>>> Does anyone have any thoughts on what might be wrong, or what i should
>>> look at?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Paul
>>>
>>>
>>>
>>> Sent from my Windows 10 phone
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Thu Jan 12 03:30:14 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 11 Jan 2017 19:30:14 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
Message-ID: <CAFGRPVqWfY5C4GOE7EHTRJdQ_6SKp5yycEWLiEyoms+hb0wBkA@mail.gmail.com>

Anyone?
?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337

On Tue, Jan 10, 2017 at 9:45 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> hello guys,
>
> i have 2 profiles for two different ip addresses (IP1 and IP2), both
> profiles have force-register-domain and force-register-db-domain to the
> IP1, i can register on IP2 but when calling this user like user at IP1, fs
> returns USER_NOT_REGISTERED... i thought by forcing the register domain i
> could call the user to any of the 2 ips... any ideas?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
> ?
>
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From yehavi.bourvine at gmail.com  Thu Jan 12 07:44:21 2017
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 12 Jan 2017 06:44:21 +0200
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
Message-ID: <CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>

Can you post your bridge command call? Do you use sofia_contact() to build
the dial string?

              Regards, __Yehavi:

2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:

> hello guys,
>
> i have 2 profiles for two different ip addresses (IP1 and IP2), both
> profiles have force-register-domain and force-register-db-domain to the
> IP1, i can register on IP2 but when calling this user like user at IP1, fs
> returns USER_NOT_REGISTERED... i thought by forcing the register domain i
> could call the user to any of the 2 ips... any ideas?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vivek at advaitamtech.com  Thu Jan 12 08:20:08 2017
From: vivek at advaitamtech.com (vivek at advaitamtech.com)
Date: Thu, 12 Jan 2017 10:50:08 +0530 (IST)
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
	Authentication Required"
In-Reply-To: <1483957702.686832416@apps.rackspace.com>
References: <1483957702.686832416@apps.rackspace.com>
Message-ID: <1484198408.261517009@apps.rackspace.com>


Hi, 
 
Any help on this?
 
 
-----Original Message-----
From: "vivek at advaitamtech.com" <vivek at advaitamtech.com>
Sent: Monday, 9 January, 2017 3:58pm
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy Authentication Required"



Hi All,
 
Am trying to proxy the "407 Proxy Authentication Required" through freeswitch,
 
My Objective is to proxy the "407 Proxy Authentication Required" received from the third party server to another server which is connected to freeswitch.
 
Is it possible to do this?. If possible how do I do it. Please help me out.
 
Thanks,
Vivek.
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From s.safarov at gmail.com  Thu Jan 12 08:50:27 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 12 Jan 2017 05:50:27 +0000
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
 Authentication Required"
In-Reply-To: <1484198408.261517009@apps.rackspace.com>
References: <1483957702.686832416@apps.rackspace.com>
	<1484198408.261517009@apps.rackspace.com>
Message-ID: <CAHtxdDfUuYe8zW5XCHNKKsu=ia7tbW_3PdDmQQ7i=+24GfpsiA@mail.gmail.com>

No FreeSwith is not designed to do it.
You can use kamailio

??, 12 ???. 2017, 8:20 vivek at advaitamtech.com <vivek at advaitamtech.com>:

> Hi,
>
>
>
> Any help on this?
>
>
>
>
>
> -----Original Message-----
> From: "vivek at advaitamtech.com" <vivek at advaitamtech.com>
> Sent: Monday, 9 January, 2017 3:58pm
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
> Authentication Required"
>
> Hi All,
>
>
>
> Am trying to proxy the "407 Proxy Authentication Required" through
> freeswitch,
>
>
>
> My Objective is to proxy the "407 Proxy Authentication Required" received
> from the third party server to another server which is connected to
> freeswitch.
>
>
>
> Is it possible to do this?. If possible how do I do it. Please help me out.
>
>
>
> Thanks,
>
> Vivek.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rtreleaven at bunnykick.ca  Thu Jan 12 08:52:01 2017
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Thu, 12 Jan 2017 00:52:01 -0500
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
 Authentication Required"
In-Reply-To: <1484198408.261517009@apps.rackspace.com>
References: <1483957702.686832416@apps.rackspace.com>
	<1484198408.261517009@apps.rackspace.com>
Message-ID: <CAHzfiZ5+sJKX6tgkqKxVSMfg2PkzPf0CPtYisMsAHC7hAbXP9Q@mail.gmail.com>

Freeswitch is not a proxy it's a b2bua. Google Sofia gateway, it might be
what you need.

On Jan 12, 2017 12:20 AM, "vivek at advaitamtech.com" <vivek at advaitamtech.com>
wrote:

> Hi,
>
>
>
> Any help on this?
>
>
>
>
>
> -----Original Message-----
> From: "vivek at advaitamtech.com" <vivek at advaitamtech.com>
> Sent: Monday, 9 January, 2017 3:58pm
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
> Authentication Required"
>
> Hi All,
>
>
>
> Am trying to proxy the "407 Proxy Authentication Required" through
> freeswitch,
>
>
>
> My Objective is to proxy the "407 Proxy Authentication Required" received
> from the third party server to another server which is connected to
> freeswitch.
>
>
>
> Is it possible to do this?. If possible how do I do it. Please help me out.
>
>
>
> Thanks,
>
> Vivek.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Rich.freeswitch at Branham.us  Thu Jan 12 08:41:31 2017
From: Rich.freeswitch at Branham.us (Richard A. Branham, Jr.)
Date: Thu, 12 Jan 2017 00:41:31 -0500
Subject: [Freeswitch-users] Merging two calls
Message-ID: <1f653343f2be49b3904bcef0a1436ca4.squirrel@webmail.branham.us>

I am trying to merge 2 separate calls together into a conference or 3-way
call while maintaining the ability to put each one on/off hold or hang up
without affecting the other.

Using Verto in a web app, the usual call flows to FS extensions and the
PSTN are working.  The user can put an existing call on hold and make
another call.  If a call comes in and the user answers it, the first call
is placed on hold and the second one becomes active.

I have implemented various switch commands using ESL and the UUIDs of
calls and channels can be retrieved.

>From here we'd like to place all 3 into a conference or 3-way call while
enabling the "main" user to place the other calls on/off hold or hang them
up individually.  Is it possible to merge 2 separate calls into 1 while
maintaining the same control over the calls as when they're just 1-to-1?

Thanks!
Richard



From jose.lopes at itcenter.com.pt  Thu Jan 12 13:02:56 2017
From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=)
Date: Thu, 12 Jan 2017 10:02:56 +0000
Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
 to sip external voicemail
In-Reply-To: <CAKbxfG8_4QqXZqjsMFF5nPHGcdpcFzBdSsNqjVO+0eTwAGULFQ@mail.gmail.com>
References: <CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com>
	<4b92cc8f-575a-ce6c-fd16-e34273911129@xbipin.com>
	<CAEJMVkDiw3EBiKZ=byFyc2fxGJ8EQ7ixzn8djDZpYCv=EmN+Cg@mail.gmail.com>
	<CAK5rbQuc1atHOS=jiXLBkFBa8Zfy2gNpbo8J6QXZJsxmyH+TWQ@mail.gmail.com>
	<CAKbxfG91mpFdc346Ezg2fjoHJeJtXQM=KFdSgT=PWg4mQ3HXrg@mail.gmail.com>
	<CAKbxfG_Z6iLx1-7S+j_GM0azYyt5uuh_Aof10LObVqVqA4vqVg@mail.gmail.com>
	<CAK5rbQuFhZU4jEPuwTMHdrimn25oGFXEKmGo-ep7EspCHo+r7w@mail.gmail.com>
	<CAKbxfG-zXz4zoiQXq_M=WHUqYoRzR-dpXdeo+nmJNGuR_amROg@mail.gmail.com>
	<CAK5rbQujsmYT2RXfVvENTMpPV0jo2ix150u3YGRTmZKcbfnGWg@mail.gmail.com>
	<CAKbxfG9HizcGj4MiFiXU3enO4KdYgUtZVsZj1n1aDdJufhtR_g@mail.gmail.com>
	<CAK5rbQs_M5TyTeerb5iZ+RfCKubstdWfh1=gGEHtQWqPbZCqig@mail.gmail.com>
	<CAKbxfG8_4QqXZqjsMFF5nPHGcdpcFzBdSsNqjVO+0eTwAGULFQ@mail.gmail.com>
Message-ID: <CAK5rbQsKP+ZN-7fMwsWESoCXwXuJ46TUhbKNHis1TF+UxjGsOw@mail.gmail.com>

I created the issue at JIRA: https://freeswitch.org/jira/browse/FS-9939 .
I tried to put all the information discussed here.

Thanks again for your help.


2017-01-10 18:59 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Yes, please file a JIRA because that was my main concern.  Its very easy
> for me to forget about this thread or not notice it was updated.
> Once it is in JIRA it serves a place holder for all the info.
>
>
> On Tue, Jan 10, 2017 at 12:27 PM, Jos? Lopes <jose.lopes at itcenter.com.pt>
> wrote:
>
>> Hello Anthony
>>
>> Thanks for your reply. I am sorry, I didn't notice that it was a diagnose
>> test.
>> I am available to make the tests that you need to analyse this situation
>> and i put bellow information about the test you ask.
>> If you see it is better to create a issue on JIRA, i will do it.
>> Thanks for your and FreeSwitch Team effort.
>>
>>
>> If I add a sleep of 2000 at voicemail server, after the answer there is
>> no audio cut off (I put bellow the change on dialplan).
>> But there is silence of 3/4 seconds between the ivr-say_name and the
>> initial message from voicemail without audio cut off.
>>
>> Let me know if you need more tests or information.
>>
>>
>> Dialplan on FreeSwitch Test:
>> <include>
>>   <context name="default">
>>     <extension name="itsp_send_call">
>>       <condition field="destination_number" expression="^.*$">
>>         <action application="answer"/>
>>         <action application="playback" data="/usr/share/freeswitch/so
>> unds/en/us/callie/ivr/8000/ivr-say_name.wav"/>
>>         <action application="set" data="ringback=$${us-ring}"/>
>>         <action application="bridge" data="{absolute_codec_string='
>> $${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${des
>> tination_number}"/>
>>       </condition>
>>     </extension>
>>   </context>
>> </include>
>>
>> Extract of Dialplan on FreeSwitch External Voicemail Server
>>
>>     <!-- voicemail main extension -->
>>     <extension name="vmain">
>>       <condition field="destination_number" expression="^vmain$|^4000$|^\*
>> 98$">
>>         <action application="answer"/>
>>         <action application="sleep" data="1000"/>
>>         <action application="sleep" data="2000"/> <!-- line added -->
>>         <action application="voicemail" data="check default
>> ${domain_name}"/>
>>       </condition>
>>     </extension>
>>
>>
>> Best Regards,
>> Jose Lopes
>>
>>
>>
>>
>> 2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> The minute you call it an Issue you should be filing it on JIRA.
>>> We get countless emails a day to the list so I don't always read them
>>> all so you are lucky I have managed to follow this thread.
>>>
>>> https://freeswitch.org/jira
>>>
>>> We have a small team and dealing with the mailing list is a volunteer
>>> effort.
>>>
>>> Here is also a tip.  Just provide the info to questions asked.  I asked
>>> you to do a diagnostic test for me by adding sleep to the other FS.
>>> Regardless if you can change the production or not, its still relevant
>>> to me what happens when you change it.
>>>
>>>
>>>
>>>
>>> On Tue, Jan 10, 2017 at 3:58 AM, Jos? Lopes <jose.lopes at itcenter.com.pt>
>>> wrote:
>>>
>>>> Hello Anthony,
>>>>
>>>> At this replicated scenario, the box I am calling on SIP is FS.
>>>> But on real scenario, the box I am calling on SIP is not Freeswitch, it
>>>> is an external voicemail server and the initial message have audio cut off.
>>>>
>>>> Thanks for the information about variable ringback, I am already using
>>>> on real scenario.
>>>>
>>>> One strange thing is if I use the codec OPUS at verto, this issue
>>>> doesn't happen.
>>>> But I need to use codec PCMU to avoid audio transcoding.
>>>>
>>>> Let me know if you need more information to debug this issue.
>>>>
>>>> Best Regards,
>>>> Jose Lopes
>>>>
>>>>
>>>>
>>>>
>>>> 2017-01-09 18:25 GMT+00:00 Anthony Minessale <
>>>> anthony.minessale at gmail.com>:
>>>>
>>>>> So that concludes that media is already established on the webrtc end
>>>>> and there is no problem with that.
>>>>> The box you are calling on SIP is also FS, you may want to add a sleep
>>>>> 2000 in that dialplan before the voicemail.
>>>>> Also since webrtc has no ringing indication you may want to set the
>>>>> variable ringback to get some audible feedback when making calls.
>>>>>
>>>>>
>>>>> On Fri, Jan 6, 2017 at 5:08 AM, Jos? Lopes <jose.lopes at itcenter.com.pt
>>>>> > wrote:
>>>>>
>>>>>> Hello Anthony,
>>>>>>
>>>>>> Thanks for your reply.
>>>>>>
>>>>>> I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
>>>>>> with ~2 seconds) instead of silence_stream.
>>>>>> When i make the call from verto client, i ear the audio file, then no
>>>>>> audio for ~2/3 seconds and then i ear "id followed by pound" (audio
>>>>>> cut off from voicemail initial message "Please enter your id followed by
>>>>>> pound").
>>>>>>
>>>>>> I checked if i have the variable answer_delay and i don't have it.
>>>>>>
>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>> .org/view/e130e172 .
>>>>>>
>>>>>> There is any thing more that i can do?
>>>>>>
>>>>>>
>>>>>> Best Regards,
>>>>>> Jose Lopes
>>>>>>
>>>>>> 2017-01-05 18:14 GMT+00:00 Anthony Minessale <
>>>>>> anthony.minessale at gmail.com>:
>>>>>>
>>>>>>> Also make sure you don't have answer_delay set in your vars.xml
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> Try making the call with
>>>>>>>>
>>>>>>>> fsctl debug_level 10
>>>>>>>>
>>>>>>>> and observe the logs, answer followed by silence_stream should send
>>>>>>>> audio to the client.
>>>>>>>> Also try playing an audio file instead of silence stream to see if
>>>>>>>> you hear it.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, Jan 5, 2017 at 11:58 AM, Jos? Lopes <
>>>>>>>> jose.lopes at itcenter.com.pt> wrote:
>>>>>>>>
>>>>>>>>> Hello Brian,
>>>>>>>>>
>>>>>>>>> Thanks for your reply.
>>>>>>>>>
>>>>>>>>> I tried the dialplan bellow with silence_stream://2000, and i have
>>>>>>>>> that issue.
>>>>>>>>> I tried with silence_stream://3000 and the audio cut off is
>>>>>>>>> greater.
>>>>>>>>> Without the playback, there is no audio cut off, but FreeSwitch
>>>>>>>>> doesn't send any rtp packets to verto client before the bridge.
>>>>>>>>>
>>>>>>>>> There is any thing more that i can do?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> <include>
>>>>>>>>>   <context name="default">
>>>>>>>>>     <extension name="call_debug" continue="true">
>>>>>>>>>       <condition field="${call_debug}" expression="^true$"
>>>>>>>>> break="never">
>>>>>>>>>         <action application="info"/>
>>>>>>>>>       </condition>
>>>>>>>>>     </extension>
>>>>>>>>>     <extension name="itsp_send_call">
>>>>>>>>>       <condition field="destination_number" expression="^.*$">
>>>>>>>>>         <action application="answer"/>
>>>>>>>>>         <action application="playback"
>>>>>>>>> data="silence_stream://2000"/>
>>>>>>>>>         <action application="bridge" data="{absolute_codec_string='
>>>>>>>>> PCMU'}sofia/gateway/1002/${destination_number}"/>
>>>>>>>>>       </condition>
>>>>>>>>>     </extension>
>>>>>>>>>   </context>
>>>>>>>>> </include>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Best Regards,
>>>>>>>>> Jose Lopes
>>>>>>>>>
>>>>>>>>> 2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:
>>>>>>>>>
>>>>>>>>>> Prefix them with silence_stream://2000 or 3000 and it should go
>>>>>>>>>> away.
>>>>>>>>>>
>>>>>>>>>> /b
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> hi,
>>>>>>>>>>>
>>>>>>>>>>> i have the same issue, i think its related to slow audio setup
>>>>>>>>>>> during the call
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Regards,
>>>>>>>>>>> Bipin
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> ------------------------------
>>>>>>>>>>> -------- Original Message --------
>>>>>>>>>>> Subject: [Freeswitch-users] Audio cut off at the begin of the
>>>>>>>>>>> verto call to sip external voicemail
>>>>>>>>>>> From: Jos? Lopes <jose.lopes at itcenter.com.pt>
>>>>>>>>>>> <jose.lopes at itcenter.com.pt>
>>>>>>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freesw
>>>>>>>>>>> itch.org> <freeswitch-users at lists.freeswitch.org>
>>>>>>>>>>> Date: 1/5/2017, 6:35:45 PM
>>>>>>>>>>>
>>>>>>>>>>> Hello Guys,
>>>>>>>>>>>
>>>>>>>>>>> I have audio cut off at the begin of the verto call to
>>>>>>>>>>>  FreeSwitch that redirect to sip external voicemail (Access voicemail
>>>>>>>>>>> mailbox) .
>>>>>>>>>>>
>>>>>>>>>>> This happen when I use PCMU at verto codecs and sip codecs (if i
>>>>>>>>>>> use opus at verto codecs, there is no issue, but this causes audio
>>>>>>>>>>> transcoding) .
>>>>>>>>>>>
>>>>>>>>>>> At dialplan i used the example "Bridging from WebRTC (mod_verto)
>>>>>>>>>>> to PSTN/ITSPs" from https://freeswitch.org/conflue
>>>>>>>>>>> nce/display/FREESWITCH/mod_verto.
>>>>>>>>>>> I notice if i remove the playback action, there is no issue. But
>>>>>>>>>>> I need the playback action to send rtp packets to verto client.
>>>>>>>>>>>
>>>>>>>>>>> I simulate this using another FreeSwitch as external voicemail
>>>>>>>>>>> server and I only listen "id followed by pound" from the initial message of
>>>>>>>>>>> voicemail ("Please enter your id followed by pound").
>>>>>>>>>>> The log of this call is at https://pastebin.freeswitch
>>>>>>>>>>> .org/view/507fa115
>>>>>>>>>>>
>>>>>>>>>>> What I can do to use PCMU at verto codecs and sip codecs on type
>>>>>>>>>>> of call?
>>>>>>>>>>> Should i open a issue on FreeSwitch JIRA ?
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Best regards,
>>>>>>>>>>> Jose Lopes
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> 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/free
>>>>>>>>>>> switch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>>
>>>>>>>>>> *Brian West*
>>>>>>>>>> brian at freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>>>>> http://www.freeswitchbook.com
>>>>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>>>>>
>>>>>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>>>>>>>>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>>>>
>>>>>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>>>>>> *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/free
>>>>>>>>>> switch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ____________________________________________________________
>>>>>>>>> _____________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>> switch-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
>>>>>>>> <(919)%20386-9900>
>>>>>>>>
>>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> 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
>>>>>>> <(919)%20386-9900>
>>>>>>>
>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-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 <(919)%20386-9900>
>>>>>
>>>>>
>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-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 <(919)%20386-9900>
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 <(919)%20386-9900>
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Thu Jan 12 17:24:00 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 14:24:00 +0000
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
Message-ID: <CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>

Hello,

Thanks for replying! I'm simply bridging to the user like

user/1001

This fails, of course.

David
On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

> Can you post your bridge command call? Do you use sofia_contact() to build
> the dial string?
>
>               Regards, __Yehavi:
>
> 2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>
> :
>
> hello guys,
>
> i have 2 profiles for two different ip addresses (IP1 and IP2), both
> profiles have force-register-domain and force-register-db-domain to the
> IP1, i can register on IP2 but when calling this user like user at IP1, fs
> returns USER_NOT_REGISTERED... i thought by forcing the register domain i
> could call the user to any of the 2 ips... any ideas?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From yehavi.bourvine at gmail.com  Thu Jan 12 18:45:51 2017
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 12 Jan 2017 17:45:51 +0200
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
Message-ID: <CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>

try bridging to sofia_contact(1001) and see whether it works.

       __Yehavi:

2017-01-12 16:24 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:

> Hello,
>
> Thanks for replying! I'm simply bridging to the user like
>
> user/1001
>
> This fails, of course.
>
> David
>
> On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Can you post your bridge command call? Do you use sofia_contact() to
>> build the dial string?
>>
>>               Regards, __Yehavi:
>>
>> 2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com
>> >:
>>
>> hello guys,
>>
>> i have 2 profiles for two different ip addresses (IP1 and IP2), both
>> profiles have force-register-domain and force-register-db-domain to the
>> IP1, i can register on IP2 but when calling this user like user at IP1, fs
>> returns USER_NOT_REGISTERED... i thought by forcing the register domain i
>> could call the user to any of the 2 ips... any ideas?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> ?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From magnus.kelly at gmail.com  Thu Jan 12 18:55:29 2017
From: magnus.kelly at gmail.com (Magnus Kelly)
Date: Thu, 12 Jan 2017 15:55:29 +0000
Subject: [Freeswitch-users] Quick Question
Message-ID: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>

Hello All,

would be very helpful if someone could help me with quick reminder on how
to change SIP header on bridge

as in  - incoming call on LegA Bridged to Leg B while changing the
 "ip:FreeSWITCH" to sip: +44xxxxxxxxxx

now this format appears as default -

From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX:5068;transport=tcp>;tag=
UB7tNX0mmBt7j

And I would like help to change it to be in this format -

From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X:5068;
transport=tcp>;tag=UB7tNX0mmBt7j


Thanks
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From david.villasmil.work at gmail.com  Thu Jan 12 19:20:36 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 11:20:36 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
Message-ID: <CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>

In the form of:

<action application="bridge" data="user/${sofia_contact(361110000)}"/>

didn't work... was that the correct way of using it?

thanks again
?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337

On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> try bridging to sofia_contact(1001) and see whether it works.
>
>        __Yehavi:
>
> 2017-01-12 16:24 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com
> >:
>
>> Hello,
>>
>> Thanks for replying! I'm simply bridging to the user like
>>
>> user/1001
>>
>> This fails, of course.
>>
>> David
>>
>> On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>> Can you post your bridge command call? Do you use sofia_contact() to
>>> build the dial string?
>>>
>>>               Regards, __Yehavi:
>>>
>>> 2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.co
>>> m>:
>>>
>>> hello guys,
>>>
>>> i have 2 profiles for two different ip addresses (IP1 and IP2), both
>>> profiles have force-register-domain and force-register-db-domain to the
>>> IP1, i can register on IP2 but when calling this user like user at IP1, fs
>>> returns USER_NOT_REGISTERED... i thought by forcing the register domain i
>>> could call the user to any of the 2 ips... any ideas?
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>> ?
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From devang.nathwani31589 at gmail.com  Thu Jan 12 19:44:09 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Thu, 12 Jan 2017 22:14:09 +0530
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAGOm0PKeJSJpqKvcQVYY_C0WNneofdiUBuiVbvJRzekrZdqFdQ@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
	<CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>
	<CAGOm0P+Ld6kDthxmYbcnPmPjnXfQbNQtdnwqoMreZDTeUF9V1A@mail.gmail.com>
	<CAGOm0PKeJSJpqKvcQVYY_C0WNneofdiUBuiVbvJRzekrZdqFdQ@mail.gmail.com>
Message-ID: <CAGOm0PJdLn01W4jKiaUTf4Q5tRv0uPGHccYO9u313B7Ox50yWw@mail.gmail.com>

Can i have any suggestion on this issue?
i have tried creating new dialplan as shown here
https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause
as well as i have tried external lua to replace the cause code, suggest
here,
https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause

i am also attaching the sip log for the 480
https://pastebin.freeswitch.org/view/0dbbf7d7

On Fri, Jan 6, 2017 at 3:45 PM, devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> yeah, i am getting reference from here,
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables
>
> On Fri, Jan 6, 2017 at 3:38 PM, devang nathwani <
> devang.nathwani31589 at gmail.com> wrote:
>
>> Hello,
>>
>> no i haven't read when i drafted the previous mail!
>>
>> The scenario is something, gateway is completely down suppose because of
>> hardware failure or electricity issue.
>>
>> so freeswitch trying to send caller's request to gateway, as gateway is
>> completely down freeswitch continuously trying to send 'INVITE' request to
>> gateway, getting nothing in response so freeswitch sending 480
>> TEMPORARILY_UNAVAILABLE to caller but i want to change with '503
>> SERVICE_UNAVAILABLE'
>>
>> and fscli showing, [CS_EXECUTE] [ALLOTTED_TIMEOUT] and [CS_CONSUME_MEDIA]
>> [ORIGINATOR_CANCEL]
>>
>> On Fri, Jan 6, 2017 at 3:18 PM, Gonzalo Gasca Meza <
>> gascagonzalo at gmail.com> wrote:
>>
>>>
>>> Can you share more details about the call flow and debug if possible?
>>> Are you planning to replace an incoming disconnect code or you want to send
>>> an specific disconnect code?
>>>
>>> <action application="set" data="continue_on_fail=TEMPORA
>>> RILY_UNAVAILABLE"/>
>>>
>>> *continue_on_fail* "Controls what happens when the called party can not
>>> be reached (busy/offline). If "true" the dialplan continues to be
>>> processed. If "false" the dialplan will stop processing. Can contain the
>>> return messages that will continue on fail also."
>>>
>>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>>
>>> *failure_causes*  Controls which failure causes will be considered as a
>>> failure to the bridge(s). This will change the values for which
>>> continue_on_fail will fail by default unless continue_on_fail is set to
>>> true.
>>>
>>> Depending of your flow you can use:
>>>
>>> <action application="hangup" data="SERVICE_UNAVAILABLE"/>
>>>
>>>
>>> On Fri, Jan 6, 2017 at 1:07 AM, Vladyslav Zakhozhai <
>>> v.zakhozhai at gmail.com> wrote:
>>>
>>>> Devang, I'm no sure but this config should not work.
>>>> I think that you need to handle failure_causes after the bridge and use
>>>> respond command to respond with cause you need.
>>>>
>>>> Maybe I am wrong but this approach will do the work.
>>>>
>>>> 2017-01-06 10:43 GMT+02:00 devang nathwani <
>>>> devang.nathwani31589 at gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with
>>>>> '503 SERVICE_UNAVAILABLE';
>>>>>
>>>>> tried;
>>>>> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
>>>>> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
>>>>> before bridge application
>>>>> But its not working
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> ? ?????????,
>>>> ????????? ???????
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From david.villasmil.work at gmail.com  Thu Jan 12 19:46:06 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 11:46:06 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
	<CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
Message-ID: <CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>

this does work:

<action application="bridge" data="${sofia_contact(internal-second/1001)}"/>

but i have 2 profiles through which the user can be called, "internal" and
"internal-second", and doing it like this kind of kills the purpose, no?

I can think of maybe doing:

<extension name="call_internal" continue="true">
<condition field="context" expression="public" />
<condition field="destination_number" expression="1001">
<action application="set"
data="sofia_result=${sofia_contact(internal/${destination_number})}"/>
<action application="log" data="INFO sofia contact result:
${sofia_result}"/>
<condition field="${result}" data="sofia.*">

<action application="bridge" data="sofia/internal/${destination_number}"/>
<anti-action application="bridge"
data="sofia/internal-second/${destination_number}"/>

</condition>
</condition>
</extension>

i will try this
?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337

On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> In the form of:
>
> <action application="bridge" data="user/${sofia_contact(361110000)}"/>
>
> didn't work... was that the correct way of using it?
>
> thanks again
> ?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> try bridging to sofia_contact(1001) and see whether it works.
>>
>>        __Yehavi:
>>
>> 2017-01-12 16:24 GMT+02:00 David Villasmil <david.villasmil.work at gmail.co
>> m>:
>>
>>> Hello,
>>>
>>> Thanks for replying! I'm simply bridging to the user like
>>>
>>> user/1001
>>>
>>> This fails, of course.
>>>
>>> David
>>>
>>> On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>> Can you post your bridge command call? Do you use sofia_contact() to
>>>> build the dial string?
>>>>
>>>>               Regards, __Yehavi:
>>>>
>>>> 2017-01-11 4:45 GMT+02:00 David Villasmil <
>>>> david.villasmil.work at gmail.com>:
>>>>
>>>> hello guys,
>>>>
>>>> i have 2 profiles for two different ip addresses (IP1 and IP2), both
>>>> profiles have force-register-domain and force-register-db-domain to the
>>>> IP1, i can register on IP2 but when calling this user like user at IP1,
>>>> fs returns USER_NOT_REGISTERED... i thought by forcing the register domain
>>>> i could call the user to any of the 2 ips... any ideas?
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>> ?
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From covici at ccs.covici.com  Thu Jan 12 20:12:00 2017
From: covici at ccs.covici.com (John Covici)
Date: Thu, 12 Jan 2017 12:12:00 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
	<CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
	<CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>
Message-ID: <m3tw94fbj3.wl-covici@ccs.covici.com>

There is a form of sofia-contact which will search all profiles, if I
remember its */ followed by user name.

On Thu, 12 Jan 2017 11:46:06 -0500,
David Villasmil wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (quoted-printable)>]
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> this does work:
> 
> <action application="bridge" data="${sofia_contact(internal-second/1001)}"/>
> 
> but i have 2 profiles through which the user can be called, "internal" and "internal-second", and doing it like this kind of kills the purpose, no?
> 
> I can think of maybe doing:
> 
> <extension name="call_internal" continue="true">
> <condition field="context" expression="public" />
> <condition field="destination_number" expression="1001">
> <action application="set" data="sofia_result=${sofia_contact(internal/${destination_number})}"/>
> <action application="log" data="INFO sofia contact result: ${sofia_result}"/>
> <condition field="${result}" data="sofia.*">
> 
> <action application="bridge" data="sofia/internal/${destination_number}"/>
> <anti-action application="bridge" data="sofia/internal-second/${destination_number}"/>
> 
> </condition>
> </condition>
> </extension>
> 
> i will try this
> *
> ?
> 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> 
> On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
>  In the form of:
> 
>  <action application="bridge" data="user/${sofia_contact(361110000)}"/>
> 
>  didn't work... was that the correct way of using it?
> 
>  thanks again
>  *
>  ?
> 
>  Regards,
> 
>  David Villasmil
>  email: david.villasmil.work at gmail.com
>  phone: +34669448337
> 
>  On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> 
>  try bridging to sofia_contact(1001) and see whether it works.
> 
>  __Yehavi:
> 
>  2017-01-12 16:24 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:
> 
>  Hello,
> 
>  Thanks for replying! I'm simply bridging to the user like
> 
>  user/1001
> 
>  This fails, of course.
> 
>  David
> 
>  On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> 
>  Can you post your bridge command call? Do you use sofia_contact() to build the dial string?
> 
>  Regards, __Yehavi:
> 
>  2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:
> 
>  hello guys, 
> 
>  i have 2 profiles for two different ip addresses (IP1 and IP2), both profiles have force-register-domain and force-register-db-domain to the IP1, i can register on IP2 but when calling this user like user at IP1, fs
>  returns USER_NOT_REGISTERED... i thought by forcing the register domain i could call the user to any of the 2 ips... any ideas? 
> 
>  Regards,
> 
>  David Villasmil
>  email: david.villasmil.work at gmail.com
>  phone: +34669448337
>  *
>  ?
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> [2  <text/plain; us-ascii (7bit)>]
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From nb3868 at yp.com  Thu Jan 12 20:19:55 2017
From: nb3868 at yp.com (Nagendra Babu)
Date: Thu, 12 Jan 2017 17:19:55 +0000
Subject: [Freeswitch-users] load multiple grammar
Message-ID: <819781D671CC1D44944772177554935003C4C2E3@ASH-EXCH-MB06.corp.yp.com>

Hi,

I need help regarding loading and activating multiple grammars.

Per FS-2906<https://freeswitch.org/jira/browse/FS-2906>, it is possible to load multiple grammars from FreeSwitch.

I have tried the same using following commands from my C# code. The FS version I have is 1.3.17.

InboundSession.Execute("detect_speech", "unimrcp {start-recognize=false}yesnod yesnod");
InboundSession.Execute("detect_speech", "unimrcp {start-recognize=true}yesno yesno");

I can see that there are two DEFINE-GRAMMAR requests sent to the LumenVox Speech Server. One with yesnod.gram and another with yesno.gram
But there is only one session mentioned in the RECOGNIZE request where as I am expecting it to be having both the sessions described in the DEFINE-GRAMMAR request.
Any idea?


                MRCP/2.0 226 RECOGNIZE 3
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: text/uri-list
                Cancel-If-Queue: false
                Start-Input-Timers: false
                Speech-Complete-Timeout: 3000
                Content-Length: 13

                session:yesno
>>>> There should be session:yesnod also.

I do not see a way to do this in detect_speech.

Any help would be great.



Here is the define grammar message too...

                MRCP/2.0 318 DEFINE-GRAMMAR 1
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: application/srgs
                Content-Id: yesnod
                Content-Length: 158

                #ABNF 1.0 UTF-8;
language en-US;
mode dtmf;
tag-format <semantics/1.0>;

root $yesorno;

$yes = 1 {out = "yes"};
$no = 2 {out = "no"};
$yesorno = $yes | $no;

01/12/2017 07:16:09.790,RECV,OnLoadGrammarRe,Received cseq grammar load (1) reply for CallIndex 51
01/12/2017 07:16:09.790,SEND,SendPacket     ,

                MRCP/2.0 116 1 200 COMPLETE
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Completion-Cause: 000 success


01/12/2017 07:16:10.123,RECV,ProcessRcvdPkt ,Got MRCP Message -

                MRCP/2.0 408 DEFINE-GRAMMAR 2
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: application/srgs
                Content-Id: yesno
                Content-Length: 249

                #ABNF 1.0 UTF-8;
                language en-US; //use the American English pronunciation dictionary.
                mode voice;  //the input for this grammar will be spoken words.
                tag-format <semantics/1.0>;

                root $yesorno;

                $yes = yes;
                $no = no;
                $yesorno = $yes | $no;

01/12/2017 07:16:10.128,RECV,OnLoadGrammarRe,Received cseq grammar load (2) reply for CallIndex 51
01/12/2017 07:16:10.129,SEND,SendPacket     ,

                MRCP/2.0 116 2 200 COMPLETE
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Completion-Cause: 000 success


01/12/2017 07:16:10.459,RECV,ProcessRcvdPkt ,Got MRCP Message -

                MRCP/2.0 226 RECOGNIZE 3
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: text/uri-list
                Cancel-If-Queue: false
                Start-Input-Timers: false
                Speech-Complete-Timeout: 3000
                Content-Length: 13

                session:yesno


01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Speech-Complete-Timeout' with value '3000'
01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Start-Input-Timers' with value 'false'
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting speech_complete_timeout to 3000 ms
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting sensitivity level to 50
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting Speed vs Accuracy to 0.5
01/12/2017 07:16:10.461,SEND,SendPacket     ,

                MRCP/2.0 87 3 200 IN-PROGRESS




Thanks and regards,
Nagendra

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From david.villasmil.work at gmail.com  Thu Jan 12 22:27:38 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 19:27:38 +0000
Subject: [Freeswitch-users] Quick Question
In-Reply-To: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
References: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
Message-ID: <CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>

To set the 'from' header you need to use effective_caller_id_number or with
origination_caller_id_number.

Caveat is, *effective_caller_id_number* can ONLY be set *BEFORE* the
bridge, like so:

        <action application="set"
data="effective_caller_id_number=${caller_id_number}"/>
        <action application="set"
data="effective_caller_id_name=${caller_id_number}"/>
        <action application="bridge" data="sofia/gateway/your-gateway/$1"/>

if you set it on the bridge it will not work.

And you can also use origination_caller_id_number, BUT you need to set it *on
the bridge itself*:

<action application="bridge"
data="{origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_number}}sofia/gateway/your'gateway/$1"/>

hope it's clear.

On Thu, Jan 12, 2017 at 10:56 AM Magnus Kelly <magnus.kelly at gmail.com>
wrote:

Hello All,

would be very helpful if someone could help me with quick reminder on how
to change SIP header on bridge

as in  - incoming call on LegA Bridged to Leg B while changing the
 "ip:FreeSWITCH" to sip: +44xxxxxxxxxx

now this format appears as default -

From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX:5068;transport=tcp>;tag=
UB7tNX0mmBt7j

And I would like help to change it to be in this format -

From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X:5068;
transport=tcp>;tag=UB7tNX0mmBt7j


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 david.villasmil.work at gmail.com  Thu Jan 12 22:29:24 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 19:29:24 +0000
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <m3tw94fbj3.wl-covici@ccs.covici.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
	<CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
	<CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>
	<m3tw94fbj3.wl-covici@ccs.covici.com>
Message-ID: <CAFGRPVq3Hn1=pCx1GxfGeTLd_+-8FgiAXYWSpq=tBSiiFqaiJw@mail.gmail.com>

Hello

That didn't worked for me, and i can't find anything similar to that

David
On Thu, Jan 12, 2017 at 12:12 PM John Covici <covici at ccs.covici.com> wrote:

> There is a form of sofia-contact which will search all profiles, if I
> remember its */ followed by user name.
>
> On Thu, 12 Jan 2017 11:46:06 -0500,
> David Villasmil wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; UTF-8 (quoted-printable)>]
> > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > this does work:
> >
> > <action application="bridge"
> data="${sofia_contact(internal-second/1001)}"/>
> >
> > but i have 2 profiles through which the user can be called, "internal"
> and "internal-second", and doing it like this kind of kills the purpose, no?
> >
> > I can think of maybe doing:
> >
> > <extension name="call_internal" continue="true">
> > <condition field="context" expression="public" />
> > <condition field="destination_number" expression="1001">
> > <action application="set"
> data="sofia_result=${sofia_contact(internal/${destination_number})}"/>
> > <action application="log" data="INFO sofia contact result:
> ${sofia_result}"/>
> > <condition field="${result}" data="sofia.*">
> >
> > <action application="bridge"
> data="sofia/internal/${destination_number}"/>
> > <anti-action application="bridge"
> data="sofia/internal-second/${destination_number}"/>
> >
> > </condition>
> > </condition>
> > </extension>
> >
> > i will try this
> > *
> > ?
> >
> > Regards,
> >
> > David Villasmil
> > email: david.villasmil.work at gmail.com
> > phone: +34669448337
> >
> > On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
> >
> >  In the form of:
> >
> >  <action application="bridge" data="user/${sofia_contact(361110000)}"/>
> >
> >  didn't work... was that the correct way of using it?
> >
> >  thanks again
> >  *
> >  ?
> >
> >  Regards,
> >
> >  David Villasmil
> >  email: david.villasmil.work at gmail.com
> >  phone: +34669448337
> >
> >  On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
> >
> >  try bridging to sofia_contact(1001) and see whether it works.
> >
> >  __Yehavi:
> >
> >  2017-01-12 16:24 GMT+02:00 David Villasmil <
> david.villasmil.work at gmail.com>:
> >
> >  Hello,
> >
> >  Thanks for replying! I'm simply bridging to the user like
> >
> >  user/1001
> >
> >  This fails, of course.
> >
> >  David
> >
> >  On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
> >
> >  Can you post your bridge command call? Do you use sofia_contact() to
> build the dial string?
> >
> >  Regards, __Yehavi:
> >
> >  2017-01-11 4:45 GMT+02:00 David Villasmil <
> david.villasmil.work at gmail.com>:
> >
> >  hello guys,
> >
> >  i have 2 profiles for two different ip addresses (IP1 and IP2), both
> profiles have force-register-domain and force-register-db-domain to the
> IP1, i can register on IP2 but when calling this user like user at IP1, fs
> >  returns USER_NOT_REGISTERED... i thought by forcing the register domain
> i could call the user to any of the 2 ips... any ideas?
> >
> >  Regards,
> >
> >  David Villasmil
> >  email: david.villasmil.work at gmail.com
> >  phone: +34669448337
> >  *
> >  ?
> >
> >
> _________________________________________________________________________
> >  Professional FreeSWITCH Consulting Services:
> >  consulting at freeswitch.org
> >  http://www.freeswitchsolutions.com
> >
> >  Official FreeSWITCH Sites
> >  http://www.freeswitch.org
> >  http://confluence.freeswitch.org
> >  http://www.cluecon.com
> >
> >  FreeSWITCH-users mailing list
> >  FreeSWITCH-users at lists.freeswitch.org
> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> >
> _________________________________________________________________________
> >  Professional FreeSWITCH Consulting Services:
> >  consulting at freeswitch.org
> >  http://www.freeswitchsolutions.com
> >
> >  Official FreeSWITCH Sites
> >  http://www.freeswitch.org
> >  http://confluence.freeswitch.org
> >  http://www.cluecon.com
> >
> >  FreeSWITCH-users mailing list
> >  FreeSWITCH-users at lists.freeswitch.org
> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> >
> _________________________________________________________________________
> >  Professional FreeSWITCH Consulting Services:
> >  consulting at freeswitch.org
> >  http://www.freeswitchsolutions.com
> >
> >  Official FreeSWITCH Sites
> >  http://www.freeswitch.org
> >  http://confluence.freeswitch.org
> >  http://www.cluecon.com
> >
> >  FreeSWITCH-users mailing list
> >  FreeSWITCH-users at lists.freeswitch.org
> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> >
> _________________________________________________________________________
> >  Professional FreeSWITCH Consulting Services:
> >  consulting at freeswitch.org
> >  http://www.freeswitchsolutions.com
> >
> >  Official FreeSWITCH Sites
> >  http://www.freeswitch.org
> >  http://confluence.freeswitch.org
> >  http://www.cluecon.com
> >
> >  FreeSWITCH-users mailing list
> >  FreeSWITCH-users at lists.freeswitch.org
> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> > [2  <text/plain; us-ascii (7bit)>]
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Thu Jan 12 22:32:52 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 Jan 2017 14:32:52 -0500
Subject: [Freeswitch-users] Quick Question
In-Reply-To: <CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>
References: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
	<CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>
Message-ID: <FCDD9292-FE12-47B8-8B67-5DB0446A2931@jerris.com>

you REALLY should not be using effective_* vars

use the originate_* ones on the dial string instead

> On Jan 12, 2017, at 2:27 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> To set the 'from' header you need to use effective_caller_id_number or with origination_caller_id_number.
> 
> Caveat is, effective_caller_id_number can ONLY be set BEFORE the bridge, like so:
> 
>         <action application="set" data="effective_caller_id_number=${caller_id_number}"/>
>         <action application="set" data="effective_caller_id_name=${caller_id_number}"/>
>         <action application="bridge" data="sofia/gateway/your-gateway/$1"/>
> 
> if you set it on the bridge it will not work.
> 
> And you can also use origination_caller_id_number, BUT you need to set it on the bridge itself:
> 
> <action application="bridge" data="{origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_number}}sofia/gateway/your'gateway/$1"/>
> 
> hope it's clear. 
> 
> On Thu, Jan 12, 2017 at 10:56 AM Magnus Kelly <magnus.kelly at gmail.com <mailto:magnus.kelly at gmail.com>> wrote:
> Hello All,
> 
> would be very helpful if someone could help me with quick reminder on how to change SIP header on bridge
> 
> as in  - incoming call on LegA Bridged to Leg B while changing the  "ip:FreeSWITCH" to sip: +44xxxxxxxxxx
> 
> now this format appears as default -
> 
> From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX:5068;transport=tcp>;tag=UB7tNX0mmBt7j
> 
> And I would like help to change it to be in this format -
> 
> From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X:5068;transport=tcp>;tag=UB7tNX0mmBt7j
> 
> 
> Thanks
> 
> 
> _________________________________________________________________________
> 
> 
> 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 covici at ccs.covici.com  Thu Jan 12 22:59:06 2017
From: covici at ccs.covici.com (John Covici)
Date: Thu, 12 Jan 2017 14:59:06 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <CAFGRPVq3Hn1=pCx1GxfGeTLd_+-8FgiAXYWSpq=tBSiiFqaiJw@mail.gmail.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
	<CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
	<CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>
	<m3tw94fbj3.wl-covici@ccs.covici.com>
	<CAFGRPVq3Hn1=pCx1GxfGeTLd_+-8FgiAXYWSpq=tBSiiFqaiJw@mail.gmail.com>
Message-ID: <m3shoof3sl.wl-covici@ccs.covici.com>

OK, the form of this is on the bridge itself use <action
application="bridge"
data="${sofia_contact(*/${dialed_extension}@${domain})}   and what
ever else.

Hope this helps.

On Thu, 12 Jan 2017 14:29:24 -0500,
David Villasmil wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (quoted-printable)>]
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> Hello 
> 
> That didn't worked for me, and i can't find anything similar to that 
> 
> David
> 
> On Thu, Jan 12, 2017 at 12:12 PM John Covici <covici at ccs.covici.com> wrote:
> 
>  There is a form of sofia-contact which will search all profiles, if I
>  remember its */ followed by user name.
> 
>  On Thu, 12 Jan 2017 11:46:06 -0500,
>  David Villasmil wrote:
>  >
>  > [1 <multipart/alternative (7bit)>]
>  > [1.1 <text/plain; UTF-8 (quoted-printable)>]
>  > [1.2 <text/html; UTF-8 (quoted-printable)>]
>  > this does work:
>  >
>  > <action application="bridge" data="${sofia_contact(internal-second/1001)}"/>
>  >
>  > but i have 2 profiles through which the user can be called, "internal" and "internal-second", and doing it like this kind of kills the purpose, no?
>  >
>  > I can think of maybe doing:
>  >
>  > <extension name="call_internal" continue="true">
>  > <condition field="context" expression="public" />
>  > <condition field="destination_number" expression="1001">
>  > <action application="set" data="sofia_result=${sofia_contact(internal/${destination_number})}"/>
>  > <action application="log" data="INFO sofia contact result: ${sofia_result}"/>
>  > <condition field="${result}" data="sofia.*">
>  >
>  > <action application="bridge" data="sofia/internal/${destination_number}"/>
>  > <anti-action application="bridge" data="sofia/internal-second/${destination_number}"/>
>  >
>  > </condition>
>  > </condition>
>  > </extension>
>  >
>  > i will try this
>  > *
>  > ?
>  >
>  > Regards,
>  >
>  > David Villasmil
>  > email: david.villasmil.work at gmail.com
>  > phone: +34669448337
>  >
>  > On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
>  >
>  > In the form of:
>  >
>  > <action application="bridge" data="user/${sofia_contact(361110000)}"/>
>  >
>  > didn't work... was that the correct way of using it?
>  >
>  > thanks again
>  > *
>  > ?
>  >
>  > Regards,
>  >
>  > David Villasmil
>  > email: david.villasmil.work at gmail.com
>  > phone: +34669448337
>  >
>  > On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
>  >
>  > try bridging to sofia_contact(1001) and see whether it works.
>  >
>  > __Yehavi:
>  >
>  > 2017-01-12 16:24 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:
>  >
>  > Hello,
>  >
>  > Thanks for replying! I'm simply bridging to the user like
>  >
>  > user/1001
>  >
>  > This fails, of course.
>  >
>  > David
>  >
>  > On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
>  >
>  > Can you post your bridge command call? Do you use sofia_contact() to build the dial string?
>  >
>  > Regards, __Yehavi:
>  >
>  > 2017-01-11 4:45 GMT+02:00 David Villasmil <david.villasmil.work at gmail.com>:
>  >
>  > hello guys,
>  >
>  > i have 2 profiles for two different ip addresses (IP1 and IP2), both profiles have force-register-domain and force-register-db-domain to the IP1, i can register on IP2 but when calling this user like user at IP1, fs
>  > returns USER_NOT_REGISTERED... i thought by forcing the register domain i could call the user to any of the 2 ips... any ideas?
>  >
>  > Regards,
>  >
>  > David Villasmil
>  > email: david.villasmil.work at gmail.com
>  > phone: +34669448337
>  > *
>  > ?
>  >
>  > _________________________________________________________________________
>  > Professional FreeSWITCH Consulting Services:
>  > consulting at freeswitch.org
>  > http://www.freeswitchsolutions.com
>  >
>  > Official FreeSWITCH Sites
>  > http://www.freeswitch.org
>  > http://confluence.freeswitch.org
>  > http://www.cluecon.com
>  >
>  > FreeSWITCH-users mailing list
>  > FreeSWITCH-users at lists.freeswitch.org
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > http://www.freeswitch.org
>  >
>  > _________________________________________________________________________
>  > Professional FreeSWITCH Consulting Services:
>  > consulting at freeswitch.org
>  > http://www.freeswitchsolutions.com
>  >
>  > Official FreeSWITCH Sites
>  > http://www.freeswitch.org
>  > http://confluence.freeswitch.org
>  > http://www.cluecon.com
>  >
>  > FreeSWITCH-users mailing list
>  > FreeSWITCH-users at lists.freeswitch.org
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > http://www.freeswitch.org
>  >
>  > _________________________________________________________________________
>  > Professional FreeSWITCH Consulting Services:
>  > consulting at freeswitch.org
>  > http://www.freeswitchsolutions.com
>  >
>  > Official FreeSWITCH Sites
>  > http://www.freeswitch.org
>  > http://confluence.freeswitch.org
>  > http://www.cluecon.com
>  >
>  > FreeSWITCH-users mailing list
>  > FreeSWITCH-users at lists.freeswitch.org
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > http://www.freeswitch.org
>  >
>  > _________________________________________________________________________
>  > Professional FreeSWITCH Consulting Services:
>  > consulting at freeswitch.org
>  > http://www.freeswitchsolutions.com
>  >
>  > Official FreeSWITCH Sites
>  > http://www.freeswitch.org
>  > http://confluence.freeswitch.org
>  > http://www.cluecon.com
>  >
>  > FreeSWITCH-users mailing list
>  > FreeSWITCH-users at lists.freeswitch.org
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > http://www.freeswitch.org
>  >
>  > [2 <text/plain; us-ascii (7bit)>]
>  > _________________________________________________________________________
>  > Professional FreeSWITCH Consulting Services:
>  > consulting at freeswitch.org
>  > http://www.freeswitchsolutions.com
>  >
>  > Official FreeSWITCH Sites
>  > http://www.freeswitch.org
>  > http://confluence.freeswitch.org
>  > http://www.cluecon.com
>  >
>  > FreeSWITCH-users mailing list
>  > FreeSWITCH-users at lists.freeswitch.org
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > http://www.freeswitch.org
> 
>  --
>  Your life is like a penny. You're going to lose it. The question is:
>  How do
>  you spend it?
> 
>  John Covici
>  covici at ccs.covici.com
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> [2  <text/plain; us-ascii (7bit)>]
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

         John Covici
         covici at ccs.covici.com


From manpower13.cse at gmail.com  Thu Jan 12 23:01:39 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Fri, 13 Jan 2017 01:31:39 +0530
Subject: [Freeswitch-users] B-leg for each user
In-Reply-To: <CAMRuXGg+ues0obdPXGa5QL+deZRCXwaXBqEx1bFomQVUO_e5jw@mail.gmail.com>
References: <CAMRuXGiSAndmvkf6z=eAk2V6Q3CF60ijEvYrXx3pcF9-0cSDPw@mail.gmail.com>
	<CAMRuXGjrRoj+QK+eDVvxEpjnzNGoxR4+0XZjF94Hqn9Cn5XvmA@mail.gmail.com>
	<CAMRuXGgBbsHOTqv4fcb1vskQQxY0-b5bOHoqSFj-OwjJE5xqBQ@mail.gmail.com>
	<CAMRuXGg+ues0obdPXGa5QL+deZRCXwaXBqEx1bFomQVUO_e5jw@mail.gmail.com>
Message-ID: <CAMRuXGjQMdi-4VbA-zzin7Q1djGODX0jx-Wu9y-bG0BQY8HPgQ@mail.gmail.com>

I try get each user cdr from goup call.its work fine if user are sip when i
make call user pstn using loopbackm i am getting only one leg event all
user busy
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From lionelfont at free.fr  Fri Jan 13 01:23:23 2017
From: lionelfont at free.fr (lionelfont at free.fr)
Date: Thu, 12 Jan 2017 23:23:23 +0100 (CET)
Subject: [Freeswitch-users] rtp port update after reinvite when initiator is
	behind nat
In-Reply-To: <1081243594.346661790.1484259166223.JavaMail.root@zimbra97-e18.priv.proxad.net>
Message-ID: <1051713214.346672454.1484259803558.JavaMail.root@zimbra97-e18.priv.proxad.net>

Hi,

i'm an user from freeswitch because my trunk's provider is using a freeswitch.

i can't give you the configuration using by my provider and i can't give you any trace from freeswitch

but i'd like explain my problem to know if anyone have an idea from what can cause my issue.

My situation:

       Freeswitch (public ip A.A.A.A) <----> WAN <---> (my public ip B.B.B.B) FIREWALL <----> (my private ip C.C.C.C) IPBX ALCATEL

       During a call, after invite or reinvite, Freeswitch send me RTP DATA on wrong RTP port... 

       For sample:
           I receive a call, after invite from FS and OK from my IPBX, we have a communication who is good. We hear on the two sides.
               FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX

           If i need to hold the call, all is ok. My holding music is hearing.
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ---------------- C.C.C.C:RTPC1 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX


           But when i unhold the call, "randomly", FS send me RTP DATA on the RTP port used before the unhold.. And we have no sound on the two sides...  

           Case good:
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <> IPBX
        

           Case bad:
           FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 <> IPBX
           FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  -------------- C.C.C.C:SIPC1 <-- IPBX (INVITE)
          (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 -------------- C.C.C.C:SIPC1 --> IPBX
           FS<-- A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ---------------- C.C.C.C:RTPC3 <-- IPBX
           FS--> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ---------------- C.C.C.C:RTPC2 --> IPBX


I don't understand why the rtp port is not good updating by freeswitch randomly. I know freeswitch don't use the port that my IPBX give in INVITE packet because it's not the good with the nat from my firewall. I suppose that's the "NDLB-connectile-dysfunction" who is used by freeswitch to find the good rtp port.

If with a little congestion, Freeswitch receive after my reinvite, one or two RTP packet from previous RTP DATA, Freeswitch can use the wrong RTP port? Is there a parameter to ignore after reinvite the five first RTP packet to autoadjust the RTP port?

Thanks to all helping and sorry for my poor english.

Best regards



From david.villasmil.work at gmail.com  Fri Jan 13 02:51:58 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 23:51:58 +0000
Subject: [Freeswitch-users] Quick Question
In-Reply-To: <FCDD9292-FE12-47B8-8B67-5DB0446A2931@jerris.com>
References: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
	<CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>
	<FCDD9292-FE12-47B8-8B67-5DB0446A2931@jerris.com>
Message-ID: <CAFGRPVpeJDQPrnXhmV3K8xf8Qk-iSRaLARieWJLHf15x524T_g@mail.gmail.com>

Curious, effective does work outside the bridge, why shouldn't it be used?
On Thu, Jan 12, 2017 at 2:33 PM Michael Jerris <mike at jerris.com> wrote:

> you REALLY should not be using effective_* vars
>
> use the originate_* ones on the dial string instead
>
> On Jan 12, 2017, at 2:27 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> To set the 'from' header you need to use effective_caller_id_number or
> with origination_caller_id_number.
>
> Caveat is, *effective_caller_id_number* can ONLY be set *BEFORE* the
> bridge, like so:
>
>         <action application="set"
> data="effective_caller_id_number=${caller_id_number}"/>
>         <action application="set"
> data="effective_caller_id_name=${caller_id_number}"/>
>         <action application="bridge" data="sofia/gateway/your-gateway/$1"/>
>
> if you set it on the bridge it will not work.
>
> And you can also use origination_caller_id_number, BUT you need to set it *on
> the bridge itself*:
>
> <action application="bridge"
> data="{origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_number}}sofia/gateway/your'gateway/$1"/>
>
> hope it's clear.
>
> On Thu, Jan 12, 2017 at 10:56 AM Magnus Kelly <magnus.kelly at gmail.com>
> wrote:
>
> Hello All,
>
> would be very helpful if someone could help me with quick reminder on how
> to change SIP header on bridge
>
> as in  - incoming call on LegA Bridged to Leg B while changing the
>  "ip:FreeSWITCH" to sip: +44xxxxxxxxxx
>
> now this format appears as default -
>
> From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX:5068;transport=tcp>;tag=
> UB7tNX0mmBt7j
>
> And I would like help to change it to be in this format -
>
> From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X:5068;
> transport=tcp>;tag=UB7tNX0mmBt7j
>
>
> 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 david.villasmil.work at gmail.com  Fri Jan 13 02:52:03 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 23:52:03 +0000
Subject: [Freeswitch-users] multiple profiles and user not registered
In-Reply-To: <m3shoof3sl.wl-covici@ccs.covici.com>
References: <CAFGRPVrOnv_jq-HQPGvx0OucZdinmUpPrLNaxPkSDsXiUu6xMA@mail.gmail.com>
	<CAAtWKeFfsvUeCSeCXs-HY5nYxHWYFvoF1kDEgJpYpOuU85bs2g@mail.gmail.com>
	<CAFGRPVpqtgZPE87OfEHG9AEhi8mquT35+FZ4hjgDRsp2MO+6pA@mail.gmail.com>
	<CAAtWKeHomGqOgPc99+bcQK51bi+MT_zpv4xdQUX_xZd90i+iLg@mail.gmail.com>
	<CAFGRPVqeDSoP+0zFDssUQT9X6=4EdjhE33WkLdKxKTBF2FNmUQ@mail.gmail.com>
	<CAFGRPVqeuUBW_8YKBxF9oQHsBsR+6wp-SHJCa3M5t2+_SU0YCg@mail.gmail.com>
	<m3tw94fbj3.wl-covici@ccs.covici.com>
	<CAFGRPVq3Hn1=pCx1GxfGeTLd_+-8FgiAXYWSpq=tBSiiFqaiJw@mail.gmail.com>
	<m3shoof3sl.wl-covici@ccs.covici.com>
Message-ID: <CAFGRPVqcYa+M75=M8amDzv2Rn3W4SmKgCbrWARuzhmesNJiZtA@mail.gmail.com>

I will try this as soon as i can

Thanks!
On Thu, Jan 12, 2017 at 2:59 PM John Covici <covici at ccs.covici.com> wrote:

> OK, the form of this is on the bridge itself use <action
> application="bridge"
> data="${sofia_contact(*/${dialed_extension}@${domain})}   and what
> ever else.
>
> Hope this helps.
>
> On Thu, 12 Jan 2017 14:29:24 -0500,
> David Villasmil wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; UTF-8 (quoted-printable)>]
> > [1.2  <text/html; UTF-8 (quoted-printable)>]
> > Hello
> >
> > That didn't worked for me, and i can't find anything similar to that
> >
> > David
> >
> > On Thu, Jan 12, 2017 at 12:12 PM John Covici <covici at ccs.covici.com>
> wrote:
> >
> >  There is a form of sofia-contact which will search all profiles, if I
> >  remember its */ followed by user name.
> >
> >  On Thu, 12 Jan 2017 11:46:06 -0500,
> >  David Villasmil wrote:
> >  >
> >  > [1 <multipart/alternative (7bit)>]
> >  > [1.1 <text/plain; UTF-8 (quoted-printable)>]
> >  > [1.2 <text/html; UTF-8 (quoted-printable)>]
> >  > this does work:
> >  >
> >  > <action application="bridge"
> data="${sofia_contact(internal-second/1001)}"/>
> >  >
> >  > but i have 2 profiles through which the user can be called,
> "internal" and "internal-second", and doing it like this kind of kills the
> purpose, no?
> >  >
> >  > I can think of maybe doing:
> >  >
> >  > <extension name="call_internal" continue="true">
> >  > <condition field="context" expression="public" />
> >  > <condition field="destination_number" expression="1001">
> >  > <action application="set"
> data="sofia_result=${sofia_contact(internal/${destination_number})}"/>
> >  > <action application="log" data="INFO sofia contact result:
> ${sofia_result}"/>
> >  > <condition field="${result}" data="sofia.*">
> >  >
> >  > <action application="bridge"
> data="sofia/internal/${destination_number}"/>
> >  > <anti-action application="bridge"
> data="sofia/internal-second/${destination_number}"/>
> >  >
> >  > </condition>
> >  > </condition>
> >  > </extension>
> >  >
> >  > i will try this
> >  > *
> >  > ?
> >  >
> >  > Regards,
> >  >
> >  > David Villasmil
> >  > email: david.villasmil.work at gmail.com
> >  > phone: +34669448337
> >  >
> >  > On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
> >  >
> >  > In the form of:
> >  >
> >  > <action application="bridge" data="user/${sofia_contact(361110000)}"/>
> >  >
> >  > didn't work... was that the correct way of using it?
> >  >
> >  > thanks again
> >  > *
> >  > ?
> >  >
> >  > Regards,
> >  >
> >  > David Villasmil
> >  > email: david.villasmil.work at gmail.com
> >  > phone: +34669448337
> >  >
> >  > On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
> >  >
> >  > try bridging to sofia_contact(1001) and see whether it works.
> >  >
> >  > __Yehavi:
> >  >
> >  > 2017-01-12 16:24 GMT+02:00 David Villasmil <
> david.villasmil.work at gmail.com>:
> >  >
> >  > Hello,
> >  >
> >  > Thanks for replying! I'm simply bridging to the user like
> >  >
> >  > user/1001
> >  >
> >  > This fails, of course.
> >  >
> >  > David
> >  >
> >  > On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
> >  >
> >  > Can you post your bridge command call? Do you use sofia_contact() to
> build the dial string?
> >  >
> >  > Regards, __Yehavi:
> >  >
> >  > 2017-01-11 4:45 GMT+02:00 David Villasmil <
> david.villasmil.work at gmail.com>:
> >  >
> >  > hello guys,
> >  >
> >  > i have 2 profiles for two different ip addresses (IP1 and IP2), both
> profiles have force-register-domain and force-register-db-domain to the
> IP1, i can register on IP2 but when calling this user like user at IP1, fs
> >  > returns USER_NOT_REGISTERED... i thought by forcing the register
> domain i could call the user to any of the 2 ips... any ideas?
> >  >
> >  > Regards,
> >  >
> >  > David Villasmil
> >  > email: david.villasmil.work at gmail.com
> >  > phone: +34669448337
> >  > *
> >  > ?
> >  >
> >  >
> _________________________________________________________________________
> >  > Professional FreeSWITCH Consulting Services:
> >  > consulting at freeswitch.org
> >  > http://www.freeswitchsolutions.com
> >  >
> >  > Official FreeSWITCH Sites
> >  > http://www.freeswitch.org
> >  > http://confluence.freeswitch.org
> >  > http://www.cluecon.com
> >  >
> >  > FreeSWITCH-users mailing list
> >  > FreeSWITCH-users at lists.freeswitch.org
> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  > http://www.freeswitch.org
> >  >
> >  >
> _________________________________________________________________________
> >  > Professional FreeSWITCH Consulting Services:
> >  > consulting at freeswitch.org
> >  > http://www.freeswitchsolutions.com
> >  >
> >  > Official FreeSWITCH Sites
> >  > http://www.freeswitch.org
> >  > http://confluence.freeswitch.org
> >  > http://www.cluecon.com
> >  >
> >  > FreeSWITCH-users mailing list
> >  > FreeSWITCH-users at lists.freeswitch.org
> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  > http://www.freeswitch.org
> >  >
> >  >
> _________________________________________________________________________
> >  > Professional FreeSWITCH Consulting Services:
> >  > consulting at freeswitch.org
> >  > http://www.freeswitchsolutions.com
> >  >
> >  > Official FreeSWITCH Sites
> >  > http://www.freeswitch.org
> >  > http://confluence.freeswitch.org
> >  > http://www.cluecon.com
> >  >
> >  > FreeSWITCH-users mailing list
> >  > FreeSWITCH-users at lists.freeswitch.org
> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  > http://www.freeswitch.org
> >  >
> >  >
> _________________________________________________________________________
> >  > Professional FreeSWITCH Consulting Services:
> >  > consulting at freeswitch.org
> >  > http://www.freeswitchsolutions.com
> >  >
> >  > Official FreeSWITCH Sites
> >  > http://www.freeswitch.org
> >  > http://confluence.freeswitch.org
> >  > http://www.cluecon.com
> >  >
> >  > FreeSWITCH-users mailing list
> >  > FreeSWITCH-users at lists.freeswitch.org
> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  > http://www.freeswitch.org
> >  >
> >  > [2 <text/plain; us-ascii (7bit)>]
> >  >
> _________________________________________________________________________
> >  > Professional FreeSWITCH Consulting Services:
> >  > consulting at freeswitch.org
> >  > http://www.freeswitchsolutions.com
> >  >
> >  > Official FreeSWITCH Sites
> >  > http://www.freeswitch.org
> >  > http://confluence.freeswitch.org
> >  > http://www.cluecon.com
> >  >
> >  > FreeSWITCH-users mailing list
> >  > FreeSWITCH-users at lists.freeswitch.org
> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  > http://www.freeswitch.org
> >
> >  --
> >  Your life is like a penny. You're going to lose it. The question is:
> >  How do
> >  you spend it?
> >
> >  John Covici
> >  covici at ccs.covici.com
> >
> >
> _________________________________________________________________________
> >  Professional FreeSWITCH Consulting Services:
> >  consulting at freeswitch.org
> >  http://www.freeswitchsolutions.com
> >
> >  Official FreeSWITCH Sites
> >  http://www.freeswitch.org
> >  http://confluence.freeswitch.org
> >  http://www.cluecon.com
> >
> >  FreeSWITCH-users mailing list
> >  FreeSWITCH-users at lists.freeswitch.org
> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >  http://www.freeswitch.org
> >
> > [2  <text/plain; us-ascii (7bit)>]
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Fri Jan 13 03:02:26 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 12 Jan 2017 19:02:26 -0500
Subject: [Freeswitch-users] Quick Question
In-Reply-To: <CAFGRPVpeJDQPrnXhmV3K8xf8Qk-iSRaLARieWJLHf15x524T_g@mail.gmail.com>
References: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
	<CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>
	<FCDD9292-FE12-47B8-8B67-5DB0446A2931@jerris.com>
	<CAFGRPVpeJDQPrnXhmV3K8xf8Qk-iSRaLARieWJLHf15x524T_g@mail.gmail.com>
Message-ID: <82EB36C3-7B78-42C1-A0B9-074FC7652E26@jerris.com>

gets stuck on the channel, and re-used in transfer scenarios where you don?t intend it to be used.  The origination ones are just used in the original originate, the effective ones were an earlier attempt at solving the problem that caused these other issues.  really effective should never need to be used anymore.


> On Jan 12, 2017, at 6:51 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Curious, effective does work outside the bridge, why shouldn't it be used?
> On Thu, Jan 12, 2017 at 2:33 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> you REALLY should not be using effective_* vars
> 
> use the originate_* ones on the dial string instead
> 
>> On Jan 12, 2017, at 2:27 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> To set the 'from' header you need to use effective_caller_id_number or with origination_caller_id_number.
>> 
>> Caveat is, effective_caller_id_number can ONLY be set BEFORE the bridge, like so:
>> 
>>         <action application="set" data="effective_caller_id_number=${caller_id_number}"/>
>>         <action application="set" data="effective_caller_id_name=${caller_id_number}"/>
>>         <action application="bridge" data="sofia/gateway/your-gateway/$1"/>
>> 
>> if you set it on the bridge it will not work.
>> 
>> And you can also use origination_caller_id_number, BUT you need to set it on the bridge itself:
>> 
>> <action application="bridge" data="{origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_number}}sofia/gateway/your'gateway/$1"/>
>> 
>> hope it's clear. 
>> 
>> On Thu, Jan 12, 2017 at 10:56 AM Magnus Kelly <magnus.kelly at gmail.com <mailto:magnus.kelly at gmail.com>> wrote:
>> Hello All,
>> 
>> would be very helpful if someone could help me with quick reminder on how to change SIP header on bridge
>> 
>> as in  - incoming call on LegA Bridged to Leg B while changing the  "ip:FreeSWITCH" to sip: +44xxxxxxxxxx
>> 
>> now this format appears as default -
>> 
>> From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX: <>5068;transport=tcp>;tag=UB7tNX0mmBt7j
>> 
>> And I would like help to change it to be in this format -
>> 
>> From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X <mailto:44xxxxxxxxxx at 10.11X.X.X>:5068;transport=tcp>;tag=UB7tNX0mmBt7j
>> 
>> 
>> Thanks
>> 
>> 
>> _________________________________________________________________________
>> 
>> 
>> Professional FreeSWITCH Consulting Services:
>> 
>> 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> 
>> 
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> 
>> 
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> 
>> 
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> 
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> 
>> 
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> 
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> 
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> 
>> 
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.villasmil.work at gmail.com  Fri Jan 13 03:14:12 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 13 Jan 2017 00:14:12 +0000
Subject: [Freeswitch-users] Quick Question
In-Reply-To: <82EB36C3-7B78-42C1-A0B9-074FC7652E26@jerris.com>
References: <CAAPhwbc8vE6xu4OzMPhdYemrxGz6Qef3HOsUv-GN=bbZRPpD-A@mail.gmail.com>
	<CAFGRPVp3C2uOqXKao+M+qGHU+k9cWkz_BeuX3Ha=crAO=yyqag@mail.gmail.com>
	<FCDD9292-FE12-47B8-8B67-5DB0446A2931@jerris.com>
	<CAFGRPVpeJDQPrnXhmV3K8xf8Qk-iSRaLARieWJLHf15x524T_g@mail.gmail.com>
	<82EB36C3-7B78-42C1-A0B9-074FC7652E26@jerris.com>
Message-ID: <CAFGRPVo_2HHwcTe_tp-oicscnycs=Jpg5w1gdLG-Ui2Oqw2feA@mail.gmail.com>

Ok, don't use it. I actually never use it, i always use origination
straights in the bridge.

Good to know, thanks.

David
On Thu, Jan 12, 2017 at 7:03 PM Michael Jerris <mike at jerris.com> wrote:

> gets stuck on the channel, and re-used in transfer scenarios where you
> don?t intend it to be used.  The origination ones are just used in the
> original originate, the effective ones were an earlier attempt at solving
> the problem that caused these other issues.  really effective should never
> need to be used anymore.
>
>
> On Jan 12, 2017, at 6:51 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> Curious, effective does work outside the bridge, why shouldn't it be used?
> On Thu, Jan 12, 2017 at 2:33 PM Michael Jerris <mike at jerris.com> wrote:
>
> you REALLY should not be using effective_* vars
>
> use the originate_* ones on the dial string instead
>
> On Jan 12, 2017, at 2:27 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> To set the 'from' header you need to use effective_caller_id_number or
> with origination_caller_id_number.
>
> Caveat is, *effective_caller_id_number* can ONLY be set *BEFORE* the
> bridge, like so:
>
>         <action application="set"
> data="effective_caller_id_number=${caller_id_number}"/>
>         <action application="set"
> data="effective_caller_id_name=${caller_id_number}"/>
>         <action application="bridge" data="sofia/gateway/your-gateway/$1"/>
>
> if you set it on the bridge it will not work.
>
> And you can also use origination_caller_id_number, BUT you need to set it *on
> the bridge itself*:
>
> <action application="bridge"
> data="{origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_number}}sofia/gateway/your'gateway/$1"/>
>
> hope it's clear.
>
> On Thu, Jan 12, 2017 at 10:56 AM Magnus Kelly <magnus.kelly at gmail.com>
> wrote:
>
> Hello All,
>
> would be very helpful if someone could help me with quick reminder on how
> to change SIP header on bridge
>
> as in  - incoming call on LegA Bridged to Leg B while changing the
>  "ip:FreeSWITCH" to sip: +44xxxxxxxxxx
>
> now this format appears as default -
>
> From: "+44xxxxxxxxxx" <sip:FreeSWITCH at 10.11X.X.XX:5068;transport=tcp>;tag=
> UB7tNX0mmBt7j
>
> And I would like help to change it to be in this format -
>
> From: "+44xxxxxxxxxx " <sip: +44xxxxxxxxxx at 10.11X.X.X:5068;
> transport=tcp>;tag=UB7tNX0mmBt7j
>
>
> 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
>
>
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>
>
> http://www.freeswitch.org
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
>
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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From david.villasmil.work at gmail.com  Fri Jan 13 04:42:52 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 12 Jan 2017 20:42:52 -0500
Subject: [Freeswitch-users] multiple profiles and user not registered
	(SOLVED)
Message-ID: <CAFGRPVorML9AamhrN4EcG2T+GMh+046qHaMVURTRoGL0HtGF5A@mail.gmail.com>

John,

<action application="bridge" data="${sofia_contact(*/1001 at 1.2.3.4)}"/>

note: 1.2.3.4 being my domain (or IP1)

worked perfectly!

Thanks!

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337

On Thu, Jan 12, 2017 at 6:52 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I will try this as soon as i can
>
> Thanks!
>
> On Thu, Jan 12, 2017 at 2:59 PM John Covici <covici at ccs.covici.com> wrote:
>
>> OK, the form of this is on the bridge itself use <action
>> application="bridge"
>> data="${sofia_contact(*/${dialed_extension}@${domain})}   and what
>> ever else.
>>
>> Hope this helps.
>>
>> On Thu, 12 Jan 2017 14:29:24 -0500,
>> David Villasmil wrote:
>> >
>> > [1  <multipart/alternative (7bit)>]
>> > [1.1  <text/plain; UTF-8 (quoted-printable)>]
>> > [1.2  <text/html; UTF-8 (quoted-printable)>]
>> > Hello
>> >
>> > That didn't worked for me, and i can't find anything similar to that
>> >
>> > David
>> >
>> > On Thu, Jan 12, 2017 at 12:12 PM John Covici <covici at ccs.covici.com>
>> wrote:
>> >
>> >  There is a form of sofia-contact which will search all profiles, if I
>> >  remember its */ followed by user name.
>> >
>> >  On Thu, 12 Jan 2017 11:46:06 -0500,
>> >  David Villasmil wrote:
>> >  >
>> >  > [1 <multipart/alternative (7bit)>]
>> >  > [1.1 <text/plain; UTF-8 (quoted-printable)>]
>> >  > [1.2 <text/html; UTF-8 (quoted-printable)>]
>> >  > this does work:
>> >  >
>> >  > <action application="bridge" data="${sofia_contact(
>> internal-second/1001)}"/>
>> >  >
>> >  > but i have 2 profiles through which the user can be called,
>> "internal" and "internal-second", and doing it like this kind of kills the
>> purpose, no?
>> >  >
>> >  > I can think of maybe doing:
>> >  >
>> >  > <extension name="call_internal" continue="true">
>> >  > <condition field="context" expression="public" />
>> >  > <condition field="destination_number" expression="1001">
>> >  > <action application="set" data="sofia_result=${sofia_
>> contact(internal/${destination_number})}"/>
>> >  > <action application="log" data="INFO sofia contact result:
>> ${sofia_result}"/>
>> >  > <condition field="${result}" data="sofia.*">
>> >  >
>> >  > <action application="bridge" data="sofia/internal/${
>> destination_number}"/>
>> >  > <anti-action application="bridge" data="sofia/internal-second/${
>> destination_number}"/>
>> >  >
>> >  > </condition>
>> >  > </condition>
>> >  > </extension>
>> >  >
>> >  > i will try this
>> >  > *
>> >  > ?
>> >  >
>> >  > Regards,
>> >  >
>> >  > David Villasmil
>> >  > email: david.villasmil.work at gmail.com
>> >  > phone: +34669448337 <+34%20669%2044%2083%2037>
>> >  >
>> >  > On Thu, Jan 12, 2017 at 11:20 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>> >  >
>> >  > In the form of:
>> >  >
>> >  > <action application="bridge" data="user/${sofia_contact(
>> 361110000)}"/>
>> >  >
>> >  > didn't work... was that the correct way of using it?
>> >  >
>> >  > thanks again
>> >  > *
>> >  > ?
>> >  >
>> >  > Regards,
>> >  >
>> >  > David Villasmil
>> >  > email: david.villasmil.work at gmail.com
>> >  > phone: +34669448337 <+34%20669%2044%2083%2037>
>> >  >
>> >  > On Thu, Jan 12, 2017 at 10:45 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>> >  >
>> >  > try bridging to sofia_contact(1001) and see whether it works.
>> >  >
>> >  > __Yehavi:
>> >  >
>> >  > 2017-01-12 16:24 GMT+02:00 David Villasmil <
>> david.villasmil.work at gmail.com>:
>> >  >
>> >  > Hello,
>> >  >
>> >  > Thanks for replying! I'm simply bridging to the user like
>> >  >
>> >  > user/1001
>> >  >
>> >  > This fails, of course.
>> >  >
>> >  > David
>> >  >
>> >  > On Wed, Jan 11, 2017 at 11:45 PM Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>> >  >
>> >  > Can you post your bridge command call? Do you use sofia_contact() to
>> build the dial string?
>> >  >
>> >  > Regards, __Yehavi:
>> >  >
>> >  > 2017-01-11 4:45 GMT+02:00 David Villasmil <
>> david.villasmil.work at gmail.com>:
>> >  >
>> >  > hello guys,
>> >  >
>> >  > i have 2 profiles for two different ip addresses (IP1 and IP2), both
>> profiles have force-register-domain and force-register-db-domain to the
>> IP1, i can register on IP2 but when calling this user like user at IP1, fs
>> >  > returns USER_NOT_REGISTERED... i thought by forcing the register
>> domain i could call the user to any of the 2 ips... any ideas?
>> >  >
>> >  > Regards,
>> >  >
>> >  > David Villasmil
>> >  > email: david.villasmil.work at gmail.com
>> >  > phone: +34669448337 <+34%20669%2044%2083%2037>
>> >  > *
>> >  > ?
>> >  >
>> >  > ____________________________________________________________
>> _____________
>> >  > Professional FreeSWITCH Consulting Services:
>> >  > consulting at freeswitch.org
>> >  > http://www.freeswitchsolutions.com
>> >  >
>> >  > Official FreeSWITCH Sites
>> >  > http://www.freeswitch.org
>> >  > http://confluence.freeswitch.org
>> >  > http://www.cluecon.com
>> >  >
>> >  > FreeSWITCH-users mailing list
>> >  > FreeSWITCH-users at lists.freeswitch.org
>> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  > http://www.freeswitch.org
>> >  >
>> >  > ____________________________________________________________
>> _____________
>> >  > Professional FreeSWITCH Consulting Services:
>> >  > consulting at freeswitch.org
>> >  > http://www.freeswitchsolutions.com
>> >  >
>> >  > Official FreeSWITCH Sites
>> >  > http://www.freeswitch.org
>> >  > http://confluence.freeswitch.org
>> >  > http://www.cluecon.com
>> >  >
>> >  > FreeSWITCH-users mailing list
>> >  > FreeSWITCH-users at lists.freeswitch.org
>> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  > http://www.freeswitch.org
>> >  >
>> >  > ____________________________________________________________
>> _____________
>> >  > Professional FreeSWITCH Consulting Services:
>> >  > consulting at freeswitch.org
>> >  > http://www.freeswitchsolutions.com
>> >  >
>> >  > Official FreeSWITCH Sites
>> >  > http://www.freeswitch.org
>> >  > http://confluence.freeswitch.org
>> >  > http://www.cluecon.com
>> >  >
>> >  > FreeSWITCH-users mailing list
>> >  > FreeSWITCH-users at lists.freeswitch.org
>> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  > http://www.freeswitch.org
>> >  >
>> >  > ____________________________________________________________
>> _____________
>> >  > Professional FreeSWITCH Consulting Services:
>> >  > consulting at freeswitch.org
>> >  > http://www.freeswitchsolutions.com
>> >  >
>> >  > Official FreeSWITCH Sites
>> >  > http://www.freeswitch.org
>> >  > http://confluence.freeswitch.org
>> >  > http://www.cluecon.com
>> >  >
>> >  > FreeSWITCH-users mailing list
>> >  > FreeSWITCH-users at lists.freeswitch.org
>> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  > http://www.freeswitch.org
>> >  >
>> >  > [2 <text/plain; us-ascii (7bit)>]
>> >  > ____________________________________________________________
>> _____________
>> >  > Professional FreeSWITCH Consulting Services:
>> >  > consulting at freeswitch.org
>> >  > http://www.freeswitchsolutions.com
>> >  >
>> >  > Official FreeSWITCH Sites
>> >  > http://www.freeswitch.org
>> >  > http://confluence.freeswitch.org
>> >  > http://www.cluecon.com
>> >  >
>> >  > FreeSWITCH-users mailing list
>> >  > FreeSWITCH-users at lists.freeswitch.org
>> >  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  > http://www.freeswitch.org
>> >
>> >  --
>> >  Your life is like a penny. You're going to lose it. The question is:
>> >  How do
>> >  you spend it?
>> >
>> >  John Covici
>> >  covici at ccs.covici.com
>> >
>> >  ____________________________________________________________
>> _____________
>> >  Professional FreeSWITCH Consulting Services:
>> >  consulting at freeswitch.org
>> >  http://www.freeswitchsolutions.com
>> >
>> >  Official FreeSWITCH Sites
>> >  http://www.freeswitch.org
>> >  http://confluence.freeswitch.org
>> >  http://www.cluecon.com
>> >
>> >  FreeSWITCH-users mailing list
>> >  FreeSWITCH-users at lists.freeswitch.org
>> >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >  http://www.freeswitch.org
>> >
>> > [2  <text/plain; us-ascii (7bit)>]
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>          John Covici
>>          covici at ccs.covici.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
?
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From s.safarov at gmail.com  Fri Jan 13 08:05:27 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 13 Jan 2017 05:05:27 +0000
Subject: [Freeswitch-users] rtp port update after reinvite when
 initiator is behind nat
In-Reply-To: <1051713214.346672454.1484259803558.JavaMail.root@zimbra97-e18.priv.proxad.net>
References: <1081243594.346661790.1484259166223.JavaMail.root@zimbra97-e18.priv.proxad.net>
	<1051713214.346672454.1484259803558.JavaMail.root@zimbra97-e18.priv.proxad.net>
Message-ID: <CAHtxdDcsj++2-cius3FFM8BXbDMXQx+6CF=asn+J6BJaJfNZWA@mail.gmail.com>

Look this ticket FS-9206 <https://freeswitch.org/jira/browse/FS-9206>
Try this branch bugfix/FS-9206-proxy-media-with-enable-3pcc-proxy
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=refs/heads/bugfix/FS-9206-proxy-media-with-enable-3pcc-proxy>

Sergey

??, 13 ???. 2017 ?. ? 1:33, <lionelfont at free.fr>:

> Hi,
>
> i'm an user from freeswitch because my trunk's provider is using a
> freeswitch.
>
> i can't give you the configuration using by my provider and i can't give
> you any trace from freeswitch
>
> but i'd like explain my problem to know if anyone have an idea from what
> can cause my issue.
>
> My situation:
>
>        Freeswitch (public ip A.A.A.A) <----> WAN <---> (my public ip
> B.B.B.B) FIREWALL <----> (my private ip C.C.C.C) IPBX ALCATEL
>
>        During a call, after invite or reinvite, Freeswitch send me RTP
> DATA on wrong RTP port...
>
>        For sample:
>            I receive a call, after invite from FS and OK from my IPBX, we
> have a communication who is good. We hear on the two sides.
>                FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ----------------
> C.C.C.C:RTPC1 <> IPBX
>
>            If i need to hold the call, all is ok. My holding music is
> hearing.
>            FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB1 ----------------
> C.C.C.C:RTPC1 <> IPBX
>            FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  --------------
> C.C.C.C:SIPC1 <-- IPBX (INVITE)
>           (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 --------------
> C.C.C.C:SIPC1 --> IPBX
>            FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ----------------
> C.C.C.C:RTPC2 <> IPBX
>
>
>            But when i unhold the call, "randomly", FS send me RTP DATA on
> the RTP port used before the unhold.. And we have no sound on the two
> sides...
>
>            Case good:
>            FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ----------------
> C.C.C.C:RTPC2 <> IPBX
>            FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  --------------
> C.C.C.C:SIPC1 <-- IPBX (INVITE)
>           (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 --------------
> C.C.C.C:SIPC1 --> IPBX
>            FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ----------------
> C.C.C.C:RTPC3 <> IPBX
>
>
>            Case bad:
>            FS<-> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ----------------
> C.C.C.C:RTPC2 <> IPBX
>            FS<-- A.A.A.A:SIPA1 -------- B.B.B.B:SIPB1  --------------
> C.C.C.C:SIPC1 <-- IPBX (INVITE)
>           (OK) FS --> A.A.A.A:SIPA1 ------- B.B.B.B:SIPB1 --------------
> C.C.C.C:SIPC1 --> IPBX
>            FS<-- A.A.A.A:RTPA1 -------- B.B.B.B:RTPB3 ----------------
> C.C.C.C:RTPC3 <-- IPBX
>            FS--> A.A.A.A:RTPA1 -------- B.B.B.B:RTPB2 ----------------
> C.C.C.C:RTPC2 --> IPBX
>
>
> I don't understand why the rtp port is not good updating by freeswitch
> randomly. I know freeswitch don't use the port that my IPBX give in INVITE
> packet because it's not the good with the nat from my firewall. I suppose
> that's the "NDLB-connectile-dysfunction" who is used by freeswitch to find
> the good rtp port.
>
> If with a little congestion, Freeswitch receive after my reinvite, one or
> two RTP packet from previous RTP DATA, Freeswitch can use the wrong RTP
> port? Is there a parameter to ignore after reinvite the five first RTP
> packet to autoadjust the RTP port?
>
> Thanks to all helping and sorry for my poor english.
>
> Best 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 devang.nathwani31589 at gmail.com  Fri Jan 13 09:04:01 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Fri, 13 Jan 2017 11:34:01 +0530
Subject: [Freeswitch-users] issue with disposition cause overriding
Message-ID: <CAGOm0PLfhJJFKkYb16Z+sn0Jfn4=w6yCcgz+=1iY=aFf4G_Y+A@mail.gmail.com>

i am trying to override specific disposition cause coming from leg B and
sending the modified cause to leg A

using,
https://freeswitch.org/confluence/display/FREESWITCH/Cause+Code+Substitution+Example

the lua file returns
sip_invite_failure_status variable as '-ERR No such channel!' and not the
code 480
However, from sip trace i can see '480 temporary unavailable' from leg B
and because of that my lua script which i am calling before bridge
application is not able to override the cause from 480 to 503
What i am missing?
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From igorolhovskiy at gmail.com  Fri Jan 13 12:18:25 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 13 Jan 2017 11:18:25 +0200
Subject: [Freeswitch-users] Profile misunderstanding
Message-ID: <30c9da9c-d523-4c49-acec-e38b83b10bbe@Spark>

Hi!
I have profile named default:
sofia status profile default
=================================================================================================
Name? ? ? ? ? ? ?? ?default
Domain Name? ? ? ? ?N/A
Auto-NAT? ? ? ? ?? ?false
DBName? ? ? ? ? ?? ?sofia_reg_default
Pres Hosts? ? ? ?? ?172.31.1.100,172.31.1.100
Dialplan? ? ? ? ?? ?XML
Context? ? ? ? ? ? ?default
Challenge Realm? ? ?auto_from
RTP-IP? ? ? ? ? ?? ?172.31.1.100
Ext-RTP-IP ? ? ? ? ?<EXT_ADDR>
SIP-IP? ? ? ? ? ?? ?172.31.1.100
Ext-SIP-IP? ? ? ?? ?<EXT_ADDR>
URL? ? ? ? ? ? ? ? ?sip:mod_sofia@<EXT_ADDR>:5060
BIND-URL? ? ? ? ?? ?sip:mod_sofia@<EXT_ADDR>:5060;maddr=172.31.1.100;transport=udp,tcp
HOLD-MUSIC? ? ? ?? ?local_stream://moh
OUTBOUND-PROXY? ?? ?N/A
CODECS IN? ? ? ? ? ?PCMA,PCMU,G729
CODECS OUT? ? ? ?? ?PCMA,PCMU,G729
TEL-EVENT? ? ? ? ? ?101
DTMF-MODE? ? ? ? ? ?rfc2833
CNG? ? ? ? ? ? ? ? ?13
SESSION-TO? ? ? ?? ?0
MAX-DIALOG? ? ? ?? ?0
NOMEDIA? ? ? ? ? ? ?false
LATE-NEG? ? ? ? ?? ?false
PROXY-MEDIA? ? ? ? ?false
ZRTP-PASSTHRU? ? ? ?false
AGGRESSIVENAT? ? ? ?true


Also:

sip-ip [172.31.1.100]
sip-port [5060]
rtp-ip [172.31.1.100]
dialplan [XML]
user-agent-string [ASTPP]
debug [0]
sip-trace [no]
tls [false]
inbound-reg-force-matching-username [true]
disable-transcoding [true]
all-reg-options-ping [false]
unregister-on-options-fail [true]
log-auth-failures [true]
status [0]
inbound-bypass-media [false]
inbound-proxy-media [false]
disable-transfer [true]
enable-100rel [false]
rtp-timeout-sec [400]
dtmf-duration [2000]
manual-redirect [true]
aggressive-nat-detection [true]
enable-timer [false]
minimum-session-expires [120]
session-timeout-pt [1800]
auth-calls [true]
apply-inbound-acl [default]
inbound-codec-prefs [PCMA,PCMU,G729]
outbound-codec-prefs [PCMA,PCMU,G729]
inbound-late-negotiation [false]
sip-capture [no]
forward-unsolicited-mwi-notify [false]
context [default]
rfc2833-pt [101]
rtp-timer-name [soft]
hold-music [local_stream://moh]
manage-presence [true]
presence-hosts [172.31.1.100,172.31.1.100]
presence-privacy [false]
inbound-codec-negotiation [generous]
auth-all-packets [false]
ext-rtp-ip [<EXT_ADDR>]
ext-sip-ip [<EXT_ADDR>]
rtp-hold-timeout-sec [1800]
force-register-domain [172.31.1.100]
force-subscription-domain [172.31.1.100]
force-register-db-domain [172.31.1.100]
challenge-realm [auto_from]
nonce-ttl [60]
pass-callee-id [false]
rtcp-audio-interval-msec [5000]
local-network-acl [localnet.auto]
NDLB-force-rport [true]

But problem is, when client tries to connect to FS, it still answers to port 5060, not looking on NDLB-force-rport option. As well, as private addresses shown across packets, not using ext-sip and ext-rtp options.

------------------------------------------------------------------------
recv 378 bytes from udp/[136.169.20.219]:3589?at 19:54:34.619718:
------------------------------------------------------------------------
REGISTER sip:172.31.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305
From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
To: <sip:6399081327 at 172.31.1.100:5060>
Call-ID:?1500202051 at 192.168.0.101
CSeq: 197 REGISTER
Contact: <sip:6399081327 at 192.168.0.101:5060>;expires=5
Max-Forwards: 30
User-Agent: dble
Expires: 5
Content-Length: 0

------------------------------------------------------------------------
send 571 bytes to udp/[136.169.20.219]:5060?at 19:54:34.619864:
------------------------------------------------------------------------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305;received=136.169.20.219
From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
To: <sip:6399081327 at 172.31.1.100:5060>;tag=jD7SQFyXcK1gr
Call-ID:?1500202051 at 192.168.0.101
CSeq: 197 REGISTER
User-Agent: ASTPP
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
WWW-Authenticate: Digest realm="172.31.1.100", nonce="7d83af94-2a69-40cc-a9c3-1659945a34b2", algorithm=MD5, qop="auth"
Content-Length: 0


Can you please point what I?m missing?
Thanks!

Regards, Igor
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From s.safarov at gmail.com  Fri Jan 13 12:30:46 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 13 Jan 2017 09:30:46 +0000
Subject: [Freeswitch-users] Profile misunderstanding
In-Reply-To: <30c9da9c-d523-4c49-acec-e38b83b10bbe@Spark>
References: <30c9da9c-d523-4c49-acec-e38b83b10bbe@Spark>
Message-ID: <CAHtxdDcCotK1uZbyZegpWi3jnCciT+j1MVojBj5SsQFaek0kOQ@mail.gmail.com>

First required to fix
Ext-SIP-IP          <EXT_ADDR>
To do it edit "default" profile setting and set ext-sip-ip and ext-rtp-ip
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#284-285>
to
ip address values.


??, 13 ???. 2017 ?. ? 12:22, Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Hi!
> I have profile named default:
> sofia status profile default
>
> =================================================================================================
> Name                default
> Domain Name         N/A
> Auto-NAT            false
> DBName              sofia_reg_default
> Pres Hosts          172.31.1.100,172.31.1.100
> Dialplan            XML
> Context             default
> Challenge Realm     auto_from
> RTP-IP              172.31.1.100
> Ext-RTP-IP          <EXT_ADDR>
> SIP-IP              172.31.1.100
> Ext-SIP-IP          <EXT_ADDR>
> URL                 sip:mod_sofia@<EXT_ADDR>:5060
> BIND-URL            sip:mod_sofia@
> <EXT_ADDR>:5060;maddr=172.31.1.100;transport=udp,tcp
> HOLD-MUSIC          local_stream://moh
> OUTBOUND-PROXY      N/A
> CODECS IN           PCMA,PCMU,G729
> CODECS OUT          PCMA,PCMU,G729
> TEL-EVENT           101
> DTMF-MODE           rfc2833
> CNG                 13
> SESSION-TO          0
> MAX-DIALOG          0
> NOMEDIA             false
> LATE-NEG            false
> PROXY-MEDIA         false
> ZRTP-PASSTHRU       false
> AGGRESSIVENAT       true
>
>
> Also:
>
> sip-ip [172.31.1.100]
> sip-port [5060]
> rtp-ip [172.31.1.100]
> dialplan [XML]
> user-agent-string [ASTPP]
> debug [0]
> sip-trace [no]
> tls [false]
> inbound-reg-force-matching-username [true]
> disable-transcoding [true]
> all-reg-options-ping [false]
> unregister-on-options-fail [true]
> log-auth-failures [true]
> status [0]
> inbound-bypass-media [false]
> inbound-proxy-media [false]
> disable-transfer [true]
> enable-100rel [false]
> rtp-timeout-sec [400]
> dtmf-duration [2000]
> manual-redirect [true]
> aggressive-nat-detection [true]
> enable-timer [false]
> minimum-session-expires [120]
> session-timeout-pt [1800]
> auth-calls [true]
> apply-inbound-acl [default]
> inbound-codec-prefs [PCMA,PCMU,G729]
> outbound-codec-prefs [PCMA,PCMU,G729]
> inbound-late-negotiation [false]
> sip-capture [no]
> forward-unsolicited-mwi-notify [false]
> context [default]
> rfc2833-pt [101]
> rtp-timer-name [soft]
> hold-music [local_stream://moh]
> manage-presence [true]
> presence-hosts [172.31.1.100,172.31.1.100]
> presence-privacy [false]
> inbound-codec-negotiation [generous]
> auth-all-packets [false]
> ext-rtp-ip [<EXT_ADDR>]
> ext-sip-ip [<EXT_ADDR>]
> rtp-hold-timeout-sec [1800]
> force-register-domain [172.31.1.100]
> force-subscription-domain [172.31.1.100]
> force-register-db-domain [172.31.1.100]
> challenge-realm [auto_from]
> nonce-ttl [60]
> pass-callee-id [false]
> rtcp-audio-interval-msec [5000]
> local-network-acl [localnet.auto]
> NDLB-force-rport [true]
>
> But problem is, when client tries to connect to FS, it still answers to
> port 5060, not looking on NDLB-force-rport option. As well, as private
> addresses shown across packets, not using ext-sip and ext-rtp options.
>
> ------------------------------------------------------------------------
> recv 378 bytes from udp/[136.169.20.219]:*3589* at 19:54:34.619718:
> ------------------------------------------------------------------------
> REGISTER sip:172.31.1.100:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305
> From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> To: <sip:6399081327 at 172.31.1.100:5060>
> Call-ID: 1500202051 at 192.168.0.101
> CSeq: 197 REGISTER
> Contact: <sip:6399081327 at 192.168.0.101:5060>;expires=5
> Max-Forwards: 30
> User-Agent: dble
> Expires: 5
> Content-Length: 0
>
> ------------------------------------------------------------------------
> send 571 bytes to udp/[136.169.20.219]:*5060* at 19:54:34.619864:
> ------------------------------------------------------------------------
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.101:5060
> ;branch=z9hG4bK322184305;received=136.169.20.219
> From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> To: <sip:6399081327 at 172.31.1.100:5060>;tag=jD7SQFyXcK1gr
> Call-ID: 1500202051 at 192.168.0.101
> CSeq: 197 REGISTER
> User-Agent: ASTPP
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> NOTIFY, PUBLISH, SUBSCRIBE
> Supported: path, replaces
> WWW-Authenticate: Digest realm="172.31.1.100",
> nonce="7d83af94-2a69-40cc-a9c3-1659945a34b2", algorithm=MD5, qop="auth"
> Content-Length: 0
>
>
> Can you please point what I?m missing?
> Thanks!
>
> Regards, Igor
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From igorolhovskiy at gmail.com  Fri Jan 13 13:50:02 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 13 Jan 2017 12:50:02 +0200
Subject: [Freeswitch-users] Profile misunderstanding
In-Reply-To: <CAHtxdDcCotK1uZbyZegpWi3jnCciT+j1MVojBj5SsQFaek0kOQ@mail.gmail.com>
References: <30c9da9c-d523-4c49-acec-e38b83b10bbe@Spark>
	<CAHtxdDcCotK1uZbyZegpWi3jnCciT+j1MVojBj5SsQFaek0kOQ@mail.gmail.com>
Message-ID: <9b1912bb-86b4-43fa-b3ce-c2e417887f1d@Spark>


<EXT_ADDR>? ?is a real external address, just hidden here :)

Regards, Igor

On 13 ???. 2017 ?., 11:31 +0200, Sergey Safarov <s.safarov at gmail.com>, wrote:
> First required to fix
> Ext-SIP-IP? ? ? ?? ?<EXT_ADDR>
> To do it edit "default" profile setting and set?ext-sip-ip and ext-rtp-ip?to ip address values.
>
>
> > ??, 13 ???. 2017 ?. ? 12:22, Igor Olhovskiy <igorolhovskiy at gmail.com>:
> > > Hi!
> > > I have profile named default:
> > > sofia status profile default
> > > =================================================================================================
> > > Name? ? ? ? ? ? ?? ?default
> > > Domain Name? ? ? ? ?N/A
> > > Auto-NAT? ? ? ? ?? ?false
> > > DBName? ? ? ? ? ?? ?sofia_reg_default
> > > Pres Hosts? ? ? ?? ?172.31.1.100,172.31.1.100
> > > Dialplan? ? ? ? ?? ?XML
> > > Context? ? ? ? ? ? ?default
> > > Challenge Realm? ? ?auto_from
> > > RTP-IP? ? ? ? ? ?? ?172.31.1.100
> > > Ext-RTP-IP ? ? ? ? ?<EXT_ADDR>
> > > SIP-IP? ? ? ? ? ?? ?172.31.1.100
> > > Ext-SIP-IP? ? ? ?? ?<EXT_ADDR>
> > > URL? ? ? ? ? ? ? ? ?sip:mod_sofia@<EXT_ADDR>:5060
> > > BIND-URL? ? ? ? ?? ?sip:mod_sofia@<EXT_ADDR>:5060;maddr=172.31.1.100;transport=udp,tcp
> > > HOLD-MUSIC? ? ? ?? ?local_stream://moh
> > > OUTBOUND-PROXY? ?? ?N/A
> > > CODECS IN? ? ? ? ? ?PCMA,PCMU,G729
> > > CODECS OUT? ? ? ?? ?PCMA,PCMU,G729
> > > TEL-EVENT? ? ? ? ? ?101
> > > DTMF-MODE? ? ? ? ? ?rfc2833
> > > CNG? ? ? ? ? ? ? ? ?13
> > > SESSION-TO? ? ? ?? ?0
> > > MAX-DIALOG? ? ? ?? ?0
> > > NOMEDIA? ? ? ? ? ? ?false
> > > LATE-NEG? ? ? ? ?? ?false
> > > PROXY-MEDIA? ? ? ? ?false
> > > ZRTP-PASSTHRU? ? ? ?false
> > > AGGRESSIVENAT? ? ? ?true
> > >
> > >
> > > Also:
> > >
> > > sip-ip [172.31.1.100]
> > > sip-port [5060]
> > > rtp-ip [172.31.1.100]
> > > dialplan [XML]
> > > user-agent-string [ASTPP]
> > > debug [0]
> > > sip-trace [no]
> > > tls [false]
> > > inbound-reg-force-matching-username [true]
> > > disable-transcoding [true]
> > > all-reg-options-ping [false]
> > > unregister-on-options-fail [true]
> > > log-auth-failures [true]
> > > status [0]
> > > inbound-bypass-media [false]
> > > inbound-proxy-media [false]
> > > disable-transfer [true]
> > > enable-100rel [false]
> > > rtp-timeout-sec [400]
> > > dtmf-duration [2000]
> > > manual-redirect [true]
> > > aggressive-nat-detection [true]
> > > enable-timer [false]
> > > minimum-session-expires [120]
> > > session-timeout-pt [1800]
> > > auth-calls [true]
> > > apply-inbound-acl [default]
> > > inbound-codec-prefs [PCMA,PCMU,G729]
> > > outbound-codec-prefs [PCMA,PCMU,G729]
> > > inbound-late-negotiation [false]
> > > sip-capture [no]
> > > forward-unsolicited-mwi-notify [false]
> > > context [default]
> > > rfc2833-pt [101]
> > > rtp-timer-name [soft]
> > > hold-music [local_stream://moh]
> > > manage-presence [true]
> > > presence-hosts [172.31.1.100,172.31.1.100]
> > > presence-privacy [false]
> > > inbound-codec-negotiation [generous]
> > > auth-all-packets [false]
> > > ext-rtp-ip [<EXT_ADDR>]
> > > ext-sip-ip [<EXT_ADDR>]
> > > rtp-hold-timeout-sec [1800]
> > > force-register-domain [172.31.1.100]
> > > force-subscription-domain [172.31.1.100]
> > > force-register-db-domain [172.31.1.100]
> > > challenge-realm [auto_from]
> > > nonce-ttl [60]
> > > pass-callee-id [false]
> > > rtcp-audio-interval-msec [5000]
> > > local-network-acl [localnet.auto]
> > > NDLB-force-rport [true]
> > >
> > > But problem is, when client tries to connect to FS, it still answers to port 5060, not looking on NDLB-force-rport option. As well, as private addresses shown across packets, not using ext-sip and ext-rtp options.
> > >
> > > ------------------------------------------------------------------------
> > > recv 378 bytes from udp/[136.169.20.219]:3589?at 19:54:34.619718:
> > > ------------------------------------------------------------------------
> > > REGISTER sip:172.31.1.100:5060 SIP/2.0
> > > Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305
> > > From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> > > To: <sip:6399081327 at 172.31.1.100:5060>
> > > Call-ID:?1500202051 at 192.168.0.101
> > > CSeq: 197 REGISTER
> > > Contact: <sip:6399081327 at 192.168.0.101:5060>;expires=5
> > > Max-Forwards: 30
> > > User-Agent: dble
> > > Expires: 5
> > > Content-Length: 0
> > >
> > > ------------------------------------------------------------------------
> > > send 571 bytes to udp/[136.169.20.219]:5060?at 19:54:34.619864:
> > > ------------------------------------------------------------------------
> > > SIP/2.0 401 Unauthorized
> > > Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK322184305;received=136.169.20.219
> > > From: <sip:6399081327 at 172.31.1.100:5060>;tag=1569039960
> > > To: <sip:6399081327 at 172.31.1.100:5060>;tag=jD7SQFyXcK1gr
> > > Call-ID:?1500202051 at 192.168.0.101
> > > CSeq: 197 REGISTER
> > > User-Agent: ASTPP
> > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
> > > Supported: path, replaces
> > > WWW-Authenticate: Digest realm="172.31.1.100", nonce="7d83af94-2a69-40cc-a9c3-1659945a34b2", algorithm=MD5, qop="auth"
> > > Content-Length: 0
> > >
> > >
> > > Can you please point what I?m missing?
> > > Thanks!
> > >
> > > Regards, Igor
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vivek at advaitamtech.com  Fri Jan 13 15:57:02 2017
From: vivek at advaitamtech.com (vivek at advaitamtech.com)
Date: Fri, 13 Jan 2017 18:27:02 +0530 (IST)
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
	Authentication Required"
In-Reply-To: <CAHzfiZ5+sJKX6tgkqKxVSMfg2PkzPf0CPtYisMsAHC7hAbXP9Q@mail.gmail.com>
References: <1483957702.686832416@apps.rackspace.com> 
	<1484198408.261517009@apps.rackspace.com> 
	<CAHzfiZ5+sJKX6tgkqKxVSMfg2PkzPf0CPtYisMsAHC7hAbXP9Q@mail.gmail.com>
Message-ID: <1484312222.50169111@apps.rackspace.com>


Hi Russell,
 
Thanks a lot for the hint :)
 
I configured the gateway to handle situation.
Right now Freeswitch is handling the "407 Proxy Authentication Required" and replying the ACK accordingly.
 
But the new INVITE sent to the third party server resulted in "403 Forbidden".
 
According to the analysis new INVITE has correct "realm" and "nonce" parameter which is received through "407 Proxy Authentication Required".
 
Any hint on this?
 
Thanks,
Vivek.
 
-----Original Message-----
From: "Russell Treleaven" <rtreleaven at bunnykick.ca>
Sent: Thursday, 12 January, 2017 11:22am
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy Authentication Required"



Freeswitch is not a proxy it's a b2bua. Google Sofia gateway, it might be what you need.


On Jan 12, 2017 12:20 AM, "[ vivek at advaitamtech.com ]( mailto:vivek at advaitamtech.com )" <[ vivek at advaitamtech.com ]( mailto:vivek at advaitamtech.com )> wrote:

Hi,
 
Any help on this?
 
 
-----Original Message-----
From: "[ vivek at advaitamtech.com ]( mailto:vivek at advaitamtech.com )" <[ vivek at advaitamtech.com ]( mailto:vivek at advaitamtech.com )>
Sent: Monday, 9 January, 2017 3:58pm
To: [ freeswitch-users at lists.freeswitch.org ]( mailto:freeswitch-users at lists.freeswitch.org )
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy Authentication Required"



Hi All,
 
Am trying to proxy the "407 Proxy Authentication Required" through freeswitch,
 
My Objective is to proxy the "407 Proxy Authentication Required" received from the third party server to another server which is connected to freeswitch.
 
Is it possible to do this?. If possible how do I do it. Please help me out.
 
Thanks,
Vivek.
_________________________________________________________________________
 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 shaun.stokes at itec-support.co.uk  Fri Jan 13 17:39:33 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 13 Jan 2017 14:39:33 +0000
Subject: [Freeswitch-users] Configuring FreeSWITCH for federation with other
	platforms
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8626A09@mbx-01.sysconfig.co.uk>

Hi All,

Has anyone tried or have any guidance on configuring FreeSWITCH for federation with other platforms (i.e. via XMPP using DNS)?

I've looked at SylkServer but as far as I can tell it's a closed platform on SIP2SIP\SIP Thor.

Perhaps this might be possible using Openfire integration with FreeSWITCH?

Thanks,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

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From krice at freeswitch.org  Fri Jan 13 18:15:48 2017
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 13 Jan 2017 09:15:48 -0600
Subject: [Freeswitch-users] Configuring FreeSWITCH for federation with
	other	platforms
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8626A09@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8626A09@mbx-01.sysconfig.co.uk>
Message-ID: <053d01d26daf$ec397aa0$c4ac6fe0$@freeswitch.org>

FreeSWITCH has an XMPP end point built in, its called mod_dingaling. however
FS is not an XMPP server.  

 

But then again SIP itself lends itself well to federation..

 

The real question is what is your end goal for this federation setup?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun
Stokes
Sent: Friday, January 13, 2017 8:40 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Configuring FreeSWITCH for federation with other
platforms

 

Hi All,

 

Has anyone tried or have any guidance on configuring FreeSWITCH for
federation with other platforms (i.e. via XMPP using DNS)?

 

I've looked at SylkServer but as far as I can tell it's a closed platform on
SIP2SIP\SIP Thor.

 

Perhaps this might be possible using Openfire integration with FreeSWITCH?

 

Thanks,

Shaun


  <http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg> 

Shaun Stokes - Infrastructure Analyst

		

T : 

01453 700713


E : 

shaun.stokes at itec-support.co.uk <mailto:shaun.stokes at itec-support.co.uk> 


W : 

www.itec-support.co.uk <http://www.itec-support.co.uk> 

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
Stroud, Gloucestershire GL5 3QF 
Company No. 06908001 


CONFIDENTIALITY NOTICE 
This communication and the information it contains are intended for the
person or organisation to which it is addressed. Its contents are
confidential and may be protected in law. Unauthorised use, copying or
disclosure of any of it may be unlawful. If you are not the intended
recipient, please contact us immediately. 
The contents of any attachments in this e-mail may contain software viruses,
which could damage your own computer system. While ITEC Support has taken
every reasonable precaution to minimise this risk, we cannot accept
liability for any damage which you sustain as a result of software viruses.
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From mike at jerris.com  Fri Jan 13 18:25:34 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 13 Jan 2017 15:25:34 +0000
Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
 Authentication Required"
In-Reply-To: <1484312222.50169111@apps.rackspace.com>
References: <1483957702.686832416@apps.rackspace.com>
	<1484198408.261517009@apps.rackspace.com>
	<CAHzfiZ5+sJKX6tgkqKxVSMfg2PkzPf0CPtYisMsAHC7hAbXP9Q@mail.gmail.com>
	<1484312222.50169111@apps.rackspace.com>
Message-ID: <CAEwTsAgMMN4Mg1QQkBE2zj=Joqy4pgv2E8WKttVJga_5JNs81g@mail.gmail.com>

are you trying to pass the call through so the devices on either end are
doing auth to each other, and freeswitch just passes these along and
doesn't try to authenticate itself?'


On Fri, Jan 13, 2017 at 8:00 AM vivek at advaitamtech.com <
vivek at advaitamtech.com> wrote:

> Hi Russell,
>
>
>
> Thanks a lot for the hint :)
>
>
>
> I configured the gateway to handle situation.
>
> Right now Freeswitch is handling the "407 Proxy Authentication Required"
> and replying the ACK accordingly.
>
>
>
> But the new INVITE sent to the third party server resulted in "403
> Forbidden".
>
>
>
> According to the analysis new INVITE has correct "realm" and "nonce"
> parameter which is received through "407 Proxy Authentication Required".
>
>
>
> Any hint on this?
>
>
>
> Thanks,
>
> Vivek.
>
>
>
> -----Original Message-----
> From: "Russell Treleaven" <rtreleaven at bunnykick.ca>
> Sent: Thursday, 12 January, 2017 11:22am
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
> Authentication Required"
>
>
>
> Freeswitch is not a proxy it's a b2bua. Google Sofia gateway, it might be
> what you need.
>
>
>
> On Jan 12, 2017 12:20 AM, "vivek at advaitamtech.com" <vivek at advaitamtech.com>
> wrote:
>
>
> Hi,
>
>
>
> Any help on this?
>
>
>
>
>
> -----Original Message-----
> From: "vivek at advaitamtech.com" <vivek at advaitamtech.com>
> Sent: Monday, 9 January, 2017 3:58pm
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] FreeSwitch as proxy for "407 Proxy
> Authentication Required"
>
>
>
> Hi All,
>
>
>
> Am trying to proxy the "407 Proxy Authentication Required" through
> freeswitch,
>
>
>
> My Objective is to proxy the "407 Proxy Authentication Required" received
> from the third party server to another server which is connected to
> freeswitch.
>
>
>
> Is it possible to do this?. If possible how do I do it. Please help me out.
>
>
>
> Thanks,
>
> Vivek.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
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From shaun.stokes at itec-support.co.uk  Fri Jan 13 18:40:55 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 13 Jan 2017 15:40:55 +0000
Subject: [Freeswitch-users] Configuring FreeSWITCH for federation
	with	other	platforms
In-Reply-To: <053d01d26daf$ec397aa0$c4ac6fe0$@freeswitch.org>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8626A09@mbx-01.sysconfig.co.uk>
	<053d01d26daf$ec397aa0$c4ac6fe0$@freeswitch.org>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8626AFF@mbx-01.sysconfig.co.uk>

We'd like to support federation with other platforms (IM, presence, VoIP) over the public internet such as Lync\Skype for Business and others. We're in the process of moving away from Lync which we are replacing with the Jitsi softphone client and FreeSWITCH.

I think the answer may be to use reSIProcate and repro (or Kamailio) with FreeSWITCH as an SBC\Proxy:
http://www.resiprocate.org/Using_reSIProcate_and_repro_for_Federated_VoIP


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: 13 January 2017 15:16
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Configuring FreeSWITCH for federation with other platforms

FreeSWITCH has an XMPP end point built in, its called mod_dingaling... however FS is not an XMPP server...

But then again SIP itself lends itself well to federation....

The real question is what is your end goal for this federation setup?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes
Sent: Friday, January 13, 2017 8:40 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Configuring FreeSWITCH for federation with other platforms

Hi All,

Has anyone tried or have any guidance on configuring FreeSWITCH for federation with other platforms (i.e. via XMPP using DNS)?

I've looked at SylkServer but as far as I can tell it's a closed platform on SIP2SIP\SIP Thor.

Perhaps this might be possible using Openfire integration with FreeSWITCH?

Thanks,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]

Shaun Stokes - Infrastructure Analyst

T :

01453 700713

E :

shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>

W :

www.itec-support.co.uk<http://www.itec-support.co.uk>


Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
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From naveen.khanna.bm at gmail.com  Mon Jan 16 08:17:47 2017
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Mon, 16 Jan 2017 10:47:47 +0530
Subject: [Freeswitch-users] Need starting point to Develop QSIG on Freeswitch
Message-ID: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>

I understand that QSIG is not supported on Freeswitch, I would need some
inputs as how this can be achieved in Freeswitch environment.


Regards,

Naveen Khanna
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From nb3868 at yp.com  Mon Jan 16 08:22:35 2017
From: nb3868 at yp.com (Nagendra Babu)
Date: Mon, 16 Jan 2017 05:22:35 +0000
Subject: [Freeswitch-users] load multiple grammar
In-Reply-To: <819781D671CC1D44944772177554935003C4C2E3@ASH-EXCH-MB06.corp.yp.com>
References: <819781D671CC1D44944772177554935003C4C2E3@ASH-EXCH-MB06.corp.yp.com>
Message-ID: <819781D671CC1D44944772177554935003C4C5B3@ASH-EXCH-MB06.corp.yp.com>

Hi All,

Any help on the ASR multi-grammar loading...

Thanks and regards,
Nagendra

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nagendra Babu
Sent: Thursday, January 12, 2017 10:50 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] load multiple grammar

Hi,

I need help regarding loading and activating multiple grammars.

Per FS-2906<https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.org_jira_browse_FS-2D2906&d=DgMFAg&c=lXkdEK1PC7UK9oKA-BBSI8p1AamzLOSncm6Vfn0C_UQ&r=4vn6GyL2RTvgDexCyKjXiQ&m=gm7PE2iEUDFcjA5CHF38lzeV6rA5-HvzuoSI7c4ByL8&s=Y4x0U8q_g_Po-H23qSYRrgM7PmcIz7m5lvGxJj7aNqk&e=>, it is possible to load multiple grammars from FreeSwitch.

I have tried the same using following commands from my C# code. The FS version I have is 1.3.17.

InboundSession.Execute("detect_speech", "unimrcp {start-recognize=false}yesnod yesnod");
InboundSession.Execute("detect_speech", "unimrcp {start-recognize=true}yesno yesno");

I can see that there are two DEFINE-GRAMMAR requests sent to the LumenVox Speech Server. One with yesnod.gram and another with yesno.gram
But there is only one session mentioned in the RECOGNIZE request where as I am expecting it to be having both the sessions described in the DEFINE-GRAMMAR request.
Any idea?


                MRCP/2.0 226 RECOGNIZE 3
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: text/uri-list
                Cancel-If-Queue: false
                Start-Input-Timers: false
                Speech-Complete-Timeout: 3000
                Content-Length: 13

                session:yesno
>>>> There should be session:yesnod also.

I do not see a way to do this in detect_speech.

Any help would be great.



Here is the define grammar message too...

                MRCP/2.0 318 DEFINE-GRAMMAR 1
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: application/srgs
                Content-Id: yesnod
                Content-Length: 158

                #ABNF 1.0 UTF-8;
language en-US;
mode dtmf;
tag-format <semantics/1.0>;

root $yesorno;

$yes = 1 {out = "yes"};
$no = 2 {out = "no"};
$yesorno = $yes | $no;

01/12/2017 07:16:09.790,RECV,OnLoadGrammarRe,Received cseq grammar load (1) reply for CallIndex 51
01/12/2017 07:16:09.790,SEND,SendPacket     ,

                MRCP/2.0 116 1 200 COMPLETE
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Completion-Cause: 000 success


01/12/2017 07:16:10.123,RECV,ProcessRcvdPkt ,Got MRCP Message -

                MRCP/2.0 408 DEFINE-GRAMMAR 2
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: application/srgs
                Content-Id: yesno
                Content-Length: 249

                #ABNF 1.0 UTF-8;
                language en-US; //use the American English pronunciation dictionary.
                mode voice;  //the input for this grammar will be spoken words.
                tag-format <semantics/1.0>;

                root $yesorno;

                $yes = yes;
                $no = no;
                $yesorno = $yes | $no;

01/12/2017 07:16:10.128,RECV,OnLoadGrammarRe,Received cseq grammar load (2) reply for CallIndex 51
01/12/2017 07:16:10.129,SEND,SendPacket     ,

                MRCP/2.0 116 2 200 COMPLETE
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Completion-Cause: 000 success


01/12/2017 07:16:10.459,RECV,ProcessRcvdPkt ,Got MRCP Message -

                MRCP/2.0 226 RECOGNIZE 3
                Channel-Identifier: 115483E487508175CFDF at speechrecog
                Content-Type: text/uri-list
                Cancel-If-Queue: false
                Start-Input-Timers: false
                Speech-Complete-Timeout: 3000
                Content-Length: 13

                session:yesno


01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Speech-Complete-Timeout' with value '3000'
01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Start-Input-Timers' with value 'false'
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting speech_complete_timeout to 3000 ms
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting sensitivity level to 50
01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting Speed vs Accuracy to 0.5
01/12/2017 07:16:10.461,SEND,SendPacket     ,

                MRCP/2.0 87 3 200 IN-PROGRESS




Thanks and regards,
Nagendra

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From david.villasmil.work at gmail.com  Mon Jan 16 12:33:13 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 16 Jan 2017 09:33:13 +0000
Subject: [Freeswitch-users] Need starting point to Develop QSIG on
	Freeswitch
In-Reply-To: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
References: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
Message-ID: <CAFGRPVrtxE2tzX3QnL2uDmLJi_rCJMWdwkUWDezHu5gaLvX12A@mail.gmail.com>

Mod_skel and
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/6587713
is a good start :)
On Mon, Jan 16, 2017 at 6:19 AM Naveen Khanna <naveen.khanna.bm at gmail.com>
wrote:

> I understand that QSIG is not supported on Freeswitch, I would need some
> inputs as how this can be achieved in Freeswitch environment.
>
>
> 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 miha at softnet.si  Mon Jan 16 13:05:56 2017
From: miha at softnet.si (Miha)
Date: Mon, 16 Jan 2017 11:05:56 +0100
Subject: [Freeswitch-users] issue with Acct-Status-Type
Message-ID: <06c1edb5-42dd-b7ec-5d71-0d6f09af5c36@softnet.si>

Hello

i need a little help.

I am getting in Acct-Status-Type = 0 and due to this "0" i get this error:

softswitch_rs: EXPAND %{tolower:type.%{Acct-Status-Type}.query}
(39) softswitch_rs:    --> type.0.query
No such configuration item .type.0.query

Reference is define like this:
reference = "%{tolower:type.%{Acct-Status-Type}.query}


How can i set in default (sites-available) that ifAcct-Status-Type = 0 i set it to Stop so that the same confg will be 
like for "Stop". Tnx Miha


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From miha at softnet.si  Mon Jan 16 13:14:03 2017
From: miha at softnet.si (Miha)
Date: Mon, 16 Jan 2017 11:14:03 +0100
Subject: [Freeswitch-users] issue with Acct-Status-Type
In-Reply-To: <06c1edb5-42dd-b7ec-5d71-0d6f09af5c36@softnet.si>
References: <06c1edb5-42dd-b7ec-5d71-0d6f09af5c36@softnet.si>
Message-ID: <378c7d20-15fd-2377-c266-4c74cf7a3d1b@softnet.si>

hey

sorry, wrong group :)


miha

On 16/01/2017 11:05, Miha wrote:
> Hello
>
> i need a little help.
>
> I am getting in Acct-Status-Type = 0 and due to this "0" i get this error:
>
> softswitch_rs: EXPAND %{tolower:type.%{Acct-Status-Type}.query}
> (39) softswitch_rs:    --> type.0.query
> No such configuration item .type.0.query
>
> Reference is define like this:
> reference = "%{tolower:type.%{Acct-Status-Type}.query}
>
>
> How can i set in default (sites-available) that if Acct-Status-Type = 0 i set it to Stop so that the same confg will be like for "Stop".
>
>
>
> Tnx
> Miha

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From naveen.khanna.bm at gmail.com  Mon Jan 16 13:34:27 2017
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Mon, 16 Jan 2017 16:04:27 +0530
Subject: [Freeswitch-users] Need starting point to Develop QSIG on
	Freeswitch
In-Reply-To: <CAFGRPVrtxE2tzX3QnL2uDmLJi_rCJMWdwkUWDezHu5gaLvX12A@mail.gmail.com>
References: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
	<CAFGRPVrtxE2tzX3QnL2uDmLJi_rCJMWdwkUWDezHu5gaLvX12A@mail.gmail.com>
Message-ID: <CAE8fD+0H7go-kkuWpTeR2Js3qOZR+hd+ZQe-94g4epnCDGTMOA@mail.gmail.com>

Dear David,

Thanks!!!.

Since QSIG is integration of PBX over PRI.

So, I was wondering would it be better if

#1 Use Sangoma PRI Card and write a Freeswitch module or extend FreeTDM .

Or

#2 Use a media gateway which converts QSIG to SIP and write a module for
Freeswitch where I can handle SIP INFO for extended functionality.






Regards,

Naveen Khanna

On 16 January 2017 at 15:03, David Villasmil <david.villasmil.work at gmail.com
> wrote:

> Mod_skel and https://freeswitch.org/confluence/plugins/servlet/
> mobile#content/view/6587713 is a good start :)
> On Mon, Jan 16, 2017 at 6:19 AM Naveen Khanna <naveen.khanna.bm at gmail.com>
> wrote:
>
>> I understand that QSIG is not supported on Freeswitch, I would need some
>> inputs as how this can be achieved in Freeswitch environment.
>>
>>
>> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vbvbrj at gmail.com  Mon Jan 16 15:05:36 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Mon, 16 Jan 2017 14:05:36 +0200
Subject: [Freeswitch-users] caller_id_name variables question
Message-ID: <333dbbed-6c19-82b9-e263-f0bc5204183a@gmail.com>

Hello.

This is a snippet:

<extension name="set_caller_name_db" continue="true">
  <condition field="${user_exists(id ${caller_id_number} $${domain})" 
expression="^false$"/>
  <condition field="${effective_caller_id_number}" expression="^$">
   <action application="set" inline="true" 
data="effective_caller_id_name=${lua(get_name_from_db.lua 
${caller_id_number})}"/>
   <action application="set" inline="true" 
data="origination_caller_id_name=${effective_caller_id_name}"/>
   <action application="set" inline="true" 
data="caller_id_name=${effective_caller_id_name}"/>
   <anti-action application="set" inline="true" 
data="effective_caller_id_name=${lua(get_name_from_db.lua 
${effective_caller_id_number})}"/>
   <anti-action application="set" inline="true" 
data="origination_caller_id_name=${effective_caller_id_name}"/>
   <anti-action application="set" inline="true" 
data="caller_id_name=${effective_caller_id_name}"/>
  </condition>
</extension>

If the caller is not in local users, then try to set caller_id_name from 
database. Script works ok.

My question is which of this three actions setting vars is redundant:
effective_caller_id_name, origination_caller_id_name, caller_id_name

Thank you


From asilva at wirelessmundi.com  Mon Jan 16 17:08:52 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Mon, 16 Jan 2017 15:08:52 +0100
Subject: [Freeswitch-users] caller_id_name variables question
In-Reply-To: <333dbbed-6c19-82b9-e263-f0bc5204183a@gmail.com>
References: <333dbbed-6c19-82b9-e263-f0bc5204183a@gmail.com>
Message-ID: <526ed062-b3b3-e54c-ead3-00d0dea00e93@wirelessmundi.com>

hi,

i think that effective_caller_id is deprecated in flavor of set_profile_var:

check: 
https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+set+profile+var


On 01/16/2017 01:05 PM, Mimiko wrote:
> Hello.
>
> This is a snippet:
>
> <extension name="set_caller_name_db" continue="true">
>    <condition field="${user_exists(id ${caller_id_number} $${domain})"
> expression="^false$"/>
>    <condition field="${effective_caller_id_number}" expression="^$">
>     <action application="set" inline="true"
> data="effective_caller_id_name=${lua(get_name_from_db.lua
> ${caller_id_number})}"/>
>     <action application="set" inline="true"
> data="origination_caller_id_name=${effective_caller_id_name}"/>
>     <action application="set" inline="true"
> data="caller_id_name=${effective_caller_id_name}"/>
>     <anti-action application="set" inline="true"
> data="effective_caller_id_name=${lua(get_name_from_db.lua
> ${effective_caller_id_number})}"/>
>     <anti-action application="set" inline="true"
> data="origination_caller_id_name=${effective_caller_id_name}"/>
>     <anti-action application="set" inline="true"
> data="caller_id_name=${effective_caller_id_name}"/>
>    </condition>
> </extension>
>
> If the caller is not in local users, then try to set caller_id_name from
> database. Script works ok.
>
> My question is which of this three actions setting vars is redundant:
> effective_caller_id_name, origination_caller_id_name, caller_id_name
>
> Thank you
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 

Saludos / Regards / Cumprimentos,
Ant?nio silva



From Paul.Mateer at outlook.com  Mon Jan 16 19:10:01 2017
From: Paul.Mateer at outlook.com (Paul Mateer)
Date: Mon, 16 Jan 2017 16:10:01 +0000
Subject: [Freeswitch-users] No input audio with FSClient
In-Reply-To: <CALn233qy3H4Jmx=e3Vemb4n70UN5ZosB_jOnfBKUVHjDCtDjJA@mail.gmail.com>
References: <AM5PR10MB0708C8FAEE4929D1F61B833AF0640@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>
	<CALn233o1wLPtzE9z6P27okxeSWkPep5hr=WvE=TNtyANGuZOow@mail.gmail.com>
	<CALn233pjfeXRxD9zqUnBLk6c+kq25zuUpBqHzKbmcoQRvTtsQA@mail.gmail.com>
	<AM5PR10MB0708DDC7C71A78B3124BF167F0670@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>,
	<CALn233qy3H4Jmx=e3Vemb4n70UN5ZosB_jOnfBKUVHjDCtDjJA@mail.gmail.com>
Message-ID: <AM5PR10MB07082830783E7A15AEAF3C62F07D0@AM5PR10MB0708.EURPRD10.PROD.OUTLOOK.COM>

Hi. Sorry for taking so long to post the logs but I got dragged into looking at something else and have only just gotten back to issue this now.


The log file for the FreeSWITCH  server is at https://pastebin.freeswitch.org/view/938cdb58 and the log file from the FSClient is at https://pastebin.freeswitch.org/view/b1255f02.


Both the server and the client had the following entries in their freeswitch.xml file:


        <X-PRE-PROCESS cmd="set" data="call_debug=true"/>
        <X-PRE-PROCESS cmd="set" data="console_loglevel=debug"/>


The server also had the following in the sip_profile file:

    <param name="debug" value="1"/>
    <param name="sip-trace" value="yes"/>
    <param name="sip-capture" value="yes"/>


The latest source code for both FreeSWITCH and the FSClient were downloaded on 21/12/16 in response to the following posting: http://lists.freeswitch.org/pipermail/freeswitch-users/2016-December/124117.html


Only some of the FSClient executables have file version numbers - the most relevant are probably that of FreeSwitch.dll (1.9.0.0), Freeswitch.Managed.dll (1.0.5.0) and FSClient.exe (1.2.3.5)


Regards,


Paul

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Mitch Capper <mitch.capper at gmail.com>
Sent: 11 January 2017 18:09:14
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] No input audio with FSClient

Can you post the logs to the FS pastbin that may be the easiest way to look at them.  fs_logger can anonymize the logs if needed.  Also we should probably get build versions and the logs should give us the codec settings.


~mitch

On Tue, Jan 10, 2017 at 3:18 AM, Paul Mateer <Paul.Mateer at outlook.com<mailto:Paul.Mateer at outlook.com>> wrote:

OK, so I amended the FreeSWITCH.xml for FSClient to add a record_session action before the bridge action of the "number" extension in the default context of the dialplan section, and the client records the input audio stream fine.

I did the same thing in the "echo" extension of the default.xml in the FreeSWITCH dialplan folder and I get a recording of silence. This would seem to fit in with the fact that I don't see any RTP traffic data from the client to the server.


I have logs from both the server and the client (Debug Level set to 9 with SIP Trace active) but there is nothing notable logged between the start of the call and it's termination (I can provide the logs if that's of any help - and a WireShark trace).


Paul


PS. Sorry about the mailing issue yesterday - I think I hit reply to all instead of just reply.


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Mitch Capper <mitch.capper at gmail.com<mailto:mitch.capper at gmail.com>>
Sent: 09 January 2017 20:44:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] No input audio with FSClient

Under options in FSClient for the event socket you can change them but otherwise just connect like normal debugging and sip_trace might be a good start,  you can use fs_logger as well to collect the logs for you if desired.  Let us know the result of the recording test as well.


~mitch

On Mon, Jan 9, 2017 at 9:08 AM, Mitch Capper <mitch.capper at gmail.com<mailto:mitch.capper at gmail.com>> wrote:
Hi Paul,
One place to start would be to record the call with FSClient and Freeswitch itself see if the audio is on the client but not FS.  Enabling logging on the client/server using fs_cli and see anything interesting there.  Finally make sure you have the right mic input selected in the FSClient options.




~mitch

On Mon, Jan 9, 2017 at 8:58 AM, Paul Mateer <paul.mateer at outlook.com<mailto:paul.mateer at outlook.com>> wrote:
I seem to have a problem with audio input when using FSClient.

I have one box running FreeSWITCH and another running FSClient. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back.

I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me.

I'm not sure if there is something amiss in the configuration of FSClient (although it should be the default config) or if something else is amiss (there doesn't appear to be anything odd in the FreeSWITCH log for the client).

Does anyone have any thoughts on what might be wrong, or what i should look at?

Thanks,

Paul

Sent from my Windows 10 phone


_________________________________________________________________________
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 tristan at mocha.net.nz  Mon Jan 16 06:43:31 2017
From: tristan at mocha.net.nz (Tristan Dean)
Date: Mon, 16 Jan 2017 16:43:31 +1300
Subject: [Freeswitch-users] T.38 parameters causing call rejections
Message-ID: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>

Hi All

 

I am having trouble on a new FreeSWITCH-based SBC we've built. We are going
through some SBCs and upgrading them. In this case, we are upgrading an old
SBC running: 1.4.15 64bit built from source.

 

The new version is running on Debian 8.3 (AMD64) Stable with the packaged
version of FreeSWITCH:  1.6.14 -23-e460bf8 64bit

We are tying this in with a hardware swap, so the 1.6.14-23 box is new.

 

We have a few instances where our upstream carrier presents us with a SDP
which contains:

m=image 56594 udptl t38

a=T38FaxUdpEC:t38UDPRedundancy

a=T38FaxRateManagement:transferredTCF

 

Historically (version 1.4.15) these seem to be ignored and FreeSWITCH passes
the call out obeying the absolute_codec_string set during bridge.
Transcoding is allowed and when negotiating a codec with the carrier, these
parameters were ignored and G.711a was usually negotiated on both legs.

 

On the new platform, FreeSWITCH seems to be attempting to parse these and
negotiate T.38. I am certain that it is something silly that I'm doing
wrong, but can't find it and would welcome others' thoughts. What have I
missed? Has anyone seen this before? I have tried t38_passthru=true to no
avail.

 

I can't set proxy_media=true or bypass_media=true as we need FreeSWITCH to
shield other devices from the SDP parameters above.

Most of the configuration is stored in a database and returned to FreeSWITCH
using mod_xml_curl. I have pasted some configuration excerpts, SIP messages
and logs in: https://pastebin.freeswitch.org/view/de059ffc

 

Please let me know if there is any other information you need from me.

 

Many thanks in advance for your assistance.

 

Tristan



---
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From kapilgupta1 at gmail.com  Mon Jan 16 14:35:59 2017
From: kapilgupta1 at gmail.com (Kapil Gupta)
Date: Mon, 16 Jan 2017 17:05:59 +0530
Subject: [Freeswitch-users] Need starting point to Develop QSIG on
	Freeswitch
In-Reply-To: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
References: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
Message-ID: <CAP=f5BJEoWtsqbGyJOXOA_6B=btNf+7VHMewohfkkaqZRP8yAQ@mail.gmail.com>

Hi Naveen,

FreeTDM has support for Sangoma PRI card which is present in
"ftmod_sangoma_isdn" module inside FreeTDM.

this module also supports QSIG protocol. Refer to below wiki to get more
idea about FreeTDM and ISDN modules.

https://wiki.freeswitch.org/wiki/FreeTDM

Regards
Kapil

On Mon, Jan 16, 2017 at 10:47 AM, Naveen Khanna <naveen.khanna.bm at gmail.com>
wrote:

> I understand that QSIG is not supported on Freeswitch, I would need some
> inputs as how this can be achieved in Freeswitch environment.
>
>
> 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  Mon Jan 16 19:21:38 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 16 Jan 2017 11:21:38 -0500
Subject: [Freeswitch-users] load multiple grammar
In-Reply-To: <819781D671CC1D44944772177554935003C4C2E3@ASH-EXCH-MB06.corp.yp.com>
References: <819781D671CC1D44944772177554935003C4C2E3@ASH-EXCH-MB06.corp.yp.com>
Message-ID: <A04AB6E3-3F66-459B-9878-8FD2A34E1920@jerris.com>


> On Jan 12, 2017, at 12:19 PM, Nagendra Babu <nb3868 at yp.com> wrote:
> 
> Hi,
>  
> I need help regarding loading and activating multiple grammars.
>  
> Per FS-2906 <https://freeswitch.org/jira/browse/FS-2906>, it is possible to load multiple grammars from FreeSwitch.
>  
> I have tried the same using following commands from my C# code. The FS version I have is 1.3.17.

That appears to be some sort of random dev version from 5+ years ago.  I?d suggest starting with looking at a a recent release and if you are still having issues work from there.

>  
> InboundSession.Execute("detect_speech", "unimrcp {start-recognize=false}yesnod yesnod");
> InboundSession.Execute("detect_speech", "unimrcp {start-recognize=true}yesno yesno");
>  
> I can see that there are two DEFINE-GRAMMAR requests sent to the LumenVox Speech Server. One with yesnod.gram and another with yesno.gram
> But there is only one session mentioned in the RECOGNIZE request where as I am expecting it to be having both the sessions described in the DEFINE-GRAMMAR request.
> Any idea?
>  
>  
>                 MRCP/2.0 226 RECOGNIZE 3
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Content-Type: text/uri-list
>                 Cancel-If-Queue: false
>                 Start-Input-Timers: false
>                 Speech-Complete-Timeout: 3000
>                 Content-Length: 13
>                 
>                 session:yesno
> >>>> There should be session:yesnod also.
>  
> I do not see a way to do this in detect_speech.
>  
> Any help would be great.
>  
>  
>  
> Here is the define grammar message too?
>  
>                 MRCP/2.0 318 DEFINE-GRAMMAR 1
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Content-Type: application/srgs
>                 Content-Id: yesnod
>                 Content-Length: 158
>                 
>                 #ABNF 1.0 UTF-8;
> language en-US;
> mode dtmf;
> tag-format <semantics/1.0>; 
>  
> root $yesorno;
>  
> $yes = 1 {out = "yes"};
> $no = 2 {out = "no"};
> $yesorno = $yes | $no;
>                 
> 01/12/2017 07:16:09.790,RECV,OnLoadGrammarRe,Received cseq grammar load (1) reply for CallIndex 51
> 01/12/2017 07:16:09.790,SEND,SendPacket     ,  
>                 
>                 MRCP/2.0 116 1 200 COMPLETE
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Completion-Cause: 000 success
>                 
>                 
> 01/12/2017 07:16:10.123,RECV,ProcessRcvdPkt ,Got MRCP Message -
>                 
>                 MRCP/2.0 408 DEFINE-GRAMMAR 2
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Content-Type: application/srgs
>                 Content-Id: yesno
>                 Content-Length: 249
>                 
>                 #ABNF 1.0 UTF-8;
>                 language en-US; //use the American English pronunciation dictionary.
>                 mode voice;  //the input for this grammar will be spoken words.
>                 tag-format <semantics/1.0>; 
>                 
>                 root $yesorno;
>                 
>                 $yes = yes;
>                 $no = no;
>                 $yesorno = $yes | $no;
>                 
> 01/12/2017 07:16:10.128,RECV,OnLoadGrammarRe,Received cseq grammar load (2) reply for CallIndex 51
> 01/12/2017 07:16:10.129,SEND,SendPacket     ,  
>                 
>                 MRCP/2.0 116 2 200 COMPLETE
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Completion-Cause: 000 success
>                 
>                 
> 01/12/2017 07:16:10.459,RECV,ProcessRcvdPkt ,Got MRCP Message -
>                 
>                 MRCP/2.0 226 RECOGNIZE 3
>                 Channel-Identifier: 115483E487508175CFDF at speechrecog
>                 Content-Type: text/uri-list
>                 Cancel-If-Queue: false
>                 Start-Input-Timers: false
>                 Speech-Complete-Timeout: 3000
>                 Content-Length: 13
>                 
>                 session:yesno
>  
>  
> 01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Speech-Complete-Timeout' with value '3000'
> 01/12/2017 07:16:10.460,DEBG,SetRecPrmFrmHdr,Setting parameter 'Start-Input-Timers' with value 'false'
> 01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting speech_complete_timeout to 3000 ms
> 01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting sensitivity level to 50
> 01/12/2017 07:16:10.460,DEBG,ConfPortForReco,Setting Speed vs Accuracy to 0.5
> 01/12/2017 07:16:10.461,SEND,SendPacket     ,  
>                 
>                 MRCP/2.0 87 3 200 IN-PROGRESS
>  
>  
>  
>  
> Thanks and regards,
> Nagendra
>  
> _________________________________________________________________________
> 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 mike at jerris.com  Mon Jan 16 19:29:14 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 16 Jan 2017 11:29:14 -0500
Subject: [Freeswitch-users] T.38 parameters causing call rejections
In-Reply-To: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
References: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
Message-ID: <D7080FB5-0E21-482F-97F9-237C8CD66860@jerris.com>

can you explain what you would like it to do?

> On Jan 15, 2017, at 10:43 PM, Tristan Dean <tristan at mocha.net.nz> wrote:
> 
> Hi All
>  
> I am having trouble on a new FreeSWITCH-based SBC we?ve built. We are going through some SBCs and upgrading them. In this case, we are upgrading an old SBC running: 1.4.15 64bit built from source.
>  
> The new version is running on Debian 8.3 (AMD64) Stable with the packaged version of FreeSWITCH:  1.6.14 -23-e460bf8 64bit
> We are tying this in with a hardware swap, so the 1.6.14-23 box is new.
>  
> We have a few instances where our upstream carrier presents us with a SDP which contains:
> m=image 56594 udptl t38
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
>  
> Historically (version 1.4.15) these seem to be ignored and FreeSWITCH passes the call out obeying the absolute_codec_string set during bridge. Transcoding is allowed and when negotiating a codec with the carrier, these parameters were ignored and G.711a was usually negotiated on both legs.
>  
> On the new platform, FreeSWITCH seems to be attempting to parse these and negotiate T.38. I am certain that it is something silly that I?m doing wrong, but can?t find it and would welcome others? thoughts. What have I missed? Has anyone seen this before? I have tried t38_passthru=true to no avail.
>  
> I can?t set proxy_media=true or bypass_media=true as we need FreeSWITCH to shield other devices from the SDP parameters above.
> Most of the configuration is stored in a database and returned to FreeSWITCH using mod_xml_curl. I have pasted some configuration excerpts, SIP messages and logs in: https://pastebin.freeswitch.org/view/de059ffc <https://pastebin.freeswitch.org/view/de059ffc>
>  
> Please let me know if there is any other information you need from me.
>  
> Many thanks in advance for your assistance.
>  
> Tristan

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From mike at jerris.com  Mon Jan 16 19:30:49 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 16 Jan 2017 11:30:49 -0500
Subject: [Freeswitch-users] Need starting point to Develop QSIG on
 Freeswitch
In-Reply-To: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
References: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
Message-ID: <865B0635-E63F-4F95-8360-EFF17E95DD3A@jerris.com>

in lib freetdm we have our own pri stack thats a good start for qsig.  Warning that code was never full production quality, it will need some love and attention, but it does have the big pieces you need in place and is fairly strait frorward code.  

> On Jan 16, 2017, at 12:17 AM, Naveen Khanna <naveen.khanna.bm at gmail.com> wrote:
> 
> I understand that QSIG is not supported on Freeswitch, I would need some inputs as how this can be achieved in Freeswitch environment.
> 
> 
> 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 olegstolyar at gmail.com  Mon Jan 16 22:15:32 2017
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 16 Jan 2017 11:15:32 -0800
Subject: [Freeswitch-users] Best way to fully open inbound calls
Message-ID: <CAC8JMOLH0YBqOND-jMrxnb3+0MGrhw=i=1rpYJ24vdkTF=n9kQ@mail.gmail.com>

Hi guys,

I need to allow all calls on a profile to go through regardless of where
they came from.  One way to do it is to create my own ACL list like this:

    <list name="open" default="allow">
      <node type="allow" cidr="192.168.42.42/32"/> //Note that it does not
work without at least one node in the list even if the list's default is
"allow"
    </list>

and then use that in my sip profile:
<param name="apply-inbound-acl" value="open"/>

This works but is there a more standard way to do it?
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From mike at jerris.com  Mon Jan 16 22:19:29 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 16 Jan 2017 14:19:29 -0500
Subject: [Freeswitch-users] Best way to fully open inbound calls
In-Reply-To: <CAC8JMOLH0YBqOND-jMrxnb3+0MGrhw=i=1rpYJ24vdkTF=n9kQ@mail.gmail.com>
References: <CAC8JMOLH0YBqOND-jMrxnb3+0MGrhw=i=1rpYJ24vdkTF=n9kQ@mail.gmail.com>
Message-ID: <34FE33E5-97A6-41F9-B1B4-890BD1D2E878@jerris.com>

Set a profile to not auth, like the external profile is by default.  Set the dial plan context in that profile appropriately to the destinations it should be able to reach.


> On Jan 16, 2017, at 2:15 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi guys,
> 
> I need to allow all calls on a profile to go through regardless of where they came from.  One way to do it is to create my own ACL list like this:
> 
>     <list name="open" default="allow">
>       <node type="allow" cidr="192.168.42.42/32 <http://192.168.42.42/32>"/> //Note that it does not work without at least one node in the list even if the list's default is "allow"
>     </list>
> 
> and then use that in my sip profile:
> <param name="apply-inbound-acl" value="open"/>
> 
> This works but is there a more standard way to do it?

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From findmeinwland at gmail.com  Mon Jan 16 22:32:30 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Tue, 17 Jan 2017 00:32:30 +0500
Subject: [Freeswitch-users] Best way to fully open inbound calls
In-Reply-To: <34FE33E5-97A6-41F9-B1B4-890BD1D2E878@jerris.com>
References: <CAC8JMOLH0YBqOND-jMrxnb3+0MGrhw=i=1rpYJ24vdkTF=n9kQ@mail.gmail.com>
	<34FE33E5-97A6-41F9-B1B4-890BD1D2E878@jerris.com>
Message-ID: <CAOLEQ7dCctziSEt5xZnmhUntZFd4DbWf4pzmic2UP9d-ROQU3Q@mail.gmail.com>

You can set this in your profile:

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

But it's too risky. Also you can use

<param name="auth-calls" value="false"/>

like M.Jerris suggested

2017-01-17 0:19 GMT+05:00 Michael Jerris <mike at jerris.com>:

> Set a profile to not auth, like the external profile is by default.  Set
> the dial plan context in that profile appropriately to the destinations it
> should be able to reach.
>
>
> On Jan 16, 2017, at 2:15 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> Hi guys,
>
> I need to allow all calls on a profile to go through regardless of where
> they came from.  One way to do it is to create my own ACL list like this:
>
>     <list name="open" default="allow">
>       <node type="allow" cidr="192.168.42.42/32"/> //Note that it does
> not work without at least one node in the list even if the list's default
> is "allow"
>     </list>
>
> and then use that in my sip profile:
> <param name="apply-inbound-acl" value="open"/>
>
> This works but is there a more standard way to do it?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?????
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From brian at freeswitch.org  Mon Jan 16 22:35:33 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Jan 2017 19:35:33 +0000
Subject: [Freeswitch-users] T.38 parameters causing call rejections
In-Reply-To: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
References: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
Message-ID: <CAEJMVkCSfG9ojTZVqAVfU=3HkD9Ukbw5UKnVy1P80W6scr9QDw@mail.gmail.com>

I suspect tiur analysis is wrong, Please fule a JIRA so I can fully vet
this issue

On Mon, Jan 16, 2017 at 10:18 AM Tristan Dean <tristan at mocha.net.nz> wrote:

> Hi All
>
>
>
> I am having trouble on a new FreeSWITCH-based SBC we?ve built. We are
> going through some SBCs and upgrading them. In this case, we are upgrading
> an old SBC running: 1.4.15 64bit built from source.
>
>
>
> The new version is running on Debian 8.3 (AMD64) Stable with the packaged
> version of FreeSWITCH:  1.6.14 -23-e460bf8 64bit
>
> We are tying this in with a hardware swap, so the 1.6.14-23 box is new.
>
>
>
> We have a few instances where our upstream carrier presents us with a SDP
> which contains:
>
> m=image 56594 udptl t38
>
> a=T38FaxUdpEC:t38UDPRedundancy
>
> a=T38FaxRateManagement:transferredTCF
>
>
>
> Historically (version 1.4.15) these seem to be ignored and FreeSWITCH
> passes the call out obeying the absolute_codec_string set during bridge.
> Transcoding is allowed and when negotiating a codec with the carrier, these
> parameters were ignored and G.711a was usually negotiated on both legs.
>
>
>
> On the new platform, FreeSWITCH seems to be attempting to parse these and
> negotiate T.38. I am certain that it is something silly that I?m doing
> wrong, but can?t find it and would welcome others? thoughts. What have I
> missed? Has anyone seen this before? I have tried t38_passthru=true to no
> avail.
>
>
>
> I can?t set proxy_media=true or bypass_media=true as we need FreeSWITCH to
> shield other devices from the SDP parameters above.
>
> Most of the configuration is stored in a database and returned to
> FreeSWITCH using mod_xml_curl. I have pasted some configuration excerpts,
> SIP messages and logs in: https://pastebin.freeswitch.org/view/de059ffc
>
>
>
> Please let me know if there is any other information you need from me.
>
>
>
> Many thanks in advance for your assistance.
>
>
>
> Tristan
>
>
>
>
>
>
> ------------------------------
>
>
>
>
>
>
>
>
>
>
>
>
>
> [image: AVG logo]
>
> <http://www.avg.com/internet-security>
>
>
>
>
>
>
> This email has been checked for viruses by AVG antivirus software.
>
>
> www.avg.com <http://www.avg.com/internet-security>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.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  Mon Jan 16 22:37:45 2017
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 16 Jan 2017 11:37:45 -0800
Subject: [Freeswitch-users] Best way to fully open inbound calls
In-Reply-To: <CAOLEQ7dCctziSEt5xZnmhUntZFd4DbWf4pzmic2UP9d-ROQU3Q@mail.gmail.com>
References: <CAC8JMOLH0YBqOND-jMrxnb3+0MGrhw=i=1rpYJ24vdkTF=n9kQ@mail.gmail.com>
	<34FE33E5-97A6-41F9-B1B4-890BD1D2E878@jerris.com>
	<CAOLEQ7dCctziSEt5xZnmhUntZFd4DbWf4pzmic2UP9d-ROQU3Q@mail.gmail.com>
Message-ID: <CAC8JMOLo0x3NLFQhUFyG+_cW3aH6AnK1X8zDFjmT6DqfK3CJhg@mail.gmail.com>

Perfect, auth-calls is what I've been looking for.  Thanks guys!

On Mon, Jan 16, 2017 at 11:32 AM, Artur Mega <findmeinwland at gmail.com>
wrote:

> You can set this in your profile:
>
> <param name="accept-blind-auth" value="true"/>
>
> But it's too risky. Also you can use
>
> <param name="auth-calls" value="false"/>
>
> like M.Jerris suggested
>
> 2017-01-17 0:19 GMT+05:00 Michael Jerris <mike at jerris.com>:
>
>> Set a profile to not auth, like the external profile is by default.  Set
>> the dial plan context in that profile appropriately to the destinations it
>> should be able to reach.
>>
>>
>> On Jan 16, 2017, at 2:15 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>> Hi guys,
>>
>> I need to allow all calls on a profile to go through regardless of where
>> they came from.  One way to do it is to create my own ACL list like this:
>>
>>     <list name="open" default="allow">
>>       <node type="allow" cidr="192.168.42.42/32"/> //Note that it does
>> not work without at least one node in the list even if the list's default
>> is "allow"
>>     </list>
>>
>> and then use that in my sip profile:
>> <param name="apply-inbound-acl" value="open"/>
>>
>> This works but is there a more standard way to do it?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?????
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mirkobrankovic at gmail.com  Tue Jan 17 11:19:33 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Tue, 17 Jan 2017 09:19:33 +0100
Subject: [Freeswitch-users] T.38 parameters causing call rejections
In-Reply-To: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
References: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
Message-ID: <CAND18T2xRaaJ6m-ad9+kfgj0sMx_ZU7Z-PT4sT-so2y0T8-wgQ@mail.gmail.com>

Take a look at https://wiki.freeswitch.org/wiki/Mod_spandsp#t38_gateway

Probably you need to disable the t38 on one of the sides.

On Mon, Jan 16, 2017 at 4:43 AM, Tristan Dean <tristan at mocha.net.nz> wrote:

> Hi All
>
>
>
> I am having trouble on a new FreeSWITCH-based SBC we?ve built. We are
> going through some SBCs and upgrading them. In this case, we are upgrading
> an old SBC running: 1.4.15 64bit built from source.
>
>
>
> The new version is running on Debian 8.3 (AMD64) Stable with the packaged
> version of FreeSWITCH:  1.6.14 -23-e460bf8 64bit
>
> We are tying this in with a hardware swap, so the 1.6.14-23 box is new.
>
>
>
> We have a few instances where our upstream carrier presents us with a SDP
> which contains:
>
> m=image 56594 udptl t38
>
> a=T38FaxUdpEC:t38UDPRedundancy
>
> a=T38FaxRateManagement:transferredTCF
>
>
>
> Historically (version 1.4.15) these seem to be ignored and FreeSWITCH
> passes the call out obeying the absolute_codec_string set during bridge.
> Transcoding is allowed and when negotiating a codec with the carrier, these
> parameters were ignored and G.711a was usually negotiated on both legs.
>
>
>
> On the new platform, FreeSWITCH seems to be attempting to parse these and
> negotiate T.38. I am certain that it is something silly that I?m doing
> wrong, but can?t find it and would welcome others? thoughts. What have I
> missed? Has anyone seen this before? I have tried t38_passthru=true to no
> avail.
>
>
>
> I can?t set proxy_media=true or bypass_media=true as we need FreeSWITCH to
> shield other devices from the SDP parameters above.
>
> Most of the configuration is stored in a database and returned to
> FreeSWITCH using mod_xml_curl. I have pasted some configuration excerpts,
> SIP messages and logs in: https://pastebin.freeswitch.org/view/de059ffc
>
>
>
> Please let me know if there is any other information you need from me.
>
>
>
> Many thanks in advance for your assistance.
>
>
>
> Tristan
>
>
> ------------------------------
> [image: AVG logo] <http://www.avg.com/internet-security>
>
> This email has been checked for viruses by AVG antivirus software.
> www.avg.com <http://www.avg.com/internet-security>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Mirko
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From naveen.khanna.bm at gmail.com  Tue Jan 17 14:51:37 2017
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 17 Jan 2017 17:21:37 +0530
Subject: [Freeswitch-users] Need starting point to Develop QSIG on
	Freeswitch
In-Reply-To: <865B0635-E63F-4F95-8360-EFF17E95DD3A@jerris.com>
References: <CAE8fD+0RCDjTSrQaZJZR6j7Pxt3fReKXH0j5wiKy3LQxcLYsAg@mail.gmail.com>
	<865B0635-E63F-4F95-8360-EFF17E95DD3A@jerris.com>
Message-ID: <CAE8fD+2gHJ02ta4YW31+7_yKE+Ds-wfOV-qF7Cqg7XRA4BznhA@mail.gmail.com>

Thanks for the inputs.

Will start working on it.

Regards,

Naveen Khanna

On 16 January 2017 at 22:00, Michael Jerris <mike at jerris.com> wrote:

> in lib freetdm we have our own pri stack thats a good start for qsig.
> Warning that code was never full production quality, it will need some love
> and attention, but it does have the big pieces you need in place and is
> fairly strait frorward code.
>
> > On Jan 16, 2017, at 12:17 AM, Naveen Khanna <naveen.khanna.bm at gmail.com>
> wrote:
> >
> > I understand that QSIG is not supported on Freeswitch, I would need some
> inputs as how this can be achieved in Freeswitch environment.
> >
> >
> > 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Jan 17 16:10:00 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 17 Jan 2017 13:10:00 +0000
Subject: [Freeswitch-users] shared memory pools
Message-ID: <CAHtxdDd9nUnCmVhwRk+nS5eq-WBWZRpuu10p=5_K2JYdNZeC8w@mail.gmail.com>

This questions for FS core developers.
I found that FreeSwitch apr libs have apr_pool_mutex_set that not exists in
APR master.
I found commit
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/ffb1cb4f690473af696fe1fe3879790fe49acecb>
where this function is introduced.
Purpose of this commit is resolve concurrency access from different threads
to same memory pool.

I asked
<http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCAHtxdDfDwmoACnYn04FTWWgtpK8giJjsgSgg5VvuSN%2Bph4G%3DBw%40mail.gmail.com%3E>
APR community why not used mutex and got responce from William A Rowe
<http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCACsi2539CFJusRo0wmReEEdrdG5ppJsJb-C9aD-g7K6tFos%2BOA%40mail.gmail.com%3E>

My questions
Does core team developers planning rewrite memory pool operation to remove
shared access to same pool from different threads? Create own pool for each
thread.

Sergey
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From krice at freeswitch.org  Tue Jan 17 18:44:11 2017
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 17 Jan 2017 09:44:11 -0600
Subject: [Freeswitch-users] How are you using FreeSWITCH?
Message-ID: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>

Hey Guys,

 

A lot of you are using FreeSWITCH for some fairly complex things. However
that is not always clear to people discovering FreeSWITCH for the first
time.

 

I would like to build a list of examples. If you don't mind this info being
compiled on Confluence.

Please  Reply with who, what, and where so we can document it and show off
some of the really cool FreeSWITCH based or FreeSWITCH using deployments out
there.

 

Thanks

Ken

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From brian at freeswitch.org  Tue Jan 17 18:45:56 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 17 Jan 2017 15:45:56 +0000
Subject: [Freeswitch-users] T.38 parameters causing call rejections
In-Reply-To: <CAND18T2xRaaJ6m-ad9+kfgj0sMx_ZU7Z-PT4sT-so2y0T8-wgQ@mail.gmail.com>
References: <000001d26faa$b547a2e0$1fd6e8a0$@mocha.net.nz>
	<CAND18T2xRaaJ6m-ad9+kfgj0sMx_ZU7Z-PT4sT-so2y0T8-wgQ@mail.gmail.com>
Message-ID: <CAEJMVkDYzSY3BAZAcVRGO45gokr_2yK3eKMKpeOzy0ULqq7w4g@mail.gmail.com>

See test cases I've done in the testing config repo,  Those docs on the
wiki are  ot exactly correct on gis topic!

On Tue, Jan 17, 2017 at 3:21 AM Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> Take a look at https://wiki.freeswitch.org/wiki/Mod_spandsp#t38_gateway
>
> Probably you need to disable the t38 on one of the sides.
>
> On Mon, Jan 16, 2017 at 4:43 AM, Tristan Dean <tristan at mocha.net.nz>
> wrote:
>
> Hi All
>
>
>
> I am having trouble on a new FreeSWITCH-based SBC we?ve built. We are
> going through some SBCs and upgrading them. In this case, we are upgrading
> an old SBC running: 1.4.15 64bit built from source.
>
>
>
> The new version is running on Debian 8.3 (AMD64) Stable with the packaged
> version of FreeSWITCH:  1.6.14 -23-e460bf8 64bit
>
> We are tying this in with a hardware swap, so the 1.6.14-23 box is new.
>
>
>
> We have a few instances where our upstream carrier presents us with a SDP
> which contains:
>
> m=image 56594 udptl t38
>
> a=T38FaxUdpEC:t38UDPRedundancy
>
> a=T38FaxRateManagement:transferredTCF
>
>
>
> Historically (version 1.4.15) these seem to be ignored and FreeSWITCH
> passes the call out obeying the absolute_codec_string set during bridge.
> Transcoding is allowed and when negotiating a codec with the carrier, these
> parameters were ignored and G.711a was usually negotiated on both legs.
>
>
>
> On the new platform, FreeSWITCH seems to be attempting to parse these and
> negotiate T.38. I am certain that it is something silly that I?m doing
> wrong, but can?t find it and would welcome others? thoughts. What have I
> missed? Has anyone seen this before? I have tried t38_passthru=true to no
> avail.
>
>
>
> I can?t set proxy_media=true or bypass_media=true as we need FreeSWITCH to
> shield other devices from the SDP parameters above.
>
> Most of the configuration is stored in a database and returned to
> FreeSWITCH using mod_xml_curl. I have pasted some configuration excerpts,
> SIP messages and logs in: https://pastebin.freeswitch.org/view/de059ffc
>
>
>
> Please let me know if there is any other information you need from me.
>
>
>
> Many thanks in advance for your assistance.
>
>
>
> Tristan
>
>
>
>
>
>
> ------------------------------
>
>
>
>
>
>
>
>
>
>
>
>
>
> [image: AVG logo]
>
> <http://www.avg.com/internet-security>
>
>
>
>
>
>
> This email has been checked for viruses by AVG antivirus software.
>
>
> www.avg.com <http://www.avg.com/internet-security>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
>
>
> Professional FreeSWITCH Consulting Services:
>
>
> consulting at freeswitch.org
>
>
> http://www.freeswitchsolutions.com
>
>
>
>
>
> Official FreeSWITCH Sites
>
>
> http://www.freeswitch.org
>
>
> http://confluence.freeswitch.org
>
>
> http://www.cluecon.com
>
>
>
>
>
> FreeSWITCH-users mailing list
>
>
> FreeSWITCH-users at 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,
> Mirko
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.freeswitch.org
>
> http://www.cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
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From nneul at mst.edu  Tue Jan 17 19:00:25 2017
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 17 Jan 2017 10:00:25 -0600
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <2563589e-3cf2-053c-a656-aed4f986bb07@mst.edu>

Phone system for a University Campus in Rolla, Missouri that replaced an old Cisco Call Manager deployment. Using 
approximately 1400 Cisco Skinny phones and a mix of 500 Yealink, Polycom, Grandstream and Linksys devices. Custom 
homegrown UI for configuring and managing the environment. Running in a dual node setup with failover using keepalived 
on freeswitch and a backend three-node Percona XTraDB MySQL Cluster.

-- Nathan

On 01/17/2017 09:44 AM, Ken Rice wrote:
> Hey Guys,
>
>
>
> A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering
> FreeSWITCH for the first time.
>
>
>
> I would like to build a list of examples. If you don?t mind this info being compiled on Confluence.
>
> Please  Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or
> FreeSWITCH using deployments out there.
>
>
>
> Thanks
>
> Ken
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike at jerris.com  Tue Jan 17 19:12:48 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 17 Jan 2017 11:12:48 -0500
Subject: [Freeswitch-users] shared memory pools
In-Reply-To: <CAHtxdDd9nUnCmVhwRk+nS5eq-WBWZRpuu10p=5_K2JYdNZeC8w@mail.gmail.com>
References: <CAHtxdDd9nUnCmVhwRk+nS5eq-WBWZRpuu10p=5_K2JYdNZeC8w@mail.gmail.com>
Message-ID: <535A8BC8-9274-44CE-9057-048FC94691F4@jerris.com>

We have no plans to make major changes in this space.

> On Jan 17, 2017, at 8:10 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> This questions for FS core developers.
> I found that FreeSwitch apr libs have apr_pool_mutex_set that not exists in APR master.
> I found commit <https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/ffb1cb4f690473af696fe1fe3879790fe49acecb> where this function is introduced.
> Purpose of this commit is resolve concurrency access from different threads to same memory pool.
> 
> I asked <http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCAHtxdDfDwmoACnYn04FTWWgtpK8giJjsgSgg5VvuSN%2Bph4G%3DBw%40mail.gmail.com%3E> APR community why not used mutex and got responce from William A Rowe <http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCACsi2539CFJusRo0wmReEEdrdG5ppJsJb-C9aD-g7K6tFos%2BOA%40mail.gmail.com%3E>
> 
> My questions
> Does core team developers planning rewrite memory pool operation to remove shared access to same pool from different threads? Create own pool for each thread.
> 
> Sergey
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From manpower13.cse at gmail.com  Tue Jan 17 19:24:13 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 17 Jan 2017 21:54:13 +0530
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <2563589e-3cf2-053c-a656-aed4f986bb07@mst.edu>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<2563589e-3cf2-053c-a656-aed4f986bb07@mst.edu>
Message-ID: <CAMRuXGi4mZBr0v6qbwMCtV+Sh5LFvDh-NRM=7tqRpdaxfW6Lpw@mail.gmail.com>

Hi,

     We are startup from india TeleCMI Technologies (https://telecmi.com),We
are provide communication from every platform Like browser(WebRTC),Mobile
APP(SIP and WebRTC) and PSTN,Without freeswitch it not possible Thanks
Freeswitch we are all youngster we did this from village because of
opensource and support from freeswitch Thanks once again .

On Tue, Jan 17, 2017 at 9:30 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> Phone system for a University Campus in Rolla, Missouri that replaced an
> old Cisco Call Manager deployment. Using
> approximately 1400 Cisco Skinny phones and a mix of 500 Yealink, Polycom,
> Grandstream and Linksys devices. Custom
> homegrown UI for configuring and managing the environment. Running in a
> dual node setup with failover using keepalived
> on freeswitch and a backend three-node Percona XTraDB MySQL Cluster.
>
> -- Nathan
>
> On 01/17/2017 09:44 AM, Ken Rice wrote:
> > Hey Guys,
> >
> >
> >
> > A lot of you are using FreeSWITCH for some fairly complex things.
> However that is not always clear to people discovering
> > FreeSWITCH for the first time.
> >
> >
> >
> > I would like to build a list of examples. If you don?t mind this info
> being compiled on Confluence.
> >
> > Please  Reply with who, what, and where so we can document it and show
> off some of the really cool FreeSWITCH based or
> > FreeSWITCH using deployments out there.
> >
> >
> >
> > Thanks
> >
> > Ken
> >
> >
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
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From flokrrr at gmail.com  Tue Jan 17 19:29:38 2017
From: flokrrr at gmail.com (Florent Krieg)
Date: Tue, 17 Jan 2017 17:29:38 +0100
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <2563589e-3cf2-053c-a656-aed4f986bb07@mst.edu>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<2563589e-3cf2-053c-a656-aed4f986bb07@mst.edu>
Message-ID: <CAABKsuR8DArLbhyt9qiTPazCTKmZ0tLS_sDCPq1cQxud1ybjNw@mail.gmail.com>

A few setups here (Sewan Group, French carrier, France):
- Freeswitch for SIP trunking platforms (1500+ trunk accounts provisioned
globally, 500 to 1000 simultaneous calls at peak per cluster, 3 clusters
and another new one soon-ish).
A cluster is an active/backup pair of physical servers where Freeswitch
acts mainly as a call processing and call routing SIP server (doing
billing, lcr, sip interworking plus some fancy tweaks for our customers).
HA is handled by keepalived and an active-active Mariadb setup.
- Freeswitch as an incoming fax2mail platform. 400k+ sessions handled over
the last 3 months (it's a recent migration from callweaver).
- Freeswitch as a multi-purposes platform (mainly handling outgoing number
portability, pre-announced calls used for paid services for instance - so
billing here is important - and emergency numbers lookup and translation).
In this setup, Kamailio is used to load-balance calls to Freeswitch hosts.
- Freeswitch used as a callcenter platform using a pool of servers that can
be automatically added/removed to handle scalability. 250+ callgroups
provisioned at the moment on a few virtualized hosts.
Dispatching and provisioning is handled by an in-house developed engine
(Consul + Python processes + Mariadb).

Florent

2017-01-17 17:00 GMT+01:00 Nathan Neulinger <nneul at mst.edu>:

> Phone system for a University Campus in Rolla, Missouri that replaced an
> old Cisco Call Manager deployment. Using
> approximately 1400 Cisco Skinny phones and a mix of 500 Yealink, Polycom,
> Grandstream and Linksys devices. Custom
> homegrown UI for configuring and managing the environment. Running in a
> dual node setup with failover using keepalived
> on freeswitch and a backend three-node Percona XTraDB MySQL Cluster.
>
> -- Nathan
>
> On 01/17/2017 09:44 AM, Ken Rice wrote:
> > Hey Guys,
> >
> >
> >
> > A lot of you are using FreeSWITCH for some fairly complex things.
> However that is not always clear to people discovering
> > FreeSWITCH for the first time.
> >
> >
> >
> > I would like to build a list of examples. If you don?t mind this info
> being compiled on Confluence.
> >
> > Please  Reply with who, what, and where so we can document it and show
> off some of the really cool FreeSWITCH based or
> > FreeSWITCH using deployments out there.
> >
> >
> >
> > Thanks
> >
> > Ken
> >
> >
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
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From sms at icefire.qza.net.au  Tue Jan 17 19:39:50 2017
From: sms at icefire.qza.net.au (Francis)
Date: Wed, 18 Jan 2017 02:39:50 +1000
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <bf62e082-4f5c-181f-7e06-22e584361429@icefire.qza.net.au>

Remote location standalone server, two separate sip provider accounts 
over 4g, one GSM gateway, basic call groups with a mix of hard phone, 
DECT and soft clients. Voicemail to email, roaming user access over 4g / 
IPSEC VPN via cloud VPS.

Francis


On 18/01/2017 1:44 AM, Ken Rice wrote:
>
> Hey Guys,
>
> A lot of you are using FreeSWITCH for some fairly complex things. 
> However that is not always clear to people discovering FreeSWITCH for 
> the first time.
>
> I would like to build a list of examples. If you don?t mind this info 
> being compiled on Confluence.
>
> Please  Reply with who, what, and where so we can document it and show 
> off some of the really cool FreeSWITCH based or FreeSWITCH using 
> deployments out there.
>
> Thanks
>
> Ken
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Jan 17 19:44:45 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 17 Jan 2017 10:44:45 -0600
Subject: [Freeswitch-users] shared memory pools
In-Reply-To: <535A8BC8-9274-44CE-9057-048FC94691F4@jerris.com>
References: <CAHtxdDd9nUnCmVhwRk+nS5eq-WBWZRpuu10p=5_K2JYdNZeC8w@mail.gmail.com>
	<535A8BC8-9274-44CE-9057-048FC94691F4@jerris.com>
Message-ID: <CAKbxfG9EMq7BkLjbkdsVAxD6paXR3fLHe6CmyKathuOy2nTSxA@mail.gmail.com>

We still try to avoid using pools from different threads.  When its done
its only done on a second hand context, eg.. uuid_setvar  Since Sessions
have their own pool, and uuid_setvar is used from another thread, the only
solution is to use mutexes.  That is why the patch we added is designed to
activate mutex mode only when needed which we do need in FreeSWITCH.  You
could go on to ask him in your thread if you wish why they have not
commented on our patch that has been submitted for several years now to get
the few patches we have into main APR so we don't have to use our own.  We
have tried to explain it to them a few times and usually the answer is:
meh!  and its dismissed.

I would argue that FreeSWITCH gets its fair share of heavy use and has a
reputation for scalability.  This is all done with mutexed pools so its
clearly not the end of the world to do it.

The final result is we are forced to use our own copy to make sure we get
the functionality we need.




On Tue, Jan 17, 2017 at 10:12 AM, Michael Jerris <mike at jerris.com> wrote:

> We have no plans to make major changes in this space.
>
> On Jan 17, 2017, at 8:10 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> This questions for FS core developers.
> I found that FreeSwitch apr libs have apr_pool_mutex_set that not exists
> in APR master.
> I found commit
> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/ffb1cb4f690473af696fe1fe3879790fe49acecb>
> where this function is introduced.
> Purpose of this commit is resolve concurrency access from different
> threads to same memory pool.
>
> I asked
> <http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCAHtxdDfDwmoACnYn04FTWWgtpK8giJjsgSgg5VvuSN%2Bph4G%3DBw%40mail.gmail.com%3E>
> APR community why not used mutex and got responce from William A Rowe
> <http://mail-archives.apache.org/mod_mbox/apr-dev/201701.mbox/%3CCACsi2539CFJusRo0wmReEEdrdG5ppJsJb-C9aD-g7K6tFos%2BOA%40mail.gmail.com%3E>
>
> My questions
> Does core team developers planning rewrite memory pool operation to remove
> shared access to same pool from different threads? Create own pool for each
> thread.
>
> Sergey
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From shaun.stokes at itec-support.co.uk  Tue Jan 17 19:46:39 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 17 Jan 2017 16:46:39 +0000
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8629459@mbx-01.sysconfig.co.uk>

ITEC Support, United Kingdom

Hosted VoIP and UC, FreeSWITCH features:

*         SIP extensions with full support for multiple registrations per extensions

*         Integration with Skype for Business with users as extensions

*         Limit number of concurrent calls per extension

*         Extension call control (call forwarding, do not disturb etc)

*         Inbound number blocking

*         Outbound voice policies (i.e. call barring for international, mobile etc)

*         IM

*         Presence

*         Call pickup groups

*         Voicemail

*         Call Centres

*         Ring Groups

*         Time of day routing

*         IVRs

*         Announcements

*         Virtual extension call forwarding

*         Fax to email

*         Email to Fax

Presence works on physical phones such as Polycom and Cisco SPA when ringing or on the phone.

Our preferred softphone client is Jitsi which supports IM and presence but when changing your status on Jitsi this will only be visible from another Jitsi client.

We have a resilient solution, each pool has a hot standby server in another data centre in case the primary fails.

Looking forward we would like to offer federation with Skype for Business and other platforms via XMPP, these are some of the options we're looking at to integrate with FreeSWITCH:

*         Repro\reSIProcate

*         Kamailio

*         Openfire



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: 17 January 2017 15:44
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] How are you using FreeSWITCH?

Hey Guys,

A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering FreeSWITCH for the first time.

I would like to build a list of examples. If you don't mind this info being compiled on Confluence.
Please  Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or FreeSWITCH using deployments out there.

Thanks
Ken

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

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______________________________________________________________________
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From areski at gmail.com  Tue Jan 17 19:52:01 2017
From: areski at gmail.com (Areski)
Date: Tue, 17 Jan 2017 17:52:01 +0100
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <CAMf7LBgKRP_Un6qjWQCW_xq24vmMYXMNgKrnxPV2HEP=zGw5rg@mail.gmail.com>

Hi,


Who?

Areski Belaid & Team

What?

We build Voice Broadcasting and Phone Survey solution using FreeSWITCH,
Python & Lua.

We provide solution for telemarketing, Political Campaign and for Mass
emergency broadcasting

Where?

Can be reach through our website https://www.newfies-dialer.org/



On Tue, Jan 17, 2017 at 4:44 PM, Ken Rice <krice at freeswitch.org> wrote:

> Hey Guys,
>
>
>
> A lot of you are using FreeSWITCH for some fairly complex things. However
> that is not always clear to people discovering FreeSWITCH for the first
> time.
>
>
>
> I would like to build a list of examples. If you don?t mind this info
> being compiled on Confluence.
>
> Please  Reply with who, what, and where so we can document it and show off
> some of the really cool FreeSWITCH based or FreeSWITCH using deployments
> out there.
>
>
>
> Thanks
>
> Ken
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kind regards,
/Areski

----
Arezqui Belaid
Founder at Star2Billing (www.star2billing.com)
& Newfies-Dialer (www.newfies-dialer.org)

Tel: +34650784355
Twitter: http://twitter.com/areskib
LinkedIn: http://www.linkedin.com/in/areski
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From kathleen at freeswitch.org  Tue Jan 17 22:05:26 2017
From: kathleen at freeswitch.org (Kathleen King)
Date: Tue, 17 Jan 2017 11:05:26 -0800
Subject: [Freeswitch-users] Annual FreeSWITCH Summit
Message-ID: <CAOStUPb9_UmC-M_xLE5Q5tyMdeMcNezuPjkMt8K4xn9Dvi2Hjg@mail.gmail.com>

Hello FreeSWITCHers,

As some of you may know the FreeSWITCH team gets together every year and
the annual FreeSWITCH Summit is fast approaching. This gathering of the
minds is where the FreeSWITCH developers hash out some of the big decisions
that go into making FreeSWITCH what it is today. In less than two weeks,
the FreeSWITCH team will descend onto a city in Wisconsin and spend a full
week tirelessly living and breathing FreeSWITCH. From dawn, well maybe
mid-morning, until dusk, the FreeSWITCH developers will work on some of the
hardest and most complicated issues facing FreeSWITCH.

We are asking for your help. In order to keep the team bright-eyed and
bushy-tailed we must keep them fed. This is costly for the growing team.
Show your appreciation donating so they can keep doing what they do
best. Support
the people that brought you the latest 1.6.14 release in all its glory. Any
and all donations are welcome.

<goog_1384205596>
https://www.gofundme.com/freeswitch-developer-meeting-2017


Kathleen King
FreeSWITCH Public Relations
Office: +1-213-286-0400
Mobile: +1-703-859-3757
http://freeswitch.org
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From jprangi at didforsale.com  Tue Jan 17 23:55:34 2017
From: jprangi at didforsale.com (Jai Rangi)
Date: Tue, 17 Jan 2017 12:55:34 -0800
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <CAPfjz3jTGuFp3Lx4rAwCksm4Bu1yE5KhySbJQ9hbs94Y0r3MEQ@mail.gmail.com>

We are using freeswitch for 2 different products:

www.cebodtelecom.com
(Hosted phone service) Multi-tenant phone system, hooked with mysql
cluster, and shared storage. Most complex features are manage with LUA
scripts. Phone service offers almost everything possible phone system
features.  (IVR, Groups, Conferencing, Call Center, Call Queue, Timebased
routing, Failover Routing, fax to email as well as fax to fax machine)).
Customers are using Yealink, GS, Polycom, Cisco and different softphones.


www.didforsale.com
(Sip Trunking service provider) Most of our customer have their own PBX and
we dont stay in media. We use freeswitch for customers where we have to
stay in media and customers who are behind NAT, wants to use softphones and
call forwarding etc.
One large project we have recently implemented is to offer a platform
similar to twilio and plivo. Where customers can build voice applications
on their web servers and use our platform to complete the calls.
https://www.didforsale.com/developer  <https://www.didforsale.com/developer>
Technology used, (Mysql Cluster, Shared Storage, LUA, Curl)

Indeed freeeswitch is great platform and possibilities are endless. Online
community great. Freeswitch Books (Core as well as cookbook) are the best
tools to learn freeswitch.

Hope this will be useful, I will be happy to provide some Snapshot if that
adds any value to new users.

Best,

*Jai Rangi*
Cebod Technologies LLC dba DIDforSale/Cebod Telecom
O 949-471-0102 | C 949-419-7634 <1-949-419-7634> | F 949-269-0449 /
949-232-1410 | jprangi at didforsale.com  www.cebod.com |  www.didforsale.com
|3200 Bristol St Suite 615, Costa Mesa, CA 92626  |






On Tue, Jan 17, 2017 at 7:44 AM, Ken Rice <krice at freeswitch.org> wrote:

> Hey Guys,
>
>
>
> A lot of you are using FreeSWITCH for some fairly complex things. However
> that is not always clear to people discovering FreeSWITCH for the first
> time.
>
>
>
> I would like to build a list of examples. If you don?t mind this info
> being compiled on Confluence.
>
> Please  Reply with who, what, and where so we can document it and show off
> some of the really cool FreeSWITCH based or FreeSWITCH using deployments
> out there.
>
>
>
> Thanks
>
> Ken
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From schoch+freeswitch.org at xwin32.com  Wed Jan 18 03:20:05 2017
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 17 Jan 2017 16:20:05 -0800
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <CAOkuKdf-SE4GXK9dPCD8AiOFo9xgOWv6dJkjzHv+48_AkOhbgQ@mail.gmail.com>

I'm using FreeSWITCH as a PBX for a small (under 50) company. It's running
on a Debian VM hosted on XenServer. Our provider is flowroute.com and we
use Comcast Business as our ISP.

One unusual thing I'm doing is I have an old Panasonic answering machine (
https://www.amazon.com/Panasonic-KXTC1501-Cordless-Digital-Answering/dp/B00004TZDL/ref=cm_cr_arp_d_product_top?ie=UTF8)
at
home (on a POTS line) that has the ability to dial a pager when a message
is received. (Remember pagers?) I set up FreeSWITCH to act as a pager, so
that the answering machine dials it via an extension number to our main
company number. It then runs a Lua script that takes the pager data, does a
CID lookup, and sends an email through vtext.com to my smartphone.  The
result is that when someone calls my home POTS line, I get a message that
tells me who called and if they left a message or not.

I also have a FreeSWITCH script that sends a SMS message (through
flowroute.com) when a voicemail message is received.

-- 
Steve

On Tue, Jan 17, 2017 at 7:44 AM, Ken Rice <krice at freeswitch.org> wrote:

> Hey Guys,
>
>
>
> A lot of you are using FreeSWITCH for some fairly complex things. However
> that is not always clear to people discovering FreeSWITCH for the first
> time.
>
>
>
> I would like to build a list of examples. If you don?t mind this info
> being compiled on Confluence.
>
> Please  Reply with who, what, and where so we can document it and show off
> some of the really cool FreeSWITCH based or FreeSWITCH using deployments
> out there.
>
>
>
> Thanks
>
> Ken
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gascagonzalo at gmail.com  Wed Jan 18 03:23:04 2017
From: gascagonzalo at gmail.com (Gonzalo Gasca Meza)
Date: Tue, 17 Jan 2017 16:23:04 -0800
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <CAPfjz3jTGuFp3Lx4rAwCksm4Bu1yE5KhySbJQ9hbs94Y0r3MEQ@mail.gmail.com>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<CAPfjz3jTGuFp3Lx4rAwCksm4Bu1yE5KhySbJQ9hbs94Y0r3MEQ@mail.gmail.com>
Message-ID: <CAL+DTgKmzBm_qVDPaGKjuKj6=oCwudo6fAUPBO6Yw_5GAQ4OBQ@mail.gmail.com>

*Who*
At Parzee <http://www.parzee.com> are using Freeswitch for 2 different
projects:

*What*
*Products:*
*Attorney:* Provides Visual Voicemail for Cisco Spark clients
<https://www.ciscospark.com/pricing/free-trial-DG.html>. Use Freeswitch
VoiceMail API to monitor user's Inbox and Speech to Text third party APIs
to create voicemail transcriptions.
*Telephonist:* Advanced call dialer to detect fixed number status. Use
Freeswitch ESL, Recording and *mod_spandsp
<https://wiki.freeswitch.org/wiki/Mod_spandsp#spandsp_start_fax_detect>.
*Production
~10 CPS scalable to 50 CPS. Stack: Python, AWS, PostgreSQL.

*Where*
Different customers across Canada, Brazil, Mexico, and US.

We can provide diagrams and performance statistics.

Gonzalo and Parzee team info at parzee.com


?


On Tue, Jan 17, 2017 at 12:55 PM, Jai Rangi <jprangi at didforsale.com> wrote:

> We are using freeswitch for 2 different products:
>
> www.cebodtelecom.com
> (Hosted phone service) Multi-tenant phone system, hooked with mysql
> cluster, and shared storage. Most complex features are manage with LUA
> scripts. Phone service offers almost everything possible phone system
> features.  (IVR, Groups, Conferencing, Call Center, Call Queue, Timebased
> routing, Failover Routing, fax to email as well as fax to fax machine)).
> Customers are using Yealink, GS, Polycom, Cisco and different softphones.
>
>
> www.didforsale.com
> (Sip Trunking service provider) Most of our customer have their own PBX
> and we dont stay in media. We use freeswitch for customers where we have to
> stay in media and customers who are behind NAT, wants to use softphones and
> call forwarding etc.
> One large project we have recently implemented is to offer a platform
> similar to twilio and plivo. Where customers can build voice applications
> on their web servers and use our platform to complete the calls.
> https://www.didforsale.com/developer
> <https://www.didforsale.com/developer>
> Technology used, (Mysql Cluster, Shared Storage, LUA, Curl)
>
> Indeed freeeswitch is great platform and possibilities are endless. Online
> community great. Freeswitch Books (Core as well as cookbook) are the best
> tools to learn freeswitch.
>
> Hope this will be useful, I will be happy to provide some Snapshot if that
> adds any value to new users.
>
> Best,
>
> *Jai Rangi*
> Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> O 949-471-0102 | C 949-419-7634 <1-949-419-7634> | F 949-269-0449 /
> 949-232-1410 | jprangi at didforsale.com  www.cebod.com |  www.didforsale.com
> |3200 Bristol St Suite 615, Costa Mesa, CA 92626  |
>
>
>
>
>
>
> On Tue, Jan 17, 2017 at 7:44 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>> Hey Guys,
>>
>>
>>
>> A lot of you are using FreeSWITCH for some fairly complex things. However
>> that is not always clear to people discovering FreeSWITCH for the first
>> time.
>>
>>
>>
>> I would like to build a list of examples. If you don?t mind this info
>> being compiled on Confluence.
>>
>> Please  Reply with who, what, and where so we can document it and show
>> off some of the really cool FreeSWITCH based or FreeSWITCH using
>> deployments out there.
>>
>>
>>
>> Thanks
>>
>> Ken
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manpower13.cse at gmail.com  Wed Jan 18 11:20:32 2017
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Wed, 18 Jan 2017 13:50:32 +0530
Subject: [Freeswitch-users] Nibblebill IVR not hangup
Message-ID: <CAMRuXGirFbeSdRcuMibgqKqO9dXt-e674t_5UN5V-yQT22+89Q@mail.gmail.com>

Hi,

    I have -2 in my account,but still i can hear ivr,Means call will never
end even i have amount in minus

  Bellow is my dial plan,

  <extension name="tollfree800">
    <condition field="destination_number" expression="^1{0,1}(800\d{7})$">
        <action application="set" data="nibble_rate=1.0"/>   <action
application="set" data="nibble_account=2"/>
   <action application="answer"/>

        <action application="lua" data="/opt/ivr.lua"/>
    </condition>
</extension>
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From italo at freeswitch.org  Wed Jan 18 12:57:55 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 18 Jan 2017 09:57:55 +0000
Subject: [Freeswitch-users] Annual FreeSWITCH Summit
In-Reply-To: <CAOStUPb9_UmC-M_xLE5Q5tyMdeMcNezuPjkMt8K4xn9Dvi2Hjg@mail.gmail.com>
References: <CAOStUPb9_UmC-M_xLE5Q5tyMdeMcNezuPjkMt8K4xn9Dvi2Hjg@mail.gmail.com>
Message-ID: <CABMX_w90qtQA9-Zy8O-2EaOBtEkCUu23=YhYj=cnmoOrLmx7ag@mail.gmail.com>

Hey guys excellent opportunity to show your support and gratitude to
FreeSWITCH team!

Go go! :-)
Em ter, 17 de jan de 2017 ?s 16:17, Kathleen King <kathleen at freeswitch.org>
escreveu:

> Hello FreeSWITCHers,
>
> As some of you may know the FreeSWITCH team gets together every year and
> the annual FreeSWITCH Summit is fast approaching. This gathering of the
> minds is where the FreeSWITCH developers hash out some of the big decisions
> that go into making FreeSWITCH what it is today. In less than two weeks,
> the FreeSWITCH team will descend onto a city in Wisconsin and spend a full
> week tirelessly living and breathing FreeSWITCH. From dawn, well maybe
> mid-morning, until dusk, the FreeSWITCH developers will work on some of the
> hardest and most complicated issues facing FreeSWITCH.
>
> We are asking for your help. In order to keep the team bright-eyed and
> bushy-tailed we must keep them fed. This is costly for the growing team.
> Show your appreciation donating so they can keep doing what they do best. Support
> the people that brought you the latest 1.6.14 release in all its glory. Any
> and all donations are welcome.
>
> <http://goog_1384205596>
> https://www.gofundme.com/freeswitch-developer-meeting-2017
>
>
> Kathleen King
> FreeSWITCH Public Relations
> Office: +1-213-286-0400
> Mobile: +1-703-859-3757
> http://freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From john.nash778 at gmail.com  Wed Jan 18 14:45:52 2017
From: john.nash778 at gmail.com (John Nash)
Date: Wed, 18 Jan 2017 17:15:52 +0530
Subject: [Freeswitch-users] mod_nibblebill question
Message-ID: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>

I am using latest freeswitch and trying to test mod_nibblebill. I am trying
to bill only b leg and it seems to cut the amount as per rates and
heartbeat. But it does not restrict call if balance is 0 or in minus. I was
hoping it will cut the calls the moment it reaches 0


        session:execute("export", "nolocal:enable_heartbeat_events=60")
        session:execute("export", "nolocal:nibble_account=1")
        session:execute("export", "nolocal:nibble_increment=1")
        session:execute("export", "nolocal:nibble_rate=0.02")


Is there anything I am doing wrong?
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From andrew at cassidywebservices.co.uk  Wed Jan 18 15:20:04 2017
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 18 Jan 2017 12:20:04 +0000
Subject: [Freeswitch-users] mod_nibblebill question
In-Reply-To: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
References: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
Message-ID: <CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>

you need nobal_action and nobal_amt

https://freeswitch.org/confluence/display/FREESWITCH/mod_nibblebill#mod_nibblebill-HangupCallWhenBalanceDepleted

On 18 January 2017 at 11:45, John Nash <john.nash778 at gmail.com> wrote:

> I am using latest freeswitch and trying to test mod_nibblebill. I am
> trying to bill only b leg and it seems to cut the amount as per rates and
> heartbeat. But it does not restrict call if balance is 0 or in minus. I was
> hoping it will cut the calls the moment it reaches 0
>
>
>         session:execute("export", "nolocal:enable_heartbeat_events=60")
>         session:execute("export", "nolocal:nibble_account=1")
>         session:execute("export", "nolocal:nibble_increment=1")
>         session:execute("export", "nolocal:nibble_rate=0.02")
>
>
> Is there anything I am 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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director

03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk>
www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
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From john.nash778 at gmail.com  Wed Jan 18 15:33:26 2017
From: john.nash778 at gmail.com (John Nash)
Date: Wed, 18 Jan 2017 18:03:26 +0530
Subject: [Freeswitch-users] mod_nibblebill question
In-Reply-To: <CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>
References: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
	<CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>
Message-ID: <CAA4rGPzDmNTXBni9gHhuU25mRx_RoMkuKPdvT3=vHL=JuoYLRg@mail.gmail.com>

Sorry i missed that but now A leg hangs up but call still goes to the
destination (b leg)

On Wed, Jan 18, 2017 at 5:50 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> you need nobal_action and nobal_amt
>
> https://freeswitch.org/confluence/display/FREESWITCH/
> mod_nibblebill#mod_nibblebill-HangupCallWhenBalanceDepleted
>
> On 18 January 2017 at 11:45, John Nash <john.nash778 at gmail.com> wrote:
>
>> I am using latest freeswitch and trying to test mod_nibblebill. I am
>> trying to bill only b leg and it seems to cut the amount as per rates and
>> heartbeat. But it does not restrict call if balance is 0 or in minus. I was
>> hoping it will cut the calls the moment it reaches 0
>>
>>
>>         session:execute("export", "nolocal:enable_heartbeat_events=60")
>>         session:execute("export", "nolocal:nibble_account=1")
>>         session:execute("export", "nolocal:nibble_increment=1")
>>         session:execute("export", "nolocal:nibble_rate=0.02")
>>
>>
>> Is there anything I am 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
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
> 03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk> ww
> w.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From john.nash778 at gmail.com  Wed Jan 18 15:59:21 2017
From: john.nash778 at gmail.com (John Nash)
Date: Wed, 18 Jan 2017 18:29:21 +0530
Subject: [Freeswitch-users] mod_nibblebill question
In-Reply-To: <CAA4rGPzDmNTXBni9gHhuU25mRx_RoMkuKPdvT3=vHL=JuoYLRg@mail.gmail.com>
References: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
	<CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>
	<CAA4rGPzDmNTXBni9gHhuU25mRx_RoMkuKPdvT3=vHL=JuoYLRg@mail.gmail.com>
Message-ID: <CAA4rGPwQ3jp8vRo3GmBPKHr_mLbL5uk=6QJnTozLHHkj+DoD0Q@mail.gmail.com>

Just to add I am exporting the nibblebill variables before I call bridge.

On Wed, Jan 18, 2017 at 6:03 PM, John Nash <john.nash778 at gmail.com> wrote:

> Sorry i missed that but now A leg hangs up but call still goes to the
> destination (b leg)
>
> On Wed, Jan 18, 2017 at 5:50 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> you need nobal_action and nobal_amt
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_
>> nibblebill#mod_nibblebill-HangupCallWhenBalanceDepleted
>>
>> On 18 January 2017 at 11:45, John Nash <john.nash778 at gmail.com> wrote:
>>
>>> I am using latest freeswitch and trying to test mod_nibblebill. I am
>>> trying to bill only b leg and it seems to cut the amount as per rates and
>>> heartbeat. But it does not restrict call if balance is 0 or in minus. I was
>>> hoping it will cut the calls the moment it reaches 0
>>>
>>>
>>>         session:execute("export", "nolocal:enable_heartbeat_events=60")
>>>         session:execute("export", "nolocal:nibble_account=1")
>>>         session:execute("export", "nolocal:nibble_increment=1")
>>>         session:execute("export", "nolocal:nibble_rate=0.02")
>>>
>>>
>>> Is there anything I am 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
>>>
>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>> 03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk>
>> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From john.nash778 at gmail.com  Wed Jan 18 16:10:02 2017
From: john.nash778 at gmail.com (John Nash)
Date: Wed, 18 Jan 2017 18:40:02 +0530
Subject: [Freeswitch-users] mod_nibblebill question
In-Reply-To: <CAA4rGPwQ3jp8vRo3GmBPKHr_mLbL5uk=6QJnTozLHHkj+DoD0Q@mail.gmail.com>
References: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
	<CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>
	<CAA4rGPzDmNTXBni9gHhuU25mRx_RoMkuKPdvT3=vHL=JuoYLRg@mail.gmail.com>
	<CAA4rGPwQ3jp8vRo3GmBPKHr_mLbL5uk=6QJnTozLHHkj+DoD0Q@mail.gmail.com>
Message-ID: <CAA4rGPy9v7HF+1hMGAiUxk_NQtPq_Fi-gV79KedVqqGjzX2jXw@mail.gmail.com>

When I tested more and saw wireshark traces, I found that it does not
happen every time and it also does not appear to be Feeswitch issue as I
can see "CANCEL" is being sent to other party just after Invite.

On Wed, Jan 18, 2017 at 6:29 PM, John Nash <john.nash778 at gmail.com> wrote:

> Just to add I am exporting the nibblebill variables before I call bridge.
>
> On Wed, Jan 18, 2017 at 6:03 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> Sorry i missed that but now A leg hangs up but call still goes to the
>> destination (b leg)
>>
>> On Wed, Jan 18, 2017 at 5:50 PM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> you need nobal_action and nobal_amt
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_nib
>>> blebill#mod_nibblebill-HangupCallWhenBalanceDepleted
>>>
>>> On 18 January 2017 at 11:45, John Nash <john.nash778 at gmail.com> wrote:
>>>
>>>> I am using latest freeswitch and trying to test mod_nibblebill. I am
>>>> trying to bill only b leg and it seems to cut the amount as per rates and
>>>> heartbeat. But it does not restrict call if balance is 0 or in minus. I was
>>>> hoping it will cut the calls the moment it reaches 0
>>>>
>>>>
>>>>         session:execute("export", "nolocal:enable_heartbeat_events=60")
>>>>         session:execute("export", "nolocal:nibble_account=1")
>>>>         session:execute("export", "nolocal:nibble_increment=1")
>>>>         session:execute("export", "nolocal:nibble_rate=0.02")
>>>>
>>>>
>>>> Is there anything I am 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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>> 03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk>
>>> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From gmaruzz at gmail.com  Wed Jan 18 16:48:51 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 18 Jan 2017 14:48:51 +0100
Subject: [Freeswitch-users] Be like Bro,
	donate to the core developers Annual FreeSWITCH Summit
Message-ID: <CALXCt0qoFhW_50MwSyPrr4ZUyh-Ej54RxQ49fm33iegO8r4MSg@mail.gmail.com>

Be like Bro, donate to the core developers!

(btw, if you - like me and like Bro - prefer to directly use PayPal, there
is a "donate" link on top right of http://www.freeswitch.org )



On 18 January 2017 at 10:57, ?talo Rossi <italo at freeswitch.org> wrote:

> Hey guys excellent opportunity to show your support and gratitude to
> FreeSWITCH team!
>
> Go go! :-)
> Em ter, 17 de jan de 2017 ?s 16:17, Kathleen King <kathleen at freeswitch.org>
> escreveu:
>
>> Hello FreeSWITCHers,
>>
>> As some of you may know the FreeSWITCH team gets together every year and
>> the annual FreeSWITCH Summit is fast approaching. This gathering of the
>> minds is where the FreeSWITCH developers hash out some of the big decisions
>> that go into making FreeSWITCH what it is today. In less than two weeks,
>> the FreeSWITCH team will descend onto a city in Wisconsin and spend a full
>> week tirelessly living and breathing FreeSWITCH. From dawn, well maybe
>> mid-morning, until dusk, the FreeSWITCH developers will work on some of the
>> hardest and most complicated issues facing FreeSWITCH.
>>
>> We are asking for your help. In order to keep the team bright-eyed and
>> bushy-tailed we must keep them fed. This is costly for the growing team.
>> Show your appreciation donating so they can keep doing what they do best. Support
>> the people that brought you the latest 1.6.14 release in all its glory. Any
>> and all donations are welcome.
>>
>> <http://goog_1384205596>
>> https://www.gofundme.com/freeswitch-developer-meeting-2017
>>
>>
>> Kathleen King
>> FreeSWITCH Public Relations
>> Office: +1-213-286-0400 <(213)%20286-0400>
>> Mobile: +1-703-859-3757 <(703)%20859-3757>
>> http://freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From my.post at hotmail.com  Wed Jan 18 17:23:52 2017
From: my.post at hotmail.com (Pavel)
Date: Wed, 18 Jan 2017 14:23:52 +0000
Subject: [Freeswitch-users] Some inputs on mod_verto neeed.
Message-ID: <DB6PR0601MB234364125604E4259C6B92FAED7F0@DB6PR0601MB2343.eurprd06.prod.outlook.com>

Hello everyone,


Can someone, please, shed some light how to implement following features, using mod_verto (in broad strokes, just some directions):


1) Have 2 users logged in using mod_verto, how to send text message between them using verto.js library?

2) Have a user logged in using mod_verto, how to receive presence notifications in user's browser about other verto users state ? Other SIP users state ?



Any thoughts will be much appreciated.

Thanks.


---

Pavel.
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From vbvbrj at gmail.com  Wed Jan 18 17:35:26 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 18 Jan 2017 16:35:26 +0200
Subject: [Freeswitch-users] mod_nibblebill question
In-Reply-To: <CAA4rGPy9v7HF+1hMGAiUxk_NQtPq_Fi-gV79KedVqqGjzX2jXw@mail.gmail.com>
References: <CAA4rGPybc8tZ33mx-hPxC4=UXEahdn7W_FBcnJeQ1EuQHvy1uA@mail.gmail.com>
	<CAMbW9oLoO8VEydPytw3uWaU5BK-FGm6+rJTrMfKtyJ6Rv7jWQg@mail.gmail.com>
	<CAA4rGPzDmNTXBni9gHhuU25mRx_RoMkuKPdvT3=vHL=JuoYLRg@mail.gmail.com>
	<CAA4rGPwQ3jp8vRo3GmBPKHr_mLbL5uk=6QJnTozLHHkj+DoD0Q@mail.gmail.com>
	<CAA4rGPy9v7HF+1hMGAiUxk_NQtPq_Fi-gV79KedVqqGjzX2jXw@mail.gmail.com>
Message-ID: <2e34afba-e7cb-ee41-3f92-68bbb249346f@gmail.com>

On 18.01.2017 15:10, John Nash wrote:
> Sorry i missed that but now A leg hangs up but call still goes to the
> destination (b leg)

I have almost same problem. Billing must be made on B-leg. If A-leg 
hangs-up, B-leg is going on till answer and hang-up, or a timeout. But 
I'm on very old version of FS. It seems that this problem was resolved 
in new versions.

-- 
Mimiko desu.


From rick at magicmail.mooo.com  Wed Jan 18 19:40:29 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 18 Jan 2017 16:40:29 +0000
Subject: [Freeswitch-users] Trunk for inbound connections
Message-ID: <5F3ADC17-7A7E-486A-90DA-9676BB82C0D4@magicmail.mooo.com>

What is the best and cleanest method for providing a trunk for someone else to register into, in the most transparent way (i.e. cross-compatible, for providing a ITSP service to third party systems)? 

From tayeb.meftah at gmail.com  Wed Jan 18 20:37:23 2017
From: tayeb.meftah at gmail.com (tayeb.meftah at gmail.com)
Date: Wed, 18 Jan 2017 18:37:23 +0100
Subject: [Freeswitch-users] Be like Bro,
	donate to the core developers Annual FreeSWITCH Summit
In-Reply-To: <CALXCt0qoFhW_50MwSyPrr4ZUyh-Ej54RxQ49fm33iegO8r4MSg@mail.gmail.com>
References: <CALXCt0qoFhW_50MwSyPrr4ZUyh-Ej54RxQ49fm33iegO8r4MSg@mail.gmail.com>
Message-ID: <317A8C00-D57F-4DFB-B2AE-4085A6CB0D63@gmail.com>

+1

Envoy? de mon iPhone

> Le 18 janv. 2017 ? 14:48, Giovanni Maruzzelli <gmaruzz at gmail.com> a ?crit :
> 
> Be like Bro, donate to the core developers! 
> 
> (btw, if you - like me and like Bro - prefer to directly use PayPal, there is a "donate" link on top right of http://www.freeswitch.org )
> 
> 
> 
>> On 18 January 2017 at 10:57, ?talo Rossi <italo at freeswitch.org> wrote:
>> Hey guys excellent opportunity to show your support and gratitude to FreeSWITCH team!
>> 
>> Go go! :-)
>>> Em ter, 17 de jan de 2017 ?s 16:17, Kathleen King <kathleen at freeswitch.org> escreveu:
>>> Hello FreeSWITCHers,
>>> 
>>> As some of you may know the FreeSWITCH team gets together every year and the annual FreeSWITCH Summit is fast approaching. This gathering of the minds is where the FreeSWITCH developers hash out some of the big decisions that go into making FreeSWITCH what it is today. In less than two weeks, the FreeSWITCH team will descend onto a city in Wisconsin and spend a full week tirelessly living and breathing FreeSWITCH. From dawn, well maybe mid-morning, until dusk, the FreeSWITCH developers will work on some of the hardest and most complicated issues facing FreeSWITCH. 
>>> 
>>> We are asking for your help. In order to keep the team bright-eyed and bushy-tailed we must keep them fed. This is costly for the growing team. Show your appreciation donating so they can keep doing what they do best. Support the people that brought you the latest 1.6.14 release in all its glory. Any and all donations are welcome. 
>>> 
>>> https://www.gofundme.com/freeswitch-developer-meeting-2017 
>>> 
>>> 
>>> Kathleen King
>>> FreeSWITCH Public Relations
>>> Office: +1-213-286-0400
>>> Mobile: +1-703-859-3757
>>> http://freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Wed Jan 18 21:18:18 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 18 Jan 2017 18:18:18 +0000
Subject: [Freeswitch-users] Trunk for inbound connections
In-Reply-To: <5F3ADC17-7A7E-486A-90DA-9676BB82C0D4@magicmail.mooo.com>
References: <5F3ADC17-7A7E-486A-90DA-9676BB82C0D4@magicmail.mooo.com>
Message-ID: <CAHtxdDeONRo-ODHXRKZjzRPLB19RsBiWKNQ+YF_Hf0ZxG5JZCw@mail.gmail.com>

I use assignments ip address of remote system to FreeSwith user in
directory.

??, 18 ???. 2017, 19:41 Rick Jarvis <rick at magicmail.mooo.com>:

> What is the best and cleanest method for providing a trunk for someone
> else to register into, in the most transparent way (i.e. cross-compatible,
> for providing a ITSP service to third party systems)?
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rick at magicmail.mooo.com  Wed Jan 18 21:25:34 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Wed, 18 Jan 2017 18:25:34 +0000
Subject: [Freeswitch-users] Trunk for inbound connections
In-Reply-To: <CAHtxdDeONRo-ODHXRKZjzRPLB19RsBiWKNQ+YF_Hf0ZxG5JZCw@mail.gmail.com>
References: <5F3ADC17-7A7E-486A-90DA-9676BB82C0D4@magicmail.mooo.com>
	<CAHtxdDeONRo-ODHXRKZjzRPLB19RsBiWKNQ+YF_Hf0ZxG5JZCw@mail.gmail.com>
Message-ID: <1807D724-8BEF-473A-B96B-C172110FAA5F@magicmail.mooo.com>

If I just set up a user, I get a problem where ?gw+? is being added to the SIP header, so calls from me to the third party system fail? something that?s better described here: https://www.3cx.com/community/threads/configuring-freeswitch-as-a-voice-provider.19656/ <https://www.3cx.com/community/threads/configuring-freeswitch-as-a-voice-provider.19656/>

That?s why I started wondering if creating a user wasn?t the right way to go..?


> On 18 Jan 2017, at 18:18, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> I use assignments ip address of remote system to FreeSwith user in directory.
> 
> 
> ??, 18 ???. 2017, 19:41 Rick Jarvis <rick at magicmail.mooo.com <mailto:rick at magicmail.mooo.com>>:
> What is the best and cleanest method for providing a trunk for someone else to register into, in the most transparent way (i.e. cross-compatible, for providing a ITSP service to third party systems)?
> _________________________________________________________________________
> 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 krice at freeswitch.org  Thu Jan 19 00:51:12 2017
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 18 Jan 2017 15:51:12 -0600
Subject: [Freeswitch-users] Annual FreeSWITCH Summit
In-Reply-To: <CABMX_w90qtQA9-Zy8O-2EaOBtEkCUu23=YhYj=cnmoOrLmx7ag@mail.gmail.com>
References: <CAOStUPb9_UmC-M_xLE5Q5tyMdeMcNezuPjkMt8K4xn9Dvi2Hjg@mail.gmail.com>
	<CABMX_w90qtQA9-Zy8O-2EaOBtEkCUu23=YhYj=cnmoOrLmx7ag@mail.gmail.com>
Message-ID: <034201d271d4$fc6d6980$f5483c80$@freeswitch.org>

Hey FreeSWITCHers,

 

Just looking over at the GoFundMe page and it looks like some of you have visited tossed a few bucks in the pile (Thanks guys!). So far we have collected enough to feed them one good meal, and maybe a couple of bags of chips for the rest of the week!

 

We?re trying to not have the guys eating MRE?s and McDonalds or Jack in the Box every day! 

 

Don?t forget keeping the dev?s fed keeps them coding and not digging around in the kitchen for a snack!

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Wednesday, January 18, 2017 3:58 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Annual FreeSWITCH Summit

 

Hey guys excellent opportunity to show your support and gratitude to FreeSWITCH team!

Go go! :-)

Em ter, 17 de jan de 2017 ?s 16:17, Kathleen King <kathleen at freeswitch.org <mailto:kathleen at freeswitch.org> > escreveu:

Hello FreeSWITCHers,

 

As some of you may know the FreeSWITCH team gets together every year and the annual FreeSWITCH Summit is fast approaching. This gathering of the minds is where the FreeSWITCH developers hash out some of the big decisions that go into making FreeSWITCH what it is today. In less than two weeks, the FreeSWITCH team will descend onto a city in Wisconsin and spend a full week tirelessly living and breathing FreeSWITCH. From dawn, well maybe mid-morning, until dusk, the FreeSWITCH developers will work on some of the hardest and most complicated issues facing FreeSWITCH. 


We are asking for your help. In order to keep the team bright-eyed and bushy-tailed we must keep them fed. This is costly for the growing team. Show your appreciation donating so they can keep doing what they do best. Support the people that brought you the latest 1.6.14 release in all its glory. Any and all donations are welcome. 

 <http://goog_1384205596> 


https://www.gofundme.com/freeswitch-developer-meeting-2017 

 




Kathleen King

FreeSWITCH Public Relations

Office: +1-213-286-0400

Mobile: +1-703-859-3757

http://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 jungleboogie0 at gmail.com  Thu Jan 19 05:14:44 2017
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Wed, 18 Jan 2017 18:14:44 -0800
Subject: [Freeswitch-users] no external registrations?
Message-ID: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>

Hi All,

My freeswitch is registered to flowroute and I know this because I can 
login to my account and view the registration.
Why does this show it's not registered??

fs_cli -x "sofia status profile external"
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.0.137
SIP-IP                  192.168.0.137
URL                     sip:mod_sofia at 192.168.0.137:5080
BIND-URL                sip:mod_sofia at 192.168.0.137:5080;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               OPUS,G722,PCMU,PCMA,VP8
CODECS OUT              OPUS,G722,PCMU,PCMA,VP8
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               1
FAILED-CALLS-OUT        0
REGISTRATIONS           0


Thanks!


From rtreleaven at bunnykick.ca  Thu Jan 19 05:18:25 2017
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Wed, 18 Jan 2017 21:18:25 -0500
Subject: [Freeswitch-users] no external registrations?
In-Reply-To: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>
References: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>
Message-ID: <CAHzfiZ7cDoG66=ONmuLEwzHy9Vga3QoOJk0zkAfbx_qKWUmzZg@mail.gmail.com>

I think you want "Sofia status gateway"

On Jan 18, 2017 9:15 PM, "jungle boogie" <jungleboogie0 at gmail.com> wrote:

> Hi All,
>
> My freeswitch is registered to flowroute and I know this because I can
> login to my account and view the registration.
> Why does this show it's not registered??
>
> fs_cli -x "sofia status profile external"
> ============================================================
> =====================================
> Name                    external
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_external
> Pres Hosts
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  192.168.0.137
> SIP-IP                  192.168.0.137
> URL                     sip:mod_sofia at 192.168.0.137:5080
> BIND-URL                sip:mod_sofia at 192.168.0.137:5080;transport=udp,tcp
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               OPUS,G722,PCMU,PCMA,VP8
> CODECS OUT              OPUS,G722,PCMU,PCMA,VP8
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                true
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           true
> AGGRESSIVENAT           false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               1
> FAILED-CALLS-OUT        0
> REGISTRATIONS           0
>
>
> 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 jungleboogie0 at gmail.com  Thu Jan 19 05:33:56 2017
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Wed, 18 Jan 2017 18:33:56 -0800
Subject: [Freeswitch-users] no external registrations?
In-Reply-To: <CAHzfiZ7cDoG66=ONmuLEwzHy9Vga3QoOJk0zkAfbx_qKWUmzZg@mail.gmail.com>
References: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>
	<CAHzfiZ7cDoG66=ONmuLEwzHy9Vga3QoOJk0zkAfbx_qKWUmzZg@mail.gmail.com>
Message-ID: <e3638c0a-3706-fa87-ce96-7b12cb31358f@gmail.com>

On 01/18/2017 06:18 PM, Russell Treleaven wrote:
> I think you want "Sofia status gateway"
>

ah, that does it.

What's sofia status profile external for, then?


From krice at freeswitch.org  Thu Jan 19 05:48:08 2017
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 18 Jan 2017 20:48:08 -0600
Subject: [Freeswitch-users] no external registrations?
In-Reply-To: <e3638c0a-3706-fa87-ce96-7b12cb31358f@gmail.com>
References: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>
	<CAHzfiZ7cDoG66=ONmuLEwzHy9Vga3QoOJk0zkAfbx_qKWUmzZg@mail.gmail.com>
	<e3638c0a-3706-fa87-ce96-7b12cb31358f@gmail.com>
Message-ID: <8BCEE27A-B631-4CDE-BACE-E5774AE8E40C@freeswitch.org>

sofia status profile <profile_name> shows you the status and details of just the profile. There are additional commands to show the status of the gateway and list any users that may be connected to that profile.



Sent from my iPhone

> On Jan 18, 2017, at 20:33, jungle boogie <jungleboogie0 at gmail.com> wrote:
> 
>> On 01/18/2017 06:18 PM, Russell Treleaven wrote:
>> I think you want "Sofia status gateway"
>> 
> 
> ah, that does it.
> 
> What's sofia status profile external for, then?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From rtreleaven at bunnykick.ca  Thu Jan 19 05:53:19 2017
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Wed, 18 Jan 2017 21:53:19 -0500
Subject: [Freeswitch-users] no external registrations?
In-Reply-To: <e3638c0a-3706-fa87-ce96-7b12cb31358f@gmail.com>
References: <80563a88-7252-99e8-0d5b-ccbd4c2ecd4e@gmail.com>
	<CAHzfiZ7cDoG66=ONmuLEwzHy9Vga3QoOJk0zkAfbx_qKWUmzZg@mail.gmail.com>
	<e3638c0a-3706-fa87-ce96-7b12cb31358f@gmail.com>
Message-ID: <CAHzfiZ5_wujiE_oK1EVSeTNCJVwbszQww_kv4Xqkt1C_-Nm8pw@mail.gmail.com>

It shows the status of the external profile.

A gateway is an optional component of a profile.
A profile is a sip endpoint for your Freeswitch server.

The gateway is used to make a trunk with another server(Flowroute in your
case).

"A gateway describes how to use a different UA to reach destinations. For
example, the gateway may provide access to the PSTN, or to a private SIP
network" - https://wiki.freeswitch.org/wiki/Sofia.conf.xml#Gateway

I couldn't find the description on the confluence wiki so I linked to the
old one.




On Wed, Jan 18, 2017 at 9:33 PM, jungle boogie <jungleboogie0 at gmail.com>
wrote:

> On 01/18/2017 06:18 PM, Russell Treleaven wrote:
> > I think you want "Sofia status gateway"
> >
>
> ah, that does it.
>
> What's sofia status profile external for, then?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Russell Treleaven
sip:rtreleaven at sip.bunnykick.ca
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From devang.nathwani31589 at gmail.com  Thu Jan 19 08:20:40 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Thu, 19 Jan 2017 10:50:40 +0530
Subject: [Freeswitch-users] ss7 integration with freeswitch
Message-ID: <CAGOm0PJE=aXJwp7UHG8Xb8WwPhuhoqFz1S85UWzTaZZ_19r7Dw@mail.gmail.com>

Hello,

Where to start to integrate ss7 with freeswitch?
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From devang.nathwani31589 at gmail.com  Thu Jan 19 08:26:57 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Thu, 19 Jan 2017 10:56:57 +0530
Subject: [Freeswitch-users] ss7 integration with freeswitch
In-Reply-To: <CAGOm0PJE=aXJwp7UHG8Xb8WwPhuhoqFz1S85UWzTaZZ_19r7Dw@mail.gmail.com>
References: <CAGOm0PJE=aXJwp7UHG8Xb8WwPhuhoqFz1S85UWzTaZZ_19r7Dw@mail.gmail.com>
Message-ID: <CAGOm0P+aoPPOnPMQoB8JaMJUGCS5AqU5R52JSqSqwTbsa+qyfA@mail.gmail.com>

https://wiki.freeswitch.org/wiki/FreeTDM  ?

On Thu, Jan 19, 2017 at 10:50 AM, devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> Hello,
>
> Where to start to integrate ss7 with freeswitch?
>
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From igorolhovskiy at gmail.com  Thu Jan 19 12:00:43 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 19 Jan 2017 11:00:43 +0200
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <CAL+DTgKmzBm_qVDPaGKjuKj6=oCwudo6fAUPBO6Yw_5GAQ4OBQ@mail.gmail.com>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<CAPfjz3jTGuFp3Lx4rAwCksm4Bu1yE5KhySbJQ9hbs94Y0r3MEQ@mail.gmail.com>
	<CAL+DTgKmzBm_qVDPaGKjuKj6=oCwudo6fAUPBO6Yw_5GAQ4OBQ@mail.gmail.com>
Message-ID: <788a0c9d-0ca3-48af-b822-a9d47a0a19a8@Spark>

Who:
Consertis GMbH (consertis.at)
What:
FlowPhone CloudPBX service (flowphone.at). Multitenant scalable PBX based on FreeSWITCH.
Where:
Austria

Actually, it?s more to FusionPBX, than plain FreeSWITCH. But ever the less.

Regards, Igor

On 18 ???. 2017 ?., 2:24 +0200, Gonzalo Gasca Meza <gascagonzalo at gmail.com>, wrote:
> Who
> At Parzee?are using Freeswitch for 2 different projects:
>
> What
> Products:
> Attorney: Provides Visual Voicemail for Cisco Spark clients. Use Freeswitch VoiceMail API to monitor user's Inbox and Speech to Text third party APIs to create voicemail transcriptions.
> Telephonist: Advanced call dialer to detect fixed number status. Use Freeswitch ESL, Recording and?mod_spandsp. Production ~10 CPS scalable to 50 CPS. Stack: Python, AWS, PostgreSQL.
>
> Where
> Different customers across Canada, Brazil, Mexico, and US.
>
> We can provide diagrams and performance statistics.
>
> Gonzalo and Parzee team info at parzee.com
>
>
>
>
>
> > On Tue, Jan 17, 2017 at 12:55 PM, Jai Rangi <jprangi at didforsale.com> wrote:
> > > We are using freeswitch for 2 different products:
> > >
> > > www.cebodtelecom.com
> > > (Hosted phone service) Multi-tenant phone system, hooked with mysql cluster, and shared storage. Most complex features are manage with LUA scripts. Phone service offers almost everything possible phone system features.? (IVR, Groups, Conferencing, Call Center, Call Queue, Timebased routing, Failover Routing, fax to email as well as fax to fax machine)). Customers are using Yealink, GS, Polycom, Cisco and different softphones.
> > >
> > >
> > > www.didforsale.com
> > > (Sip Trunking service provider) Most of our customer have their own PBX and we dont stay in media. We use freeswitch for customers where we have to stay in media and customers who are behind NAT, wants to use softphones and call forwarding etc.
> > > One large project we have recently implemented is to offer a platform similar to twilio and plivo. Where customers can build voice applications on their web servers and use our platform to complete the calls. https://www.didforsale.com/developer
> > > Technology used, (Mysql Cluster, Shared Storage, LUA, Curl)
> > >
> > > Indeed freeeswitch is great platform and possibilities are endless. Online community great. Freeswitch Books (Core as well as cookbook) are the best tools to learn freeswitch.
> > >
> > > Hope this will be useful, I will be happy to provide some Snapshot if that adds any value to new users.
> > >
> > > Best,
> > >
> > > Jai Rangi
> > > Cebod Technologies LLC dba DIDforSale/Cebod Telecom
> > > O 949-471-0102 | C 949-419-7634 | F 949-269-0449 / 949-232-1410 | jprangi at didforsale.com? www.cebod.com |? www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626? |
> > >
> > >
> > >
> > >
> > >
> > >
> > > > On Tue, Jan 17, 2017 at 7:44 AM, Ken Rice <krice at freeswitch.org> wrote:
> > > > > Hey Guys,
> > > > >
> > > > > A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering FreeSWITCH for the first time.
> > > > >
> > > > > I would like to build a list of examples. If you don?t mind this info being compiled on Confluence.
> > > > > Please ?Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or FreeSWITCH using deployments out there.
> > > > >
> > > > > Thanks
> > > > > Ken
> > > > >
> > > > > _________________________________________________________________________
> > > > > Professional FreeSWITCH Consulting Services:
> > > > > consulting at freeswitch.org
> > > > > http://www.freeswitchsolutions.com
> > > > >
> > > > > Official FreeSWITCH Sites
> > > > > http://www.freeswitch.org
> > > > > http://confluence.freeswitch.org
> > > > > http://www.cluecon.com
> > > > >
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > >
> > >
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From v.kovalyshyn at gmail.com  Thu Jan 19 12:15:35 2017
From: v.kovalyshyn at gmail.com (Vitaly Kovalyshyn)
Date: Thu, 19 Jan 2017 11:15:35 +0200
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <788a0c9d-0ca3-48af-b822-a9d47a0a19a8@Spark>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<CAPfjz3jTGuFp3Lx4rAwCksm4Bu1yE5KhySbJQ9hbs94Y0r3MEQ@mail.gmail.com>
	<CAL+DTgKmzBm_qVDPaGKjuKj6=oCwudo6fAUPBO6Yw_5GAQ4OBQ@mail.gmail.com>
	<788a0c9d-0ca3-48af-b822-a9d47a0a19a8@Spark>
Message-ID: <8e5eb8df-03f1-4fdd-8c31-8a6fb13183f0@Spark>

Who:
Webitel (http://webitel.ua/?or?http://webitel.com/)

What:
Webitel -?Multitenant?scalable, cloud-based VoIP telephony platform?based on FreeSWITCH

Technology: Docker, FreeSWITCH, Node.js, MongoDB, Elasticsearch and Kibana.
http://api.webitel.com/en/latest/
https://docs.webitel.com/display/W3/Webitel+WebClient

Verto Phone for Chrome -?https://docs.webitel.com/display/W3/Webitel+Verto+Phone


Where:
We are from?Ukraine.
Customers?across Ukraine, Russian,?APAC?and USA.


Best regards,
Vitaly Kovalyshyn

> > > >
> > > > > On Tue, Jan 17, 2017 at 7:44 AM, Ken Rice <krice at freeswitch.org> wrote:
> > > > > > Hey Guys,
> > > > > >
> > > > > > A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering FreeSWITCH for the first time.
> > > > > >
> > > > > > I would like to build a list of examples. If you don?t mind this info being compiled on Confluence.
> > > > > > Please ?Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or FreeSWITCH using deployments out there.
> > > > > >
> > > > > > Thanks
> > > > > > Ken
> > > > > >
> > > > > > _________________________________________________________________________
> > > > > > Professional FreeSWITCH Consulting Services:
> > > > > > consulting at freeswitch.org
> > > > > > http://www.freeswitchsolutions.com
> > > > > >
> > > > > > Official FreeSWITCH Sites
> > > > > > http://www.freeswitch.org
> > > > > > http://confluence.freeswitch.org
> > > > > > http://www.cluecon.com
> > > > > >
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > http://www.freeswitch.org
> > > >
> > > >
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org
> > > > http://www.freeswitchsolutions.com
> > > >
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://confluence.freeswitch.org
> > > > http://www.cluecon.com
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shaun.stokes at itec-support.co.uk  Thu Jan 19 14:49:17 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Thu, 19 Jan 2017 11:49:17 +0000
Subject: [Freeswitch-users] Best way to unset\clear export variable on
	FreeSWITCH
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E862A260@mbx-01.sysconfig.co.uk>

Hi All,

We're using execute_on_answer in our dialplan as follows:
<action application="export" data=" nolocal:execute_on_answer=record_session ${rec_file}"/>

However, under certain conditions we want to record the session immediately while the A leg is being processed by a LUA script (before we begin B leg), doing so causes a bug in the audio file as the call is actually recorded twice. As a result we need to unset\clear the execute_on_answer variable.

I was thinking of just setting re-exporting it with a null value, but as I can't find any instances of this being done before I'm not sure if this is best practice for FreeSWITCH:
session:execute("export"," nolocal:execute_on_answer=");

What is the correct way to unset\clear the above execute_on_answer variable in LUA?

Thanks in advance,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
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From rick at magicmail.mooo.com  Thu Jan 19 15:13:55 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 19 Jan 2017 12:13:55 +0000
Subject: [Freeswitch-users] 123456
Message-ID: <2BFFE67C-8097-406B-8D97-99639577B189@magicmail.mooo.com>

I see a lot of these hitting FS continuously - where would they be originating from? (The IP x.x.x.x is the local IP)

2017-01-19 12:06:07.773935 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/123456 at x.x.x.x [6ed9ba7d-4357-460d-a0b7-ff94bf871e7e]
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From shaun.stokes at itec-support.co.uk  Thu Jan 19 15:35:51 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Thu, 19 Jan 2017 12:35:51 +0000
Subject: [Freeswitch-users] 123456
In-Reply-To: <2BFFE67C-8097-406B-8D97-99639577B189@magicmail.mooo.com>
References: <2BFFE67C-8097-406B-8D97-99639577B189@magicmail.mooo.com>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E862A390@mbx-01.sysconfig.co.uk>

This is a new channel being opened to an endpoint (123456 at x.x.x.x<mailto:123456 at x.x.x.x>), the endpoint is on your internal SIP profile.

An example of this might be a call routing to extension 123456 at x.x.x.x<mailto:123456 at x.x.x.x> which is registered to your internal SIP profile.

You should work your way up through the logs, providing the information is being logged you should be able to identify why the call was routed to this extension.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis
Sent: 19 January 2017 12:14
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] 123456

I see a lot of these hitting FS continuously - where would they be originating from? (The IP x.x.x.x is the local IP)

2017-01-19 12:06:07.773935 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/123456 at x.x.x.x<mailto:sofia/internal/123456 at x.x.x.x> [6ed9ba7d-4357-460d-a0b7-ff94bf871e7e]

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________
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From tculjaga at gmail.com  Thu Jan 19 17:07:43 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 19 Jan 2017 15:07:43 +0100
Subject: [Freeswitch-users] Malformed From, fix your code!
Message-ID: <CAJoKKdUNT2J5sj+RKnqgRMv9EacSE-U_nLvygk+ux494YzbY9g@mail.gmail.com>

hi,


does anyone experience the same issue ?
im running FreeSWITCH Version 1.6.14+git~20170112T174802Z~a8d53fdd79~64bit
(git a8d53fd 2017-01-12 17:48:02Z 64bit)



im trying to originate a call like this:

bgapi originate [origination_caller_id_number=1234]sofia/internal/
1002 at 192.168.254.168 &echo()


it looks like FS brakes the from header ... pls note the extra space
between the IP address and the closing ">"


   ------------------------------------------------------------------------
send 1009 bytes to udp/[192.168.254.168]:5060 at 15:02:13.408606:
   ------------------------------------------------------------------------
   INVITE sip:1002 at 192.168.254.168 SIP/2.0
   Via: SIP/2.0/UDP 192.168.254.73;rport;branch=z9hG4bKHr6ea76mt53QQ
   Max-Forwards: 70
   From: "" <sip:1234 at 192.168.254.73 >;tag=Q0SvF26gNteKa
   To: <sip:1002 at 192.168.254.168>
   Call-ID: a588ad14-58f2-1235-07b5-525400e95788
   CSeq: 102081187 INVITE
   Contact: <sip:mod_sofia at 192.168.254.73 :5060>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.14+git~20170112T174802Z~a8d53fdd79~64bit
   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: 224
   X-FS-Support: update_display,send_info
   Remote-Party-ID: <sip:1234 at 192.168.254.73
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1484806170 1484806171 IN IP4 192.168.254.73
   s=FreeSWITCH
   c=IN IP4 192.168.254.73
   t=0 0
   m=audio 28332 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 286 bytes from udp/[192.168.254.168]:5060 at 15:02:13.427078:
   ------------------------------------------------------------------------
   SIP/2.0 400 Malformed From, fix your code!
   Via: SIP/2.0/UDP 192.168.254.73;rport=5060;branch=z9hG4bKHr6ea76mt53QQ
   To: <sip:1002 at 192.168.254.168>;tag=c8797354
   From: <sip:1234 at 192.168.254.73>
   Call-ID: a588ad14-58f2-1235-07b5-525400e95788
   CSeq: 102081187 INVITE
   Content-Length: 0
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From vbvbrj at gmail.com  Thu Jan 19 17:36:19 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 19 Jan 2017 16:36:19 +0200
Subject: [Freeswitch-users] Nested conditions and anti-actions
Message-ID: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>

Hello.

This is a snipped:

<condition field="destination_number" expression="^(1001|1002)$" require-nested="false">
  <condition field="${effective_caller_id_name}" expression="^.+$" break="never">
   <action application="export" data="caller_id_name=${effective_caller_id_name} > ${destination_number}"/>
   <action application="export" data="effective_caller_id_name=${effective_caller_id_name} > ${destination_number}"/>
   <anti-action application="export" data="caller_id_name=${caller_id_name} > ${destination_number}"/>
   <anti-action application="export" data="effective_caller_id_name=${caller_id_name} > ${destination_number}"/>
  </condition>
<action .... />
</condition>

The problem I'm facing is that anti-actions are evaluated even if destination_number is not equal to 1001 or 1002. But I need that ant-actions will 
evaluate only when parent condition is true.

Is this a bug or it is by design?

-- 
Mimiko desu.


From bipin at xbipin.com  Thu Jan 19 18:13:50 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 19 Jan 2017 19:13:50 +0400
Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
Message-ID: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>

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From mike at jerris.com  Thu Jan 19 19:01:17 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 19 Jan 2017 11:01:17 -0500
Subject: [Freeswitch-users] Malformed From, fix your code!
In-Reply-To: <CAJoKKdUNT2J5sj+RKnqgRMv9EacSE-U_nLvygk+ux494YzbY9g@mail.gmail.com>
References: <CAJoKKdUNT2J5sj+RKnqgRMv9EacSE-U_nLvygk+ux494YzbY9g@mail.gmail.com>
Message-ID: <33CB0B18-9255-46F1-9309-EC050F08CC6C@jerris.com>

grep your configs for that ip address.  Sounds like you have it somewhere with a space after it.

> On Jan 19, 2017, at 9:07 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> 
> hi,
> 
> 
> does anyone experience the same issue ?
> im running FreeSWITCH Version 1.6.14+git~20170112T174802Z~a8d53fdd79~64bit (git a8d53fd 2017-01-12 17:48:02Z 64bit)
> 
> 
> 
> im trying to originate a call like this:
> 
> bgapi originate [origination_caller_id_number=1234]sofia/internal/1002 at 192.168.254.168 <mailto:1002 at 192.168.254.168> &echo()
> 
> 
> it looks like FS brakes the from header ... pls note the extra space between the IP address and the closing ">"
> 
> 
>    ------------------------------------------------------------------------
> send 1009 bytes to udp/[192.168.254.168]:5060 at 15:02:13.408606:
>    ------------------------------------------------------------------------
>    INVITE sip:1002 at 192.168.254.168 <mailto:sip%3A1002 at 192.168.254.168> SIP/2.0
>    Via: SIP/2.0/UDP 192.168.254.73;rport;branch=z9hG4bKHr6ea76mt53QQ
>    Max-Forwards: 70
>    From: "" <sip:1234 at 192.168.254.73 <mailto:sip%3A1234 at 192.168.254.73> >;tag=Q0SvF26gNteKa
>    To: <sip:1002 at 192.168.254.168 <mailto:sip%3A1002 at 192.168.254.168>>
>    Call-ID: a588ad14-58f2-1235-07b5-525400e95788
>    CSeq: 102081187 INVITE
>    Contact: <sip:mod_sofia at 192.168.254.73 <mailto:sip%3Amod_sofia at 192.168.254.73> :5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.6.14+git~20170112T174802Z~a8d53fdd79~64bit
>    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: 224
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: <sip:1234 at 192.168.254.73 <mailto:sip%3A1234 at 192.168.254.73> >;party=calling;screen=yes;privacy=off
>    
>    v=0
>    o=FreeSWITCH 1484806170 1484806171 IN IP4 192.168.254.73
>    s=FreeSWITCH
>    c=IN IP4 192.168.254.73
>    t=0 0
>    m=audio 28332 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> recv 286 bytes from udp/[192.168.254.168]:5060 at 15:02:13.427078:
>    ------------------------------------------------------------------------
>    SIP/2.0 400 Malformed From, fix your code!
>    Via: SIP/2.0/UDP 192.168.254.73;rport=5060;branch=z9hG4bKHr6ea76mt53QQ
>    To: <sip:1002 at 192.168.254.168 <mailto:sip%3A1002 at 192.168.254.168>>;tag=c8797354
>    From: <sip:1234 at 192.168.254.73 <mailto:sip%3A1234 at 192.168.254.73>>
>    Call-ID: a588ad14-58f2-1235-07b5-525400e95788
>    CSeq: 102081187 INVITE
>    Content-Length: 0
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rick at magicmail.mooo.com  Thu Jan 19 19:14:20 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 19 Jan 2017 16:14:20 +0000
Subject: [Freeswitch-users] 123456
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E862A390@mbx-01.sysconfig.co.uk>
References: <2BFFE67C-8097-406B-8D97-99639577B189@magicmail.mooo.com>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E862A390@mbx-01.sysconfig.co.uk>
Message-ID: <C94923E0-DFF8-4B6C-8028-AA641E4FFB60@magicmail.mooo.com>

I should have looked a bit harder, thanks :D All coming from one IP, so blocked him!

> On 19 Jan 2017, at 12:35, Shaun Stokes <shaun.stokes at itec-support.co.uk> wrote:
> 
> This is a new channel being opened to an endpoint (123456 at x.x.x.x <mailto:123456 at x.x.x.x>), the endpoint is on your internal SIP profile.
>  
> An example of this might be a call routing to extension 123456 at x.x.x.x <mailto:123456 at x.x.x.x> which is registered to your internal SIP profile.
>  
> You should work your way up through the logs, providing the information is being logged you should be able to identify why the call was routed to this extension.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis
> Sent: 19 January 2017 12:14
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] 123456
>  
> I see a lot of these hitting FS continuously - where would they be originating from? (The IP x.x.x.x is the local IP)
>  
> 2017-01-19 12:06:07.773935 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/123456 at x.x.x.x <mailto:sofia/internal/123456 at x.x.x.x> [6ed9ba7d-4357-460d-a0b7-ff94bf871e7e]
> 
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
> ______________________________________________________________________
> 	
> Shaun Stokes - Infrastructure Analyst <>
> T :? <>	01453 700713 <>
> E :? <>	 <>shaun.stokes at itec-support.co.uk <mailto:shaun.stokes at itec-support.co.uk>
> W :? <>	 <>www.itec-support.co.uk <http://www.itec-support.co.uk/>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF <> 
> Company No. 06908001 <>
> 
> CONFIDENTIALITY NOTICE <> 
> This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. <> 
> The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. <>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
> ______________________________________________________________________
> _________________________________________________________________________
> 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 rick at magicmail.mooo.com  Thu Jan 19 19:16:21 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 19 Jan 2017 16:16:21 +0000
Subject: [Freeswitch-users] Trunk for inbound connections
In-Reply-To: <1807D724-8BEF-473A-B96B-C172110FAA5F@magicmail.mooo.com>
References: <5F3ADC17-7A7E-486A-90DA-9676BB82C0D4@magicmail.mooo.com>
	<CAHtxdDeONRo-ODHXRKZjzRPLB19RsBiWKNQ+YF_Hf0ZxG5JZCw@mail.gmail.com>
	<1807D724-8BEF-473A-B96B-C172110FAA5F@magicmail.mooo.com>
Message-ID: <BE17A0AD-E4BD-4A96-9D88-0293EFC7F7BE@magicmail.mooo.com>

Anyone know what the missing link is here, getting an inbound route to point successfully at a trunk (which is authenticating as a user in the directory)? I?m sure I?ve done this before countless times, but having a bit of a senior moment I think...

> On 18 Jan 2017, at 18:25, Rick Jarvis <rick at magicmail.mooo.com> wrote:
> 
> If I just set up a user, I get a problem where ?gw+? is being added to the SIP header, so calls from me to the third party system fail? something that?s better described here: https://www.3cx.com/community/threads/configuring-freeswitch-as-a-voice-provider.19656/ <https://www.3cx.com/community/threads/configuring-freeswitch-as-a-voice-provider.19656/>
> 
> That?s why I started wondering if creating a user wasn?t the right way to go..?
> 
> 
>> On 18 Jan 2017, at 18:18, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> 
>> I use assignments ip address of remote system to FreeSwith user in directory.
>> 
>> 
>> ??, 18 ???. 2017, 19:41 Rick Jarvis <rick at magicmail.mooo.com <mailto:rick at magicmail.mooo.com>>:
>> What is the best and cleanest method for providing a trunk for someone else to register into, in the most transparent way (i.e. cross-compatible, for providing a ITSP service to third party systems)?
>> _________________________________________________________________________
>> 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
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krice at freeswitch.org  Thu Jan 19 20:13:27 2017
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 19 Jan 2017 11:13:27 -0600
Subject: [Freeswitch-users] Nested conditions and anti-actions
In-Reply-To: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>
References: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>
Message-ID: <06ce01d27277$59db88c0$0d929a40$@freeswitch.org>

This is buy design...

<condition><action/></anti-action></condition>

Is the if CONDITION then ACTION else ANTI-ACTION

And keep in mind the dialplan is not a programming language... once you
reach a certain point that's what things like lua xml_curl and just
replacing the dialplan with a custom C module are for


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko
Sent: Thursday, January 19, 2017 8:36 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Nested conditions and anti-actions

Hello.

This is a snipped:

<condition field="destination_number" expression="^(1001|1002)$"
require-nested="false">
  <condition field="${effective_caller_id_name}" expression="^.+$"
break="never">
   <action application="export"
data="caller_id_name=${effective_caller_id_name} > ${destination_number}"/>
   <action application="export"
data="effective_caller_id_name=${effective_caller_id_name} >
${destination_number}"/>
   <anti-action application="export" data="caller_id_name=${caller_id_name}
> ${destination_number}"/>
   <anti-action application="export"
data="effective_caller_id_name=${caller_id_name} > ${destination_number}"/>
  </condition>
<action .... />
</condition>

The problem I'm facing is that anti-actions are evaluated even if
destination_number is not equal to 1001 or 1002. But I need that ant-actions
will evaluate only when parent condition is true.

Is this a bug or it is by design?

--
Mimiko desu.

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

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

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



From tculjaga at gmail.com  Thu Jan 19 21:45:03 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 19 Jan 2017 19:45:03 +0100
Subject: [Freeswitch-users] Malformed From, fix your code!
In-Reply-To: <33CB0B18-9255-46F1-9309-EC050F08CC6C@jerris.com>
References: <CAJoKKdUNT2J5sj+RKnqgRMv9EacSE-U_nLvygk+ux494YzbY9g@mail.gmail.com>
	<33CB0B18-9255-46F1-9309-EC050F08CC6C@jerris.com>
Message-ID: <CAJoKKdXUhc8VSjLf1fuHESCYZ=XriQeACGPR6mtn1Uh+98ud-Q@mail.gmail.com>

crap!

shame on me:

# fgrep "192.168.254.73" * -R
sip_profiles/internal.xml:    <param name="rtp-ip" value="192.168.254.73 "/>
sip_profiles/internal.xml:    <param name="sip-ip" value="192.168.254.73 "/>

thanks!



On 19 January 2017 at 17:01, Michael Jerris <mike at jerris.com> wrote:

> grep your configs for that ip address.  Sounds like you have it somewhere
> with a space after it.
>
> On Jan 19, 2017, at 9:07 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> hi,
>
>
> does anyone experience the same issue ?
> im running FreeSWITCH Version 1.6.14+git~20170112T174802Z~a8d53fdd79~64bit
> (git a8d53fd 2017-01-12 17:48:02Z 64bit)
>
>
>
> im trying to originate a call like this:
>
> bgapi originate [origination_caller_id_number=1234]sofia/internal/
> 1002 at 192.168.254.168 &echo()
>
>
> it looks like FS brakes the from header ... pls note the extra space
> between the IP address and the closing ">"
>
>
>    -----------------------------------------------------------
> -------------
> send 1009 bytes to udp/[192.168.254.168]:5060 at 15:02:13.408606:
>    -----------------------------------------------------------
> -------------
>    INVITE sip:1002 at 192.168.254.168 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.254.73;rport;branch=z9hG4bKHr6ea76mt53QQ
>    Max-Forwards: 70
>    From: "" <sip:1234 at 192.168.254.73 >;tag=Q0SvF26gNteKa
>    To: <sip:1002 at 192.168.254.168>
>    Call-ID: a588ad14-58f2-1235-07b5-525400e95788
>    CSeq: 102081187 INVITE
>    Contact: <sip:mod_sofia at 192.168.254.73 :5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.6.14+git~20170112T174802Z~
> a8d53fdd79~64bit
>    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: 224
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: <sip:1234 at 192.168.254.73 >;party=calling;screen=yes;
> privacy=off
>
>    v=0
>    o=FreeSWITCH 1484806170 1484806171 IN IP4 192.168.254.73
>    s=FreeSWITCH
>    c=IN IP4 192.168.254.73
>    t=0 0
>    m=audio 28332 RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    -----------------------------------------------------------
> -------------
> recv 286 bytes from udp/[192.168.254.168]:5060 at 15:02:13.427078:
>    -----------------------------------------------------------
> -------------
>    SIP/2.0 400 Malformed From, fix your code!
>    Via: SIP/2.0/UDP 192.168.254.73;rport=5060;branch=z9hG4bKHr6ea76mt53QQ
>    To: <sip:1002 at 192.168.254.168>;tag=c8797354
>    From: <sip:1234 at 192.168.254.73>
>    Call-ID: a588ad14-58f2-1235-07b5-525400e95788
>    CSeq: 102081187 INVITE
>    Content-Length: 0
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raman.chv at gmail.com  Thu Jan 19 19:21:12 2017
From: raman.chv at gmail.com (Ram Anji)
Date: Thu, 19 Jan 2017 21:51:12 +0530
Subject: [Freeswitch-users] Fwd: NOTIFYs not received for conference
	SUBSCRIBE events
In-Reply-To: <CAArNKuc16hnERYcbH9-C7atw-zpry4s0zUkJdAfwhfaAAZ+=KQ@mail.gmail.com>
References: <CAArNKuc16hnERYcbH9-C7atw-zpry4s0zUkJdAfwhfaAAZ+=KQ@mail.gmail.com>
Message-ID: <CAArNKue7QrSUbg-9qV6DHNAq-9ceAAUL8Y=m4XGM95y5io6quQ@mail.gmail.com>

Hi,

I am testing conference on freeswitch with jitsi. Jitsi sending SUBSCRIBE
event after receiving 200 ok of call with focus parameter along with
contact header.
Server is responding 202 of subscription but no NOTIFY is observed for the
subscription.

I am testing with FreeSWITCH version: 1.6.14~64bit ( 64bit) where it is
installed from freeswitch rpms and os is centos.

Enclosed the console logs for the reference

Regards,
Raman
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freeswitch at localhost.localdomain> 2017-01-19 17:55:08.515299 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping

freeswitch at localhost.localdomain> 
freeswitch at localhost.localdomain> sofi

[               sofia]	[       sofia_contact]	[     sofia_count_reg]	[           sofia_dig]	
[  sofia_gateway_data]	[ sofia_presence_data]	[   sofia_username_of]	

freeswitch at localhost.localdomain> sofia glo

[              global]	

freeswitch at localhost.localdomain> sofia global 2017-01-19 17:55:38.755302 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping
siptra

[            siptrace]	

freeswitch at localhost.localdomain> sofia global siptrace o

[                  on]	[                 off]	

freeswitch at localhost.localdomain> sofia global siptrace on

[                  on]	

freeswitch at localhost.localdomain> sofia global siptrace on 

+OK Global siptrace on
freeswitch at localhost.localdomain> console loglev

[            loglevel]	

freeswitch at localhost.localdomain> console loglevel debug

+OK console log level set to DEBUG

freeswitch at localhost.localdomain> 2017-01-19 17:56:09.014392 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping
sofia tracelevel alert

+OK tracelevel is ALERT
freeswitch at localhost.localdomain> recv 1422 bytes from udp/[10.90.111.21]:5060 at 17:56:35.736340:
   ------------------------------------------------------------------------
   INVITE sip:30000 at 10.90.111.196 SIP/2.0
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 1 INVITE
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-33fecedd00708bbca83bad7b51034efa
   Max-Forwards: 70
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>
   User-Agent: Jitsi2.8.5426Windows 8
   Content-Type: application/sdp
   Content-Length: 935
   
   v=0
   o=1001-jitsi.org 0 0 IN IP4 10.90.111.21
   s=-
   c=IN IP4 10.90.111.21
   t=0 0
   m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
   a=rtpmap:96 opus/48000/2
   a=fmtp:96 usedtx=1
   a=rtpmap:97 SILK/24000
   a=rtpmap:98 SILK/16000
   a=rtpmap:9 G722/8000
   a=rtpmap:100 speex/32000
   a=rtpmap:102 speex/16000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:103 iLBC/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:104 speex/8000
   a=rtpmap:4 G723/8000
   a=fmtp:4 annexa=no;bitrate=6.3
   a=rtpmap:101 telephone-event/8000
   a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=rtcp-xr:voip-metrics
   m=video 5010 RTP/AVP 105 99
   a=recvonly
   a=rtpmap:105 H264/90000
   a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
   a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
   a=rtpmap:99 H264/90000
   a=fmtp:99 profile-level-id=4DE01f
   a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]
   ------------------------------------------------------------------------
send 337 bytes to udp/[10.90.111.21]:5060 at 17:56:35.736574:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-33fecedd00708bbca83bad7b51034efa
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:35.736067 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001 at 10.90.111.196 [84ba0012-de42-11e6-bb7c-b9b014fd381e]
2017-01-19 17:56:35.736067 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1001 at 10.90.111.196) Running State Change CS_NEW (Cur 1 Tot 4)
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:9815 sofia/internal/1001 at 10.90.111.196 receiving invite from 10.90.111.21:5060 version: 1.6.14  64bit
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:9982 IP 10.90.111.21 Rejected by acl "domains". Falling back to Digest auth.
send 841 bytes to udp/[10.90.111.21]:5060 at 17:56:35.748410:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-33fecedd00708bbca83bad7b51034efa
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=7UF5N931pQypp
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Accept: application/sdp
   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
   Proxy-Authenticate: Digest realm="10.90.111.196", nonce="84ba4cac-de42-11e6-bb7d-b9b014fd381e", algorithm=MD5, qop="auth"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:2333 detaching session 84ba0012-de42-11e6-bb7c-b9b014fd381e
2017-01-19 17:56:35.736067 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1001 at 10.90.111.196) State NEW
recv 340 bytes from udp/[10.90.111.21]:5060 at 17:56:35.750809:
   ------------------------------------------------------------------------
   ACK sip:30000 at 10.90.111.196 SIP/2.0
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   Max-Forwards: 70
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=7UF5N931pQypp
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-33fecedd00708bbca83bad7b51034efa
   CSeq: 1 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 1656 bytes from udp/[10.90.111.21]:5060 at 17:56:35.752487:
   ------------------------------------------------------------------------
   INVITE sip:30000 at 10.90.111.196 SIP/2.0
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 2 INVITE
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>
   Max-Forwards: 70
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>
   User-Agent: Jitsi2.8.5426Windows 8
   Content-Type: application/sdp
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-5fc9fe0a66995f7f4289e7fa174576d0
   Proxy-Authorization: Digest username="1001",realm="10.90.111.196",nonce="84ba4cac-de42-11e6-bb7d-b9b014fd381e",uri="sip:30000 at 10.90.111.196",response="7f0ed5ee03b2e50e108ec28320829290",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001
   Content-Length: 935
   
   v=0
   o=1001-jitsi.org 0 0 IN IP4 10.90.111.21
   s=-
   c=IN IP4 10.90.111.21
   t=0 0
   m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
   a=rtpmap:96 opus/48000/2
   a=fmtp:96 usedtx=1
   a=rtpmap:97 SILK/24000
   a=rtpmap:98 SILK/16000
   a=rtpmap:9 G722/8000
   a=rtpmap:100 speex/32000
   a=rtpmap:102 speex/16000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:103 iLBC/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:104 speex/8000
   a=rtpmap:4 G723/8000
   a=fmtp:4 annexa=no;bitrate=6.3
   a=rtpmap:101 telephone-event/8000
   a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
   a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=rtcp-xr:voip-metrics
   m=video 5010 RTP/AVP 105 99
   a=recvonly
   a=rtpmap:105 H264/90000
   a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
   a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
   a=rtpmap:99 H264/90000
   a=fmtp:99 profile-level-id=4DE01f
   a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]
   ------------------------------------------------------------------------
send 337 bytes to udp/[10.90.111.21]:5060 at 17:56:35.752865:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-5fc9fe0a66995f7f4289e7fa174576d0
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:2441 Re-attaching to session 84ba0012-de42-11e6-bb7c-b9b014fd381e
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:9815 sofia/internal/1001 at 10.90.111.196 receiving invite from 10.90.111.21:5060 version: 1.6.14  64bit
2017-01-19 17:56:35.736067 [DEBUG] sofia.c:9982 IP 10.90.111.21 Rejected by acl "domains". Falling back to Digest auth.
2017-01-19 17:56:35.774369 [DEBUG] sofia.c:7041 Channel sofia/internal/1001 at 10.90.111.196 entering state [received][100]
2017-01-19 17:56:35.774369 [DEBUG] sofia.c:7051 Remote SDP:
v=0
o=1001-jitsi.org 0 0 IN IP4 10.90.111.21
s=-
c=IN IP4 10.90.111.21
t=0 0
m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5010 RTP/AVP 105 99
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=recvonly
a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]

2017-01-19 17:56:35.774369 [DEBUG] sofia.c:7443 (sofia/internal/1001 at 10.90.111.196) State Change CS_NEW -> CS_INIT
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1001 at 10.90.111.196) Running State Change CS_INIT (Cur 1 Tot 4)
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/1001 at 10.90.111.196) State INIT
2017-01-19 17:56:35.774369 [DEBUG] mod_sofia.c:90 sofia/internal/1001 at 10.90.111.196 SOFIA INIT
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at 10.90.111.196 Standard INIT
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at 10.90.111.196) State Change CS_INIT -> CS_ROUTING
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/1001 at 10.90.111.196) State INIT going to sleep
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1001 at 10.90.111.196) Running State Change CS_ROUTING (Cur 1 Tot 4)
2017-01-19 17:56:35.774369 [DEBUG] switch_channel.c:2249 (sofia/internal/1001 at 10.90.111.196) Callstate Change DOWN -> RINGING
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/1001 at 10.90.111.196) State ROUTING
2017-01-19 17:56:35.774369 [DEBUG] mod_sofia.c:143 sofia/internal/1001 at 10.90.111.196 SOFIA ROUTING
2017-01-19 17:56:35.774369 [DEBUG] switch_core_state_machine.c:236 sofia/internal/1001 at 10.90.111.196 Standard ROUTING
2017-01-19 17:56:35.815142 [INFO] mod_dialplan_xml.c:637 Processing 1001 <1001>->30000 in context default
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1001 at 10.90.111.196 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 Action set(open=true) 
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1001 at 10.90.111.196 Date/TimeMatch (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [global-intercept] destination_number(30000) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [group-intercept] destination_number(30000) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [intercept-ext] destination_number(30000) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->redial] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [redial] destination_number(30000) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->global] continue=true
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (PASS) [global] ${default_password}(1234) =~ /^1234$/ break=never
Dialplan: sofia/internal/1001 at 10.90.111.196 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action log(CRIT Open /etc/freeswitch/vars.xml and change the default_password.) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action log(CRIT Once changed type 'reloadxml' at the console.) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING ) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action sleep(10000) 
Dialplan: sofia/internal/1001 at 10.90.111.196 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 10.90.111.196 Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [global] ${switch_r_sdp}(v=0
o=1001-jitsi.org 0 0 IN IP4 10.90.111.21
s=-
c=IN IP4 10.90.111.21
t=0 0
m=audio 5008 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 4 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5010 RTP/AVP 105 99
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=recvonly
a=imageattr:105 send * recv [x=[0-1366],y=[0-768]]
a=imageattr:99 send * recv [x=[0-1366],y=[0-768]]
) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
Dialplan: sofia/internal/1001 at 10.90.111.196 Absolute Condition [global]
Dialplan: sofia/internal/1001 at 10.90.111.196 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [snom-demo-2] destination_number(30000) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [snom-demo-1] destination_number(30000) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [eavesdrop] destination_number(30000) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [eavesdrop] destination_number(30000) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [call_return] destination_number(30000) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [del-group] destination_number(30000) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [add-group] destination_number(30000) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [call-group-simo] destination_number(30000) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [call-group-order] destination_number(30000) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [extension-intercom] destination_number(30000) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [Local_Extension] destination_number(30000) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [Local_Extension_Skinny] destination_number(30000) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->group_dial_sales] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [group_dial_sales] destination_number(30000) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->group_dial_support] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [group_dial_support] destination_number(30000) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->group_dial_billing] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [group_dial_billing] destination_number(30000) =~ /^2002$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->operator] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [operator] destination_number(30000) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->vmain] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [vmain] destination_number(30000) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->sip_uri] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (FAIL) [sip_uri] destination_number(30000) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 parsing [default->my_conferences] continue=false
Dialplan: sofia/internal/1001 at 10.90.111.196 Regex (PASS) [my_conferences] destination_number(30000) =~ /^(300\d{2})$/ break=on-false
Dialplan: sofia/internal/1001 at 10.90.111.196 Action answer(is_conference) 
Dialplan: sofia/internal/1001 at 10.90.111.196 Action conference(30000-${domain_name}@default) 
2017-01-19 17:56:35.815142 [DEBUG] switch_core_state_machine.c:286 (sofia/internal/1001 at 10.90.111.196) State Change CS_ROUTING -> CS_EXECUTE
2017-01-19 17:56:35.815142 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/1001 at 10.90.111.196) State ROUTING going to sleep
2017-01-19 17:56:35.815142 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1001 at 10.90.111.196) Running State Change CS_EXECUTE (Cur 1 Tot 4)
2017-01-19 17:56:35.815142 [DEBUG] switch_core_state_machine.c:650 (sofia/internal/1001 at 10.90.111.196) State EXECUTE
2017-01-19 17:56:35.815142 [DEBUG] mod_sofia.c:198 sofia/internal/1001 at 10.90.111.196 SOFIA EXECUTE
2017-01-19 17:56:35.815142 [DEBUG] switch_core_state_machine.c:328 sofia/internal/1001 at 10.90.111.196 Standard EXECUTE
2017-01-19 17:56:35.875099 [WARNING] sofia_presence.c:1312 external-ipv6 is passive, skipping
2017-01-19 17:56:35.875099 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='external' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'external' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (external)
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:35]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:35 GMT]
Event-Date-Timestamp: [1484828795815142]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2452]
Channel-State: [CS_ROUTING]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [ringing]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [alerting]
presence-call-direction: [inbound]
event_count: [0]
Presence-Calling-File: [src/switch_core_state_machine.c]
Presence-Calling-Function: [check_presence]
Presence-Calling-Line: [524]


2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (external)
2017-01-19 17:56:35.875099 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal-ipv6' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal-ipv6' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:35]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:35 GMT]
Event-Date-Timestamp: [1484828795815142]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2452]
Channel-State: [CS_ROUTING]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [ringing]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [alerting]
presence-call-direction: [inbound]
event_count: [0]
Presence-Calling-File: [src/switch_core_state_machine.c]
Presence-Calling-Function: [check_presence]
Presence-Calling-Line: [524]


2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal-ipv6)
EXECUTE sofia/internal/1001 at 10.90.111.196 set(open=true)
2017-01-19 17:56:35.875099 [DEBUG] mod_dptools.c:1527 SET sofia/internal/1001 at 10.90.111.196 [open]=[true]
EXECUTE sofia/internal/1001 at 10.90.111.196 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING )
2017-01-19 17:56:35.875099 [CRIT] mod_dptools.c:1721 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 
EXECUTE sofia/internal/1001 at 10.90.111.196 log(CRIT Open /etc/freeswitch/vars.xml and change the default_password.)
2017-01-19 17:56:35.875099 [CRIT] mod_dptools.c:1721 Open /etc/freeswitch/vars.xml and change the default_password.
EXECUTE sofia/internal/1001 at 10.90.111.196 log(CRIT Once changed type 'reloadxml' at the console.)
2017-01-19 17:56:35.875099 [CRIT] mod_dptools.c:1721 Once changed type 'reloadxml' at the console.
EXECUTE sofia/internal/1001 at 10.90.111.196 log(CRIT WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING )
2017-01-19 17:56:35.875099 [CRIT] mod_dptools.c:1721 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 
EXECUTE sofia/internal/1001 at 10.90.111.196 sleep(10000)
2017-01-19 17:56:35.875099 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal)
2017-01-19 17:56:35.875099 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:35]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:35 GMT]
Event-Date-Timestamp: [1484828795815142]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2452]
Channel-State: [CS_ROUTING]
Channel-Call-State: [RINGING]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [ringing]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [alerting]
presence-call-direction: [inbound]
event_count: [0]
Presence-Calling-File: [src/switch_core_state_machine.c]
Presence-Calling-Function: [check_presence]
Presence-Calling-Line: [524]


2017-01-19 17:56:35.875099 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal)
2017-01-19 17:56:39.134390 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping
EXECUTE sofia/internal/1001 at 10.90.111.196 hash(insert/10.90.111.196-spymap/1001/84ba0012-de42-11e6-bb7c-b9b014fd381e)
EXECUTE sofia/internal/1001 at 10.90.111.196 hash(insert/10.90.111.196-last_dial/1001/30000)
EXECUTE sofia/internal/1001 at 10.90.111.196 hash(insert/10.90.111.196-last_dial/global/84ba0012-de42-11e6-bb7c-b9b014fd381e)
EXECUTE sofia/internal/1001 at 10.90.111.196 export(RFC2822_DATE=Thu, 19 Jan 2017 17:56:46 +0530)
2017-01-19 17:56:46.114792 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 19 Jan 2017 17:56:46 +0530]
EXECUTE sofia/internal/1001 at 10.90.111.196 answer(is_conference)
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [opus:96:48000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [opus:96:48000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:97:24000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:97:24000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:97:24000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:97:24000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:98:16000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:98:16000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:98:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [SILK:98:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:100:32000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:100:32000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:100:32000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:100:32000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:102:16000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:102:16000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:102:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:102:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4433 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [iLBC:103:8000:30:13330:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [iLBC:103:8000:30:13330:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [iLBC:103:8000:30:13330:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [iLBC:103:8000:30:13330:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:104:8000:20:0:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:104:8000:20:0:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:104:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [speex:104:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G723:4:8000:30:6300:1]/[opus:116:48000:20:0:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G723:4:8000:30:6300:1]/[G722:9:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G723:4:8000:30:6300:1]/[PCMU:0:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4378 Audio Codec Compare [G723:4:8000:30:6300:1]/[PCMA:8:8000:20:64000:1]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4294 Set telephone-event payload to 101 at 8000
2017-01-19 17:56:46.134405 [DEBUG] mod_opus.c:586 Opus encoder: set bitrate to local settings [72000bps]
2017-01-19 17:56:46.134405 [DEBUG] mod_opus.c:586 Opus encoder: set bitrate to local settings [72000bps]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:3043 Set Codec sofia/internal/1001 at 10.90.111.196 opus/48000 20 ms 960 samples 0 bits 1 channels
2017-01-19 17:56:46.134405 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at 10.90.111.196 Original read codec set to opus:116
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4696 sofia/internal/1001 at 10.90.111.196 Set 2833 dtmf send payload to 101 recv payload to 101
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4877 Video Codec Compare [H264:105]/[VP8:99]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:4877 Video Codec Compare [H264:99]/[VP8:99]
2017-01-19 17:56:46.134405 [DEBUG] switch_core_media.c:6703 AUDIO RTP [sofia/internal/1001 at 10.90.111.196] 10.90.111.196 port 24596 -> 10.90.111.21 port 5008 codec: 96 ms: 20
2017-01-19 17:56:46.134405 [DEBUG] switch_rtp.c:3878 Starting timer [soft] 960 bytes per 20ms
2017-01-19 17:56:46.154382 [DEBUG] switch_core_media.c:7009 sofia/internal/1001 at 10.90.111.196 Set 2833 dtmf send payload to 101
2017-01-19 17:56:46.154382 [DEBUG] switch_core_media.c:7016 sofia/internal/1001 at 10.90.111.196 Set 2833 dtmf receive payload to 101
2017-01-19 17:56:46.154382 [DEBUG] switch_core_media.c:7039 sofia/internal/1001 at 10.90.111.196 Set rtp dtmf delay to 40
2017-01-19 17:56:46.154382 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001 at 10.90.111.196!
2017-01-19 17:56:46.154382 [DEBUG] switch_channel.c:3473 (sofia/internal/1001 at 10.90.111.196) Callstate Change RINGING -> EARLY
2017-01-19 17:56:46.154382 [DEBUG] switch_core_media.c:6686 Audio params are unchanged for sofia/internal/1001 at 10.90.111.196.
2017-01-19 17:56:46.154382 [DEBUG] mod_sofia.c:849 Local SDP sofia/internal/1001 at 10.90.111.196:
v=0
o=FreeSWITCH 1484804210 1484804211 IN IP4 10.90.111.196
s=FreeSWITCH
c=IN IP4 10.90.111.196
t=0 0
m=audio 24596 RTP/AVP 96 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 19

send 1180 bytes to udp/[10.90.111.21]:5060 at 17:56:46.156308:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-5fc9fe0a66995f7f4289e7fa174576d0
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=848XQ4m5K0m9H
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 2 INVITE
   Contact: <sip:30000 at 10.90.111.196:5060;transport=udp>;isfocus
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Accept: application/sdp
   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-Type: application/sdp
   Content-Disposition: session
   Content-Length: 275
   Remote-Party-ID: "30000" <sip:30000 at 10.90.111.196>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1484804210 1484804211 IN IP4 10.90.111.196
   s=FreeSWITCH
   c=IN IP4 10.90.111.196
   t=0 0
   m=audio 24596 RTP/AVP 96 101
   a=rtpmap:96 opus/48000/2
   a=fmtp:96 useinbandfec=1
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   m=video 0 RTP/AVP 19
   ------------------------------------------------------------------------
2017-01-19 17:56:46.154382 [DEBUG] sofia.c:7041 Channel sofia/internal/1001 at 10.90.111.196 entering state [completed][200]
2017-01-19 17:56:46.154382 [NOTICE] mod_dptools.c:1309 Channel [sofia/internal/1001 at 10.90.111.196] has been answered
2017-01-19 17:56:46.154382 [WARNING] sofia_presence.c:1312 external-ipv6 is passive, skipping
2017-01-19 17:56:46.154382 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='external' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'external' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (external)
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806154382]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2481]
Channel-State: [CS_EXECUTE]
Channel-Call-State: [EARLY]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [answered]
Channel-Read-Codec-Name: [opus]
Channel-Read-Codec-Rate: [48000]
Channel-Read-Codec-Bit-Rate: [0]
Channel-Write-Codec-Name: [opus]
Channel-Write-Codec-Rate: [48000]
Channel-Write-Codec-Bit-Rate: [0]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [1484828806154382]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [1484828806154382]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [answered]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-direction: [inbound]
event_count: [1]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_mark_answered]
Presence-Calling-Line: [3766]


2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (external)
2017-01-19 17:56:46.154382 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal-ipv6' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal-ipv6' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806154382]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2481]
Channel-State: [CS_EXECUTE]
Channel-Call-State: [EARLY]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [answered]
Channel-Read-Codec-Name: [opus]
Channel-Read-Codec-Rate: [48000]
Channel-Read-Codec-Bit-Rate: [0]
Channel-Write-Codec-Name: [opus]
Channel-Write-Codec-Rate: [48000]
Channel-Write-Codec-Bit-Rate: [0]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [1484828806154382]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [1484828806154382]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [answered]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-direction: [inbound]
event_count: [1]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_mark_answered]
Presence-Calling-Line: [3766]


2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.154382 [DEBUG] switch_channel.c:3772 (sofia/internal/1001 at 10.90.111.196) Callstate Change EARLY -> ACTIVE
2017-01-19 17:56:46.154382 [CRIT] sofia_presence.c:1378 CHECK SQL: 1001 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '84ba0012-de42-11e6-bb7c-b9b014fd381e' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal' and ((sip_from_user='1001' and sip_from_host='10.90.111.196') or presence_id='1001 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal' or presence_hosts like '%10.90.111.196%')
recv 485 bytes from udp/[10.90.111.21]:5060 at 17:56:46.162672:
   ------------------------------------------------------------------------
   ACK sip:30000 at 10.90.111.196:5060;transport=udp SIP/2.0
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 2 ACK
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-da3b30bd24bc02315db96108b7b13aa5
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=848XQ4m5K0m9H
   Max-Forwards: 70
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>
   User-Agent: Jitsi2.8.5426Windows 8
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal)
2017-01-19 17:56:46.154382 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='1001' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806154382]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [785]
Event-Sequence: [2481]
Channel-State: [CS_EXECUTE]
Channel-Call-State: [EARLY]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-HIT-Dialplan: [true]
Channel-Presence-ID: [1001 at 10.90.111.196]
Channel-Call-UUID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Answer-State: [answered]
Channel-Read-Codec-Name: [opus]
Channel-Read-Codec-Rate: [48000]
Channel-Read-Codec-Bit-Rate: [0]
Channel-Write-Codec-Name: [opus]
Channel-Write-Codec-Rate: [48000]
Channel-Write-Codec-Bit-Rate: [0]
Caller-Direction: [inbound]
Caller-Logical-Direction: [inbound]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001]
Caller-Caller-ID-Number: [1001]
Caller-Orig-Caller-ID-Name: [1001]
Caller-Orig-Caller-ID-Number: [1001]
Caller-Network-Addr: [10.90.111.21]
Caller-ANI: [1001]
Caller-Destination-Number: [30000]
Caller-Unique-ID: [84ba0012-de42-11e6-bb7c-b9b014fd381e]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/1001 at 10.90.111.196]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1484828795774369]
Caller-Channel-Created-Time: [1484828795774369]
Caller-Channel-Answered-Time: [1484828806154382]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [1484828806154382]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Channel-Resurrect-Time: [0]
Caller-Channel-Bridged-Time: [0]
Caller-Channel-Last-Hold: [0]
Caller-Channel-Hold-Accum: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [any]
login: [src/switch_channel.c]
from: [1001 at 10.90.111.196]
rpid: [unknown]
status: [answered]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-direction: [inbound]
event_count: [1]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_mark_answered]
Presence-Calling-Line: [3766]


2017-01-19 17:56:46.154382 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal)
EXECUTE sofia/internal/1001 at 10.90.111.196 conference(30000-10.90.111.196 at default)
2017-01-19 17:56:46.154382 [DEBUG] sofia.c:7041 Channel sofia/internal/1001 at 10.90.111.196 entering state [ready][200]
2017-01-19 17:56:46.154382 [DEBUG] sofia.c:7089 QUERY SQL update sip_dialogs set sip_to_tag='848XQ4m5K0m9H' where uuid='84ba0012-de42-11e6-bb7c-b9b014fd381e' and sip_to_tag = ''
2017-01-19 17:56:46.199781 [DEBUG] mod_conference.c:3077 using channel sound prefix: /usr/share/freeswitch/sounds/en/us/callie
2017-01-19 17:56:46.199781 [DEBUG] conference_member.c:1679 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms
2017-01-19 17:56:46.199781 [DEBUG] conference_member.c:1726 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2017-01-19 17:56:46.199781 [DEBUG] switch_core_codec.c:223 sofia/internal/1001 at 10.90.111.196 Push codec L16:100
2017-01-19 17:56:46.199781 [WARNING] sofia_presence.c:1312 external-ipv6 is passive, skipping
2017-01-19 17:56:46.199781 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000-10.90.111.196 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000-10.90.111.196' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='external' and ((sip_from_user='30000-10.90.111.196' and sip_from_host='10.90.111.196') or presence_id='30000-10.90.111.196 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'external' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (external)
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806199781]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_send_presence]
Event-Calling-Line-Number: [3349]
Event-Sequence: [2488]
proto: [conf]
login: [30000-10.90.111.196]
from: [30000-10.90.111.196 at 10.90.111.196]
event_type: [presence]
alt_event_type: [dialog]
event_count: [12]
unique-id: [30000-10.90.111.196]
force-status: [Active (1 caller)]
channel-state: [CS_ROUTING]
answer-state: [early]
presence-call-direction: [outbound]


2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (external)
2017-01-19 17:56:46.199781 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000-10.90.111.196 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000-10.90.111.196' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal-ipv6' and ((sip_from_user='30000-10.90.111.196' and sip_from_host='10.90.111.196') or presence_id='30000-10.90.111.196 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal-ipv6' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806199781]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_send_presence]
Event-Calling-Line-Number: [3349]
Event-Sequence: [2488]
proto: [conf]
login: [30000-10.90.111.196]
from: [30000-10.90.111.196 at 10.90.111.196]
event_type: [presence]
alt_event_type: [dialog]
event_count: [12]
unique-id: [30000-10.90.111.196]
force-status: [Active (1 caller)]
channel-state: [CS_ROUTING]
answer-state: [early]
presence-call-direction: [outbound]


2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.199781 [ERR] switch_core_video.c:1870 This function is not available, libpng not installed
2017-01-19 17:56:46.199781 [DEBUG] mod_conference.c:220 Setup timer success interval: 20  samples: 160
2017-01-19 17:56:46.199781 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000-10.90.111.196 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000-10.90.111.196' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal' and ((sip_from_user='30000-10.90.111.196' and sip_from_host='10.90.111.196') or presence_id='30000-10.90.111.196 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal)
2017-01-19 17:56:46.199781 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000-10.90.111.196' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806199781]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_send_presence]
Event-Calling-Line-Number: [3349]
Event-Sequence: [2488]
proto: [conf]
login: [30000-10.90.111.196]
from: [30000-10.90.111.196 at 10.90.111.196]
event_type: [presence]
alt_event_type: [dialog]
event_count: [12]
unique-id: [30000-10.90.111.196]
force-status: [Active (1 caller)]
channel-state: [CS_ROUTING]
answer-state: [early]
presence-call-direction: [outbound]


2017-01-19 17:56:46.199781 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal)
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '0' to 'mute'
2017-01-19 17:56:46.234609 [INFO] switch_ivr_async.c:214 Digit parser mod_conference: Setting realm to 'conf'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 0/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071898
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '*' to 'deaf mute'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding */conf/0 callback: 0x7fccf3451770 data: 0x7fccf40718c8
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '9' to 'energy up'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 9/conf/0 callback: 0x7fccf3451770 data: 0x7fccf40718f8
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '8' to 'energy equ'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 8/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071928
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '7' to 'energy dn'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 7/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071958
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '3' to 'vol talk up'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 3/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071988
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '2' to 'vol talk zero'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 2/conf/0 callback: 0x7fccf3451770 data: 0x7fccf40719b8
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '1' to 'vol talk dn'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 1/conf/0 callback: 0x7fccf3451770 data: 0x7fccf40719e8
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '6' to 'vol listen up'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 6/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071a18
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '5' to 'vol listen zero'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 5/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071a48
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '4' to 'vol listen dn'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding 4/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071a78
2017-01-19 17:56:46.234609 [DEBUG] conference_member.c:128 sofia/internal/1001 at 10.90.111.196 binding '#' to 'hangup'
2017-01-19 17:56:46.234609 [DEBUG] switch_ivr_async.c:323 Digit parser mod_conference: binding #/conf/0 callback: 0x7fccf3451770 data: 0x7fccf4071aa8
2017-01-19 17:56:46.234609 [DEBUG] conference_loop.c:1140 Setup timer soft success interval: 20  samples: 960 from codec opus
2017-01-19 17:56:46.234609 [NOTICE] switch_core_io.c:1202 Activating write resampler
recv 806 bytes from udp/[10.90.111.21]:5060 at 17:56:46.257889:
   ------------------------------------------------------------------------
   SUBSCRIBE sip:30000 at 10.90.111.196:5060;transport=udp SIP/2.0
   CSeq: 3 SUBSCRIBE
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=848XQ4m5K0m9H
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-bde2d1b2d9287374c9da0da52a2d9ce9
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>
   Proxy-Authorization: Digest username="1001",realm="10.90.111.196",nonce="84ba4cac-de42-11e6-bb7d-b9b014fd381e",uri="sip:30000 at 10.90.111.196",response="7f0ed5ee03b2e50e108ec28320829290",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001
   User-Agent: Jitsi2.8.5426Windows 8
   Max-Forwards: 70
   Event: conference
   Accept: application/conference-info+xml
   Expires: 3600
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:46.254727 [ERR] sofia_presence.c:3774 DELTA 3600
2017-01-19 17:56:46.254727 [ERR] sofia_presence.c:3896 check subs sql: select contact from sip_subscriptions where call_id='64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0' and profile_name='internal' and hostname='localhost.localdomain' []
2017-01-19 17:56:46.254727 [NOTICE] sofia_presence.c:3976 internal SUBSCRIBE 1001 at 10.90.111.196 30000 at 10.90.111.196
insert into sip_subscriptions (proto,sip_user,sip_host,sub_to_user,sub_to_host,presence_hosts,event,contact,call_id,full_from,full_via,expires,user_agent,accept,profile_name,hostname,network_port,network_ip,version,orig_proto, full_to) values ('conf','1001','10.90.111.196','30000','10.90.111.196','10.90.111.196,10.90.111.196','conference','"user" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>','64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0','"1001" <sip:1001 at 10.90.111.196>;tag=4f65891a','SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-bde2d1b2d9287374c9da0da52a2d9ce9',1484832406,'Jitsi2.8.5426Windows 8','application/conference-info+xml ','internal','localhost.localdomain','5060','10.90.111.21',-1,'sip','<sip:30000 at 10.90.111.196>;tag=848XQ4m5K0m9H;tag=848XQ4m5K0m9H')
2017-01-19 17:56:46.254727 [DEBUG] sofia_presence.c:4089 Responding to SUBSCRIBE with 202 Accepted
send 772 bytes to udp/[10.90.111.21]:5060 at 17:56:46.271684:
   ------------------------------------------------------------------------
   SIP/2.0 202 Accepted
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-bde2d1b2d9287374c9da0da52a2d9ce9
   From: "1001" <sip:1001 at 10.90.111.196>;tag=4f65891a
   To: <sip:30000 at 10.90.111.196>;tag=848XQ4m5K0m9H
   Call-ID: 64517400ba1d554c830f45be85e95607 at 0:0:0:0:0:0:0:0
   CSeq: 3 SUBSCRIBE
   Contact: <sip:30000 at 10.90.111.196:5060>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   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
   Subscription-State: active;expires=3600
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:56:46.274433 [WARNING] sofia_presence.c:1312 external-ipv6 is passive, skipping
2017-01-19 17:56:46.274433 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='external' and ((sip_from_user='30000' and sip_from_host='10.90.111.196') or presence_id='30000 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'external' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.274433 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (external)
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='external' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806274433]
Event-Calling-File: [conference_event.c]
Event-Calling-Function: [conference_event_pres_handler]
Event-Calling-Line-Number: [808]
Event-Sequence: [2496]
proto: [conf]
login: [30000]
from: [conf+30000 at 10.90.111.196]
force-status: [Idle]
rpid: [unknown]
event_type: [presence]
alt_event_type: [dialog]
event_count: [13]
unique-id: [30000]
channel-state: [CS_HANGUP]
answer-state: [terminated]
call-direction: [inbound]


2017-01-19 17:56:46.274433 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (external)
2017-01-19 17:56:46.274433 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal-ipv6' and ((sip_from_user='30000' and sip_from_host='10.90.111.196') or presence_id='30000 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal-ipv6' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.274433 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal-ipv6' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='::1' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806274433]
Event-Calling-File: [conference_event.c]
Event-Calling-Function: [conference_event_pres_handler]
Event-Calling-Line-Number: [808]
Event-Sequence: [2496]
proto: [conf]
login: [30000]
from: [conf+30000 at 10.90.111.196]
force-status: [Idle]
rpid: [unknown]
event_type: [presence]
alt_event_type: [dialog]
event_count: [13]
unique-id: [30000]
channel-state: [CS_HANGUP]
answer-state: [terminated]
call-direction: [inbound]


2017-01-19 17:56:46.274433 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal-ipv6)
2017-01-19 17:56:46.274433 [CRIT] sofia_presence.c:1378 CHECK SQL: 30000 at 10.90.111.196 [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '30000' and call_info_state != 'seized' and hostname='localhost.localdomain' and profile_name='internal' and ((sip_from_user='30000' and sip_from_host='10.90.111.196') or presence_id='30000 at 10.90.111.196') order by rcd desc]
hits: 0
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1435 PRES SQL update sip_subscriptions set version=version+1 where hostname='localhost.localdomain' and profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') and (sip_subscriptions.profile_name = 'internal' or presence_hosts like '%10.90.111.196%')
2017-01-19 17:56:46.274433 [INFO] sofia_presence.c:1510 IN START_PRESENCE_SQL (internal)
2017-01-19 17:56:46.274433 [ERR] sofia_presence.c:1519 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','unknown','10.90.111.196',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='localhost.localdomain' and sip_subscriptions.profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='conf' and (event='presence' or event='dialog') and sub_to_user='30000' and (sub_to_host='10.90.111.196' or sub_to_host='10.90.111.196' or sub_to_host='N/A' or presence_hosts like '%10.90.111.196%') 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [a161672c-de2a-11e6-bb22-b9b014fd381e]
FreeSWITCH-Hostname: [localhost.localdomain]
FreeSWITCH-Switchname: [localhost.localdomain]
FreeSWITCH-IPv4: [10.90.111.196]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2017-01-19 17:56:46]
Event-Date-GMT: [Thu, 19 Jan 2017 12:26:46 GMT]
Event-Date-Timestamp: [1484828806274433]
Event-Calling-File: [conference_event.c]
Event-Calling-Function: [conference_event_pres_handler]
Event-Calling-Line-Number: [808]
Event-Sequence: [2496]
proto: [conf]
login: [30000]
from: [conf+30000 at 10.90.111.196]
force-status: [Idle]
rpid: [unknown]
event_type: [presence]
alt_event_type: [dialog]
event_count: [13]
unique-id: [30000]
channel-state: [CS_HANGUP]
answer-state: [terminated]
call-direction: [inbound]


2017-01-19 17:56:46.294419 [INFO] sofia_presence.c:1530 IN END_PRESENCE_SQL (internal)
2017-01-19 17:56:46.316874 [DEBUG] switch_rtp.c:6994 Correct audio ip/port confirmed.
2017-01-19 17:56:46.395284 [DEBUG] conference_member.c:1679 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms
2017-01-19 17:56:46.395284 [DEBUG] conference_member.c:1726 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2017-01-19 17:56:46.395284 [DEBUG] conference_loop.c:1140 Setup timer soft success interval: 20  samples: 960 from codec opus
2017-01-19 17:56:49.014470 [DEBUG] mod_local_stream.c:866 Opening Stream [moh/8000] 8000hz
2017-01-19 17:57:09.234360 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping
2017-01-19 17:57:39.474664 [WARNING] sofia_presence.c:5077 check_subs: external-ipv6 is passive, skipping
recv 707 bytes from udp/[10.90.111.21]:5060 at 17:57:57.095739:
   ------------------------------------------------------------------------
   REGISTER sip:10.90.111.196 SIP/2.0
   Call-ID: 0ebcb3a3c821b82ce4ed280dc5fd85db at 0:0:0:0:0:0:0:0
   CSeq: 39 REGISTER
   From: "1001" <sip:1001 at 10.90.111.196>;tag=bb27c622
   To: "1001" <sip:1001 at 10.90.111.196>
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-acfa729113d6cfd3a6d9c2141d86f299
   Max-Forwards: 70
   Authorization: Digest username="1001",realm="10.90.111.196",nonce="73541386-de41-11e6-bb7a-b9b014fd381e",uri="sip:10.90.111.196",response="3ba4f3489b81392712e88f1241ea6ff0",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001
   User-Agent: Jitsi2.8.5426Windows 8
   Expires: 600
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>;expires=600
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 647 bytes to udp/[10.90.111.21]:5060 at 17:57:57.102238:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-acfa729113d6cfd3a6d9c2141d86f299
   From: "1001" <sip:1001 at 10.90.111.196>;tag=bb27c622
   To: "1001" <sip:1001 at 10.90.111.196>;tag=9D2pSZ58g9avD
   Call-ID: 0ebcb3a3c821b82ce4ed280dc5fd85db at 0:0:0:0:0:0:0:0
   CSeq: 39 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   WWW-Authenticate: Digest realm="10.90.111.196", nonce="b538df24-de42-11e6-bb8d-b9b014fd381e", stale=true, algorithm=MD5, qop="auth"
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 707 bytes from udp/[10.90.111.21]:5060 at 17:57:57.106665:
   ------------------------------------------------------------------------
   REGISTER sip:10.90.111.196 SIP/2.0
   Call-ID: 0ebcb3a3c821b82ce4ed280dc5fd85db at 0:0:0:0:0:0:0:0
   CSeq: 40 REGISTER
   From: "1001" <sip:1001 at 10.90.111.196>;tag=bb27c622
   To: "1001" <sip:1001 at 10.90.111.196>
   Max-Forwards: 70
   User-Agent: Jitsi2.8.5426Windows 8
   Expires: 600
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>;expires=600
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-a2b494f58e33dbfa497a4637f8ca0f41
   Authorization: Digest username="1001",realm="10.90.111.196",nonce="b538df24-de42-11e6-bb8d-b9b014fd381e",uri="sip:10.90.111.196",response="d2402414eb8f01f120508b6d37f175c7",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001
   Content-Length: 0
   
   ------------------------------------------------------------------------
2017-01-19 17:57:57.114497 [ERR] sofia_reg.c:1988 DELETE PRESENCE SQL: delete from sip_presence where sip_user='1001' and sip_host='10.90.111.196' and profile_name='internal' and open_closed='closed'
send 636 bytes to udp/[10.90.111.21]:5060 at 17:57:57.117936:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.90.111.21:5060;branch=z9hG4bK-313938-a2b494f58e33dbfa497a4637f8ca0f41
   From: "1001" <sip:1001 at 10.90.111.196>;tag=bb27c622
   To: "1001" <sip:1001 at 10.90.111.196>;tag=aQUFUtpcej1eS
   Call-ID: 0ebcb3a3c821b82ce4ed280dc5fd85db at 0:0:0:0:0:0:0:0
   CSeq: 40 REGISTER
   Contact: <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>;expires=600
   Date: Thu, 19 Jan 2017 12:27:57 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 940 bytes to udp/[10.90.111.21]:5060 at 17:57:57.173115:
   ------------------------------------------------------------------------
   NOTIFY sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196 SIP/2.0
   Via: SIP/2.0/UDP 10.90.111.196;rport;branch=z9hG4bKm6aFa4663908m
   Max-Forwards: 70
   From: <sip:1001 at 10.90.111.196>;tag=B0m8vN7FBUQ1m
   To: <sip:1001 at 10.90.111.196>
   Call-ID: 8ca26d1b-58e5-1235-14ad-08002772c459
   CSeq: 102078374 NOTIFY
   Contact: <sip:mod_sofia at 10.90.111.196:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.6.14~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Event: message-summary
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Subscription-State: terminated;reason=noresource
   Content-Type: application/simple-message-summary
   Content-Length: 65
   
   Messages-Waiting: no
   Message-Account: sip:1001 at 10.90.111.196
   
   ------------------------------------------------------------------------
recv 421 bytes from udp/[10.90.111.21]:5060 at 17:57:57.181076:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   CSeq: 102078374 NOTIFY
   Call-ID: 8ca26d1b-58e5-1235-14ad-08002772c459
   From: <sip:1001 at 10.90.111.196>;tag=B0m8vN7FBUQ1m
   To: <sip:1001 at 10.90.111.196>;tag=7f9fea93
   Via: SIP/2.0/UDP 10.90.111.196;rport=5060;branch=z9hG4bKm6aFa4663908m;received=10.90.111.196
   Contact: "1001" <sip:1001 at 10.90.111.21:5060;transport=udp;registering_acc=10_90_111_196>
   User-Agent: Jitsi2.8.5426Windows 8
   Content-Length: 0
   
   ------------------------------------------------------------------------


From calvin.walton at kepstin.ca  Thu Jan 19 23:44:51 2017
From: calvin.walton at kepstin.ca (Calvin Walton)
Date: Thu, 19 Jan 2017 15:44:51 -0500
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
Message-ID: <1484858691.18111.5.camel@kepstin.ca>

Hi Ken,

I dunno if this exactly fits what you're looking for, but I think it
shows off that FreeSWITCH can be used as part of a system or product
that might not at first glance appear to be "telephony" related.

The BigBlueButton (http://bigbluebutton.org) project is an open-source
web conferencing application designed primarily for the distance
education market, but also great for team meetings, hybrid classrooms,
interviews, etc. It's controlled via API, and integrations are
available with many popular learning management systems.

We use FreeSWITCH as a core part of the product - it handles the voice
conference bridge portion of the application, including the ability to
dial in to a meeting via PSTN and - in recent versions - WebRTC audio
support through the web browser.

We highlight the open-source projects we build on here: http://bigblueb
utton.org/components/

(Hmm, that components page is actually looking a little out of date, we
might have to do some cleanup there.)

Calvin.

On Tue, 2017-01-17 at 09:44 -0600, Ken Rice wrote:
> Hey Guys,
> 
> ?
> 
> A lot of you are using FreeSWITCH for some fairly complex things.
> However
> that is not always clear to people discovering FreeSWITCH for the
> first
> time.
> 
> ?
> 
> I would like to build a list of examples. If you don't mind this info
> being
> compiled on Confluence.
> 
> Please??Reply with who, what, and where so we can document it and
> show off
> some of the really cool FreeSWITCH based or FreeSWITCH using
> deployments out
> there.
> 
> ?
> 
> Thanks
> 
> Ken
> 
> _____________________________________________________________________
> ____
> Professional FreeSWITCH Consulting Services:?
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> ers
> http://www.freeswitch.org
-- 
Calvin Walton <calvin.walton at kepstin.ca>


From samir.doshi at inextrix.com  Fri Jan 20 08:57:49 2017
From: samir.doshi at inextrix.com (Samir Doshi)
Date: Fri, 20 Jan 2017 11:27:49 +0530
Subject: [Freeswitch-users] How are you using FreeSWITCH?
In-Reply-To: <1484858691.18111.5.camel@kepstin.ca>
References: <142101d270d8$8d0f1b30$a72d5190$@freeswitch.org>
	<1484858691.18111.5.camel@kepstin.ca>
Message-ID: <CAFABgc0vH76RJRDbHqv5XG0MH7Mmev6SQDRoihpgHhK9A+0Giw@mail.gmail.com>

*Who : *
ASTPP (http://astppbilling.org)

*What : *
ASTPP - Open Source VoIP Billing Solution for FreeSwitch.

ASTPP Features :
- Accounts management (Customer, Multilevel Reseller, Provider, Admin,
Subadmin)
- Calling cards
- Accounting
- Provider Billing (Interconnect)
- Tariff & Rates management
- Call Routing
- DIDs Management
- Various Reports
- Paypal Payment Gateway
- Mass Mailling
- Mobile/PC Dialers
-  ... and much more

*Where : *
USA,Canada, South Africa, France, Spain, UK, India, Brazil and many more
countries.

We also using FreeSwitch for different telephony projects like IVR
development, fax process, pbx, webrtc etc @iNextrix
<http://www.inextrix.com>.

Best Regards
--
Samir Doshi
*iNextrix Technologie**s Pvt. Ltd*.
http://www.inextrix.com



*Disclaimer:*
The information contained in this communication is confidential and may be
legally privileged. It is intended solely for the use of the individual or
entity to whom it is addressed and others authorised to receive it. If you
are not the intended recipient you are hereby notified that any disclosure,
copying, distribution or taking action in reliance of the contents of this
information is strictly prohibited and may be unlawful.

On Fri, Jan 20, 2017 at 2:14 AM, Calvin Walton <calvin.walton at kepstin.ca>
wrote:

> Hi Ken,
>
> I dunno if this exactly fits what you're looking for, but I think it
> shows off that FreeSWITCH can be used as part of a system or product
> that might not at first glance appear to be "telephony" related.
>
> The BigBlueButton (http://bigbluebutton.org) project is an open-source
> web conferencing application designed primarily for the distance
> education market, but also great for team meetings, hybrid classrooms,
> interviews, etc. It's controlled via API, and integrations are
> available with many popular learning management systems.
>
> We use FreeSWITCH as a core part of the product - it handles the voice
> conference bridge portion of the application, including the ability to
> dial in to a meeting via PSTN and - in recent versions - WebRTC audio
> support through the web browser.
>
> We highlight the open-source projects we build on here: http://bigblueb
> utton.org/components/
>
> (Hmm, that components page is actually looking a little out of date, we
> might have to do some cleanup there.)
>
> Calvin.
>
> On Tue, 2017-01-17 at 09:44 -0600, Ken Rice wrote:
> > Hey Guys,
> >
> >
> >
> > A lot of you are using FreeSWITCH for some fairly complex things.
> > However
> > that is not always clear to people discovering FreeSWITCH for the
> > first
> > time.
> >
> >
> >
> > I would like to build a list of examples. If you don't mind this info
> > being
> > compiled on Confluence.
> >
> > Please  Reply with who, what, and where so we can document it and
> > show off
> > some of the really cool FreeSWITCH based or FreeSWITCH using
> > deployments out
> > there.
> >
> >
> >
> > Thanks
> >
> > Ken
> >
> > _____________________________________________________________________
> > ____
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> > ers
> > http://www.freeswitch.org
> --
> Calvin Walton <calvin.walton at kepstin.ca>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vbvbrj at gmail.com  Fri Jan 20 09:36:55 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 20 Jan 2017 08:36:55 +0200
Subject: [Freeswitch-users] Nested conditions and anti-actions
In-Reply-To: <06ce01d27277$59db88c0$0d929a40$@freeswitch.org>
References: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>
	<06ce01d27277$59db88c0$0d929a40$@freeswitch.org>
Message-ID: <cd3b7451-0002-0e47-051a-aacc7220e855@gmail.com>

Yes, I know this, Ken. The problem is with nested condition. Anti-actions in nested conditions are executed if parent condition fail? Also, I've 
observed that even if parent condition fail, actions or anti-actions in nested condition is executed. Ie, nested condition is checked and if fails - 
anti-action is executed, if it pass - actions is executed, regardless if parent condition fail or pass.

I expected that any actions/anti-actions in nested condition will be executed only if parent condition pass. Yes, wiki states that parent condition is 
not checked until nested-conditions are not checked. But, if at the end parent condition fails, whole extension should not be executed.

For now I changed the conditions in a simple AND without nested conditions.


On 19.01.2017 19:13, Ken Rice wrote:
> This is buy design...
>
> <condition><action/></anti-action></condition>
>
> Is the if CONDITION then ACTION else ANTI-ACTION
>
> And keep in mind the dialplan is not a programming language... once you
> reach a certain point that's what things like lua xml_curl and just
> replacing the dialplan with a custom C module are for
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mimiko
> Sent: Thursday, January 19, 2017 8:36 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Nested conditions and anti-actions
>
> Hello.
>
> This is a snipped:
>
> <condition field="destination_number" expression="^(1001|1002)$"
> require-nested="false">
>   <condition field="${effective_caller_id_name}" expression="^.+$"
> break="never">
>    <action application="export"
> data="caller_id_name=${effective_caller_id_name} > ${destination_number}"/>
>    <action application="export"
> data="effective_caller_id_name=${effective_caller_id_name} >
> ${destination_number}"/>
>    <anti-action application="export" data="caller_id_name=${caller_id_name}
>> ${destination_number}"/>
>    <anti-action application="export"
> data="effective_caller_id_name=${caller_id_name} > ${destination_number}"/>
>   </condition>
> <action .... />
> </condition>
>
> The problem I'm facing is that anti-actions are evaluated even if
> destination_number is not equal to 1001 or 1002. But I need that ant-actions
> will evaluate only when parent condition is true.
>
> Is this a bug or it is by design?
>
> --
> Mimiko desu.
>


-- 
Mimiko desu.


From bipin at xbipin.com  Fri Jan 20 11:24:30 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 20 Jan 2017 12:24:30 +0400
Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
In-Reply-To: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>
References: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>
Message-ID: <05088bf2-3a45-1d6c-9575-f51069f9d65f@xbipin.com>

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From vbvbrj at gmail.com  Fri Jan 20 11:41:35 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 20 Jan 2017 10:41:35 +0200
Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
In-Reply-To: <05088bf2-3a45-1d6c-9575-f51069f9d65f@xbipin.com>
References: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>
	<05088bf2-3a45-1d6c-9575-f51069f9d65f@xbipin.com>
Message-ID: <13c93f05-fdf0-13d4-9357-cc41bb74ddad@gmail.com>

How bleg_caller_id_name be present in a bleg?

Use sip_from_uri.

On 20.01.2017 10:24, Bipin Patel wrote:
>
> any help, what im trying to do is enable cdr for b leg only and then in cdr template print the source caller id name/number of a as well as b leg both
> so we can know what changes happened through FS. if i enable cdr for ab then i get 2 entries which im trying to avoid
>
>
> Regards,
> Bipin
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------------
> -------- Original Message --------
> Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
> From: Bipin Patel <bipin at xbipin.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: 1/19/2017, 7:13:50 PM
>> hi,
>>
>> im using cdr_csv and suppose if i log cdr for a leg then bleg_caller_id_name and caller_id_name are written to cdr but if i log cdr for b leg then
>> bleg_caller_id_name doesnt populate, only caller_id_name does, why is this so?
>>
>> when i put cdr_csv in debug mode i see both variables getting populated in cli
>>


-- 
Mimiko desu.


From tculjaga at gmail.com  Fri Jan 20 11:55:33 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 20 Jan 2017 09:55:33 +0100
Subject: [Freeswitch-users] Nested conditions and anti-actions
In-Reply-To: <cd3b7451-0002-0e47-051a-aacc7220e855@gmail.com>
References: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>
	<06ce01d27277$59db88c0$0d929a40$@freeswitch.org>
	<cd3b7451-0002-0e47-051a-aacc7220e855@gmail.com>
Message-ID: <CAJoKKdWNEz2c06D7iYhNe4ot-wrTpiPYjGOjxBKOZXYsC8RF-A@mail.gmail.com>

here is an example.

extension agent_state is called with execute_extension like that:
<action application="execute_extension" data="agent_state XML features"/>



    <extension name="agent_state">
      <condition field="destination_number" expression="^agent_state$"/>
      <condition field="${agent_state}" expression="^(IDLE|EMPTY)$">

        <action application="set" data="origination_cancel_key=#"/>
        <action application="export"
data="execute_on_answer_1=spandsp_start_dtmf"/>
        <action application="export"
data="execute_on_answer_2=bind_meta_app 4 a s execute_extension::att_xfer
XML features"/>
        <action application="set"
data="transfer_ringback=%(2000,4000,425,0)"/>
        <action application="execute_extension"
data="populate_xfer_dial_string XML features"/>
        <action application="att_xfer" data="${xfer_dial_string}"/>

        <anti-action application="set" data="transfer-digits=NULL"/>
        <anti-action application="eval" data="cancel transfer"/>
        <anti-action application="playback"
data="tone_stream://%(500,500,425,0);loops=3"/>
        <anti-action application="set" data="api_result=${uuid_hold off
${uuid}}"/>

      </condition>
    </extension>


so, if destination_number is "agent_state" and if ${agent_state} variable
is "IDLE" or "EMPTY" , perform actions else perform anti-actions




hope it helps.










On 20 January 2017 at 07:36, Mimiko <vbvbrj at gmail.com> wrote:

> Yes, I know this, Ken. The problem is with nested condition. Anti-actions
> in nested conditions are executed if parent condition fail? Also, I've
> observed that even if parent condition fail, actions or anti-actions in
> nested condition is executed. Ie, nested condition is checked and if fails -
> anti-action is executed, if it pass - actions is executed, regardless if
> parent condition fail or pass.
>
> I expected that any actions/anti-actions in nested condition will be
> executed only if parent condition pass. Yes, wiki states that parent
> condition is
> not checked until nested-conditions are not checked. But, if at the end
> parent condition fails, whole extension should not be executed.
>
> For now I changed the conditions in a simple AND without nested conditions.
>
>
> On 19.01.2017 19:13, Ken Rice wrote:
> > This is buy design...
> >
> > <condition><action/></anti-action></condition>
> >
> > Is the if CONDITION then ACTION else ANTI-ACTION
> >
> > And keep in mind the dialplan is not a programming language... once you
> > reach a certain point that's what things like lua xml_curl and just
> > replacing the dialplan with a custom C module are for
> >
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Mimiko
> > Sent: Thursday, January 19, 2017 8:36 AM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: [Freeswitch-users] Nested conditions and anti-actions
> >
> > Hello.
> >
> > This is a snipped:
> >
> > <condition field="destination_number" expression="^(1001|1002)$"
> > require-nested="false">
> >   <condition field="${effective_caller_id_name}" expression="^.+$"
> > break="never">
> >    <action application="export"
> > data="caller_id_name=${effective_caller_id_name} >
> ${destination_number}"/>
> >    <action application="export"
> > data="effective_caller_id_name=${effective_caller_id_name} >
> > ${destination_number}"/>
> >    <anti-action application="export" data="caller_id_name=${caller_
> id_name}
> >> ${destination_number}"/>
> >    <anti-action application="export"
> > data="effective_caller_id_name=${caller_id_name} >
> ${destination_number}"/>
> >   </condition>
> > <action .... />
> > </condition>
> >
> > The problem I'm facing is that anti-actions are evaluated even if
> > destination_number is not equal to 1001 or 1002. But I need that
> ant-actions
> > will evaluate only when parent condition is true.
> >
> > Is this a bug or it is by design?
> >
> > --
> > Mimiko desu.
> >
>
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Fri Jan 20 12:06:00 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 20 Jan 2017 13:06:00 +0400
Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
In-Reply-To: <13c93f05-fdf0-13d4-9357-cc41bb74ddad@gmail.com>
References: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>
	<05088bf2-3a45-1d6c-9575-f51069f9d65f@xbipin.com>
	<13c93f05-fdf0-13d4-9357-cc41bb74ddad@gmail.com>
Message-ID: <011b263a-cdc2-b8d1-452a-eaa99d263856@xbipin.com>

hi,

with the debug enabled i saw the bleg_caller_id_name variable showing 
the a leg data so i though i can get that but it doesnt print in CDR so 
then why does it show in the debug?

Regards,
Bipin



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

-------- Original Message  --------
Subject: Re: [Freeswitch-users] CDR not populating when b leg logging 
enabled
From: Mimiko <vbvbrj at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Date: 1/20/2017, 12:41:35 PM

> How bleg_caller_id_name be present in a bleg?
>
> Use sip_from_uri.
>
> On 20.01.2017 10:24, Bipin Patel wrote:
>>
>> any help, what im trying to do is enable cdr for b leg only and then in cdr template print the source caller id name/number of a as well as b leg both
>> so we can know what changes happened through FS. if i enable cdr for ab then i get 2 entries which im trying to avoid
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------------
>> -------- Original Message --------
>> Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
>> From: Bipin Patel <bipin at xbipin.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: 1/19/2017, 7:13:50 PM
>>> hi,
>>>
>>> im using cdr_csv and suppose if i log cdr for a leg then bleg_caller_id_name and caller_id_name are written to cdr but if i log cdr for b leg then
>>> bleg_caller_id_name doesnt populate, only caller_id_name does, why is this so?
>>>
>>> when i put cdr_csv in debug mode i see both variables getting populated in cli
>>>
>
>



From vbvbrj at gmail.com  Fri Jan 20 12:34:13 2017
From: vbvbrj at gmail.com (Mimiko)
Date: Fri, 20 Jan 2017 11:34:13 +0200
Subject: [Freeswitch-users] Nested conditions and anti-actions
In-Reply-To: <CAJoKKdWNEz2c06D7iYhNe4ot-wrTpiPYjGOjxBKOZXYsC8RF-A@mail.gmail.com>
References: <5bed47b7-5cdc-f760-3811-87d6b2ffbae4@gmail.com>
	<06ce01d27277$59db88c0$0d929a40$@freeswitch.org>
	<cd3b7451-0002-0e47-051a-aacc7220e855@gmail.com>
	<CAJoKKdWNEz2c06D7iYhNe4ot-wrTpiPYjGOjxBKOZXYsC8RF-A@mail.gmail.com>
Message-ID: <41334968-7c01-e4b7-772c-28671fb992ed@gmail.com>

Yes, thanks.

I wrote I changed the extension to simple AND, like you pointed out.

On 20.01.2017 10:55, Tihomir Culjaga wrote:
> so, if destination_number is "agent_state" and if ${agent_state} variable is "IDLE" or "EMPTY" , perform actions else perform anti-actions
>
>
>
>
> hope it helps.


-- 
Mimiko desu.


From bipin at xbipin.com  Fri Jan 20 13:52:54 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 20 Jan 2017 14:52:54 +0400
Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
In-Reply-To: <011b263a-cdc2-b8d1-452a-eaa99d263856@xbipin.com>
References: <b6447a80-cee4-2629-403f-d332e666625a@xbipin.com>
	<05088bf2-3a45-1d6c-9575-f51069f9d65f@xbipin.com>
	<13c93f05-fdf0-13d4-9357-cc41bb74ddad@gmail.com>
	<011b263a-cdc2-b8d1-452a-eaa99d263856@xbipin.com>
Message-ID: <5a9b285c-f82c-8ed5-2af4-f9dd24c49b88@xbipin.com>

hi,

ok i figured this out, if i log cdr for b leg only then to get some data 
of the a leg i need to use aleg_caller_id_name and then it populates it 
fine.

one other issue now im facing is suppose if the call failed with cause 
incompatible_destination which is due to codec mismatch then in CDR it 
logs both legs even though i told it to log b leg only, any way to get 
around this so FS obeys whats set in cdr_csv.conf.xml only?

Regards,
Bipin


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

-------- Original Message  --------
Subject: Re: [Freeswitch-users] CDR not populating when b leg logging 
enabled
From: Bipin Patel <bipin at xbipin.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 1/20/2017, 1:06:00 PM

> hi,
>
> with the debug enabled i saw the bleg_caller_id_name variable showing
> the a leg data so i though i can get that but it doesnt print in CDR so
> then why does it show in the debug?
>
> Regards,
> Bipin
>
>
>
> ------------------------------------------------------------------------
>
> -------- Original Message  --------
> Subject: Re: [Freeswitch-users] CDR not populating when b leg logging
> enabled
> From: Mimiko <vbvbrj at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: 1/20/2017, 12:41:35 PM
>
>> How bleg_caller_id_name be present in a bleg?
>>
>> Use sip_from_uri.
>>
>> On 20.01.2017 10:24, Bipin Patel wrote:
>>>
>>> any help, what im trying to do is enable cdr for b leg only and then in cdr template print the source caller id name/number of a as well as b leg both
>>> so we can know what changes happened through FS. if i enable cdr for ab then i get 2 entries which im trying to avoid
>>>
>>>
>>> Regards,
>>> Bipin
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------------
>>> -------- Original Message --------
>>> Subject: [Freeswitch-users] CDR not populating when b leg logging enabled
>>> From: Bipin Patel <bipin at xbipin.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Date: 1/19/2017, 7:13:50 PM
>>>> hi,
>>>>
>>>> im using cdr_csv and suppose if i log cdr for a leg then bleg_caller_id_name and caller_id_name are written to cdr but if i log cdr for b leg then
>>>> bleg_caller_id_name doesnt populate, only caller_id_name does, why is this so?
>>>>
>>>> when i put cdr_csv in debug mode i see both variables getting populated in cli
>>>>
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Fri Jan 20 18:32:42 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 20 Jan 2017 09:32:42 -0600
Subject: [Freeswitch-users] Fwd: NOTIFYs not received for conference
 SUBSCRIBE events
In-Reply-To: <CAArNKue7QrSUbg-9qV6DHNAq-9ceAAUL8Y=m4XGM95y5io6quQ@mail.gmail.com>
References: <CAArNKuc16hnERYcbH9-C7atw-zpry4s0zUkJdAfwhfaAAZ+=KQ@mail.gmail.com>
	<CAArNKue7QrSUbg-9qV6DHNAq-9ceAAUL8Y=m4XGM95y5io6quQ@mail.gmail.com>
Message-ID: <CAEJMVkBO30Gt=5-_yMB-1jexfxp47tFO5LvyQhS=LuU5Cn+01A@mail.gmail.com>

Do you have conference CDR's enabled and have you set the flag to enable
this?

On Thu, Jan 19, 2017 at 10:21 AM, Ram Anji <raman.chv at gmail.com> wrote:

> Hi,
>
> I am testing conference on freeswitch with jitsi. Jitsi sending SUBSCRIBE
> event after receiving 200 ok of call with focus parameter along with
> contact header.
> Server is responding 202 of subscription but no NOTIFY is observed for the
> subscription.
>
> I am testing with FreeSWITCH version: 1.6.14~64bit ( 64bit) where it is
> installed from freeswitch rpms and os is centos.
>
> Enclosed the console logs for the reference
>
> Regards,
> Raman
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From mail at paulzillmann.de  Fri Jan 20 20:42:06 2017
From: mail at paulzillmann.de (Paul Zillmann)
Date: Fri, 20 Jan 2017 18:42:06 +0100
Subject: [Freeswitch-users] leg-B cuts off connection while transfer
Message-ID: <2c70a7a9-eafe-5b8a-2129-64c24dc6b4ab@paulzillmann.de>

Hello Folks,

a user hooked me on, that whenever he calls certain numbers, and was put
into hold (call is transfered), that legB sends a BYE after all.
Is this a misconfigure on the user's side - or is legB misconfigured?

The siptrace attatched seems to be okay - but what I don't understand
is, why is the second reinvite marked as recvonly.
I guess the second reinvited comes from legB because of the transfer was
accepted - but why reconly - and why does legB sends a BYE after that one?

Thanks a lot

   ------------------------------------------------------------------------
recv 1075 bytes from udp/[10.0.0.2]:5060 at 16:45:47.854134:
   ------------------------------------------------------------------------
   INVITE sip:022029****@172.16.10.11 SIP/2.0
   Via: SIP/2.0/UDP
10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport
   From: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   To: <sip:022029****@172.16.10.11>
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 21 INVITE
   Contact: <sip:204 at 10.0.0.2:5060>
   Content-Type: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS,
REFER, UPDATE, PRACK
   Max-Forwards: 70
   P-Early-Media: supported
   User-Agent: SIPPER for PhonerLite
   Session-Expires: 1800
   Supported: 100rel, replaces, from-change, timer
   P-Preferred-Identity: <sip:204 at 172.16.10.11>
   Content-Length:   406

   v=0
   o=- 2104407805 1 IN IP4 10.0.0.2
   s=SIPPER for PhonerLite
   c=IN IP4 10.0.0.2
   t=0 0
   m=audio 5062 RTP/AVP 9 8 0 2 3 97 110 111 101
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:97 iLBC/8000
   a=rtpmap:110 speex/8000
   a=rtpmap:111 speex/16000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ssrc:1205340143
   a=sendrecv
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.0.0.2]:5060 at 16:45:47.854518:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060
   From: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   To: <sip:022029****@172.16.10.11>
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 21 INVITE
   User-Agent: H6G Callserver
   Content-Length: 0

   ------------------------------------------------------------------------
send 1417 bytes to udp/[217.0.23.100]:5060 at 16:45:47.911364:
   ------------------------------------------------------------------------
   INVITE sip:022029****@tel.t-online.de SIP/2.0
   Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK782XNNX3Fm2Ur
   Route: <sip:217.0.23.100:5060;transport=udp;lr>
   Max-Forwards: 69
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   To: <sip:022029****@tel.t-online.de>
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127509 INVITE
   Contact: <sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P>
   User-Agent: H6G Callserver
   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
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 360
   P-Early-Media: supported
   X-FS-Support: update_display,send_info
   P-Asserted-Identity: "PhonerLite" <sip:0220224****@tel.t-online.de>
   P-Preferred-Identity: <sip:0220224****@tel.t-online.de>

   v=0
   o=FreeSWITCH 1484904625 1484904626 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 22522 RTP/AVP 8 0 110 111 101 102
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:110 SPEEX/8000
   a=rtpmap:111 SPEEX/16000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:102 telephone-event/16000
   a=fmtp:102 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 528 bytes from udp/[217.0.23.100]:5060 at 16:45:48.072425:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required 02035034C
   Via: SIP/2.0/UDP
172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK782XNNX3Fm2Ur
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_97483a920cc4ce8660c6c8f7126c6e9d
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127509 INVITE
   Content-Length: 0
   Proxy-Authenticate: Digest
nonce="CFFDD4D3B7308258000000003356C90F",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true

   ------------------------------------------------------------------------
send 426 bytes to udp/[217.0.23.100]:5060 at 16:45:48.072612:
   ------------------------------------------------------------------------
   ACK sip:022029****@tel.t-online.de SIP/2.0
   Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK782XNNX3Fm2Ur
   Route: <sip:217.0.23.100:5060;transport=udp;lr>
   Max-Forwards: 69
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_97483a920cc4ce8660c6c8f7126c6e9d
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127509 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
send 1702 bytes to udp/[217.0.23.100]:5060 at 16:45:48.073041:
   ------------------------------------------------------------------------
   INVITE sip:022029****@tel.t-online.de SIP/2.0
   Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK8HvpQge7cXrem
   Route: <sip:217.0.23.100:5060;transport=udp;lr>
   Max-Forwards: 69
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   To: <sip:022029****@tel.t-online.de>
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 INVITE
   Contact: <sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P>
   User-Agent: H6G Callserver
   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
   Proxy-Authorization: Digest username="jbvp-scharf at t-online.de",
realm="tel.t-online.de", nonce="CFFDD4D3B7308258000000003356C90F",
cnonce="Wqe3FFnKEjWDEkCNXBvrUw", algorithm=MD5,
uri="sip:022029****@tel.t-online.de",
response="f9665e5312227392a11b97914e11593d", qop=auth, nc=00000001
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 360
   P-Early-Media: supported
   X-FS-Support: update_display,send_info
   P-Asserted-Identity: "PhonerLite" <sip:0220224****@tel.t-online.de>
   P-Preferred-Identity: <sip:0220224****@tel.t-online.de>

   v=0
   o=FreeSWITCH 1484904625 1484904626 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 22522 RTP/AVP 8 0 110 111 101 102
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:110 SPEEX/8000
   a=rtpmap:111 SPEEX/16000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:102 telephone-event/16000
   a=fmtp:102 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 320 bytes from udp/[217.0.23.100]:5060 at 16:45:48.304747:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem
   To: <sip:022029****@tel.t-online.de>
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 INVITE
   Content-Length: 0

   ------------------------------------------------------------------------
recv 876 bytes from udp/[217.0.23.100]:5060 at 16:45:49.434461:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 INVITE
   Contact: <sip:sgc_c at 217.0.23.100;transport=udp>
   Record-Route: <sip:217.0.23.100;transport=udp;lr>
   P-Early-Media: sendonly
   Supported: timer
   Content-Type: application/sdp
   Content-Length: 225
   Allow: NOTIFY, UPDATE, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER

   v=0
   o=- 1144933594 2336235026 IN IP4 217.0.23.100
   s=Basic Session
   c=IN IP4 217.0.4.167
   t=0 0
   m=audio 10224 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=rtpmap:8 PCMA/8000
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 1129 bytes to udp/[10.0.0.2]:5060 at 16:45:49.457465:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060
   From: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 21 INVITE
   Contact: <sip:022029****@172.16.10.11:5060;transport=udp>
   User-Agent: H6G Callserver
   Accept: application/sdp
   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-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   P-Early-Media: sendonly
   P-Asserted-Identity: "[P] 022029****" <sip:022029****@172.16.10.11>

   v=0
   o=FreeSWITCH 1484896495 1484896496 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 30654 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 867 bytes from udp/[217.0.23.100]:5060 at 16:45:49.838384:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP
172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 INVITE
   Contact: <sip:sgc_c at 217.0.23.100;transport=udp>
   Record-Route: <sip:217.0.23.100;transport=udp;lr>
   P-Early-Media: sendonly
   Supported: timer
   Content-Type: application/sdp
   Content-Length: 225
   Allow: NOTIFY, UPDATE, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER

   v=0
   o=- 1144933594 2336235026 IN IP4 217.0.23.100
   s=Basic Session
   c=IN IP4 217.0.4.167
   t=0 0
   m=audio 10224 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=rtpmap:8 PCMA/8000
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
recv 1132 bytes from udp/[217.0.23.100]:5060 at 16:46:02.806318:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
172.16.10.11:5080;received=91.13.238.143;rport=5080;branch=z9hG4bK8HvpQge7cXrem
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 INVITE
   Contact:
<sip:sgc_c at 217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
   Record-Route: <sip:217.0.23.100;transport=udp;lr>
   Session-Expires: 1800;refresher=uas
   Supported: timer
   Content-Type: application/sdp
   Content-Length: 225
   Session-ID: cf2f265e180b6a914ef96ffa72c934f5
   Authentication-Info:
qop=auth,rspauth="91a1f6e74a09e15ea0682500b4dc281f",cnonce="Wqe3FFnKEjWDEkCNXBvrUw",nc=00000001
   Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH,
INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

   v=0
   o=- 1144933594 2336235026 IN IP4 217.0.23.100
   s=Basic Session
   c=IN IP4 217.0.4.167
   t=0 0
   m=audio 10224 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=rtpmap:8 PCMA/8000
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 794 bytes to udp/[217.0.23.100]:5060 at 16:46:02.808824:
   ------------------------------------------------------------------------
   ACK sip:sgc_c at 217.0.23.100;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 172.16.10.11:5080;rport;branch=z9hG4bK9tNFSBZaa6e1F
   Route: <sip:217.0.23.100;transport=udp;lr>
   Max-Forwards: 70
   From: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   To:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127510 ACK
   Contact: <sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P>
   Proxy-Authorization: Digest username="jbvp-scharf at t-online.de",
realm="tel.t-online.de", nonce="CFFDD4D3B7308258000000003356C90F",
cnonce="Wqe3FFnKEjWDEkCNXBvrUw", algorithm=MD5,
uri="sip:022029****@tel.t-online.de",
response="f9665e5312227392a11b97914e11593d", qop=auth, nc=00000001
   Content-Length: 0

   ------------------------------------------------------------------------
send 1118 bytes to udp/[10.0.0.2]:5060 at 16:46:02.826970:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
10.0.0.2:5060;branch=z9hG4bK00de6a2e95dde611a221b1b68e4749c1;rport=5060
   From: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 21 INVITE
   Contact: <sip:022029****@172.16.10.11:5060;transport=udp>
   User-Agent: H6G Callserver
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   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
   Session-Expires: 1800;refresher=uac
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   P-Asserted-Identity: "[P] 022029****" <sip:022029****@172.16.10.11>

   v=0
   o=FreeSWITCH 1484896495 1484896496 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 30654 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 399 bytes from udp/[10.0.0.2]:5060 at 16:46:02.859203:
   ------------------------------------------------------------------------
   ACK sip:022029****@172.16.10.11:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP
10.0.0.2:5060;branch=z9hG4bK80af5b3795dde611a221b1b68e4749c1;rport
   From: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   To: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 21 ACK
   Contact: <sip:204 at 10.0.0.2:5060>
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1011 bytes from udp/[217.0.23.100]:5060 at 16:46:36.717056:
   ------------------------------------------------------------------------
   INVITE sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P SIP/2.0
   Max-Forwards: 66
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127511 INVITE
   Contact:
<sip:sgc_c at 217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
   Min-Se: 900
   Session-Expires: 1800;refresher=uac
   Supported: timer
   Supported: 100rel
   Content-Type: application/sdp
   Content-Length: 237
   Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH,
INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

   v=0
   o=- 1144933594 2336235027 IN IP4 217.0.23.100
   s=Basic Session
   c=IN IP4 217.0.4.167
   t=0 0
   a=sendonly
   m=audio 10224 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=rtpmap:8 PCMA/8000
   a=ptime:20
   a=sendonly
   ------------------------------------------------------------------------
send 404 bytes to udp/[217.0.23.100]:5060 at 16:46:36.717365:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127511 INVITE
   User-Agent: H6G Callserver
   Content-Length: 0

   ------------------------------------------------------------------------
send 975 bytes to udp/[217.0.23.100]:5060 at 16:46:36.969711:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7iyntgqns9719p5933wldmbjr9y
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127511 INVITE
   Contact: <sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P>
   User-Agent: H6G Callserver
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   Supported: timer, path, replaces
   Session-Expires: 1800;refresher=uac
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 232

   v=0
   o=FreeSWITCH 1484904625 1484904627 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 22522 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=inactive
   a=ptime:20
   ------------------------------------------------------------------------
recv 540 bytes from udp/[217.0.23.100]:5060 at 16:46:37.038798:
   ------------------------------------------------------------------------
   ACK sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P SIP/2.0
   Max-Forwards: 66
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ij54ucc23me1yvy9vdmn4y96yh
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127511 ACK
   Contact:
<sip:sgc_c at 217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1011 bytes from udp/[217.0.23.100]:5060 at 16:46:49.474526:
   ------------------------------------------------------------------------
   INVITE sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P SIP/2.0
   Max-Forwards: 66
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127512 INVITE
   Contact:
<sip:sgc_c at 217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
   Min-Se: 900
   Session-Expires: 1800;refresher=uac
   Supported: timer
   Supported: 100rel
   Content-Type: application/sdp
   Content-Length: 237
   Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH,
INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

   v=0
   o=- 1144933594 2336235028 IN IP4 217.0.23.100
   s=Basic Session
   c=IN IP4 217.0.4.167
   t=0 0
   a=recvonly
   m=audio 10224 RTP/AVP 8 101
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=rtpmap:8 PCMA/8000
   a=ptime:20
   a=recvonly
   ------------------------------------------------------------------------
send 404 bytes to udp/[217.0.23.100]:5060 at 16:46:49.474824:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127512 INVITE
   User-Agent: H6G Callserver
   Content-Length: 0

   ------------------------------------------------------------------------
send 963 bytes to udp/[217.0.23.100]:5060 at 16:46:49.767507:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7imeyt2ir9sutpj4jtyixqf4yq3
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127512 INVITE
   Contact: <sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P>
   User-Agent: H6G Callserver
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Require: timer
   Supported: timer, path, replaces
   Session-Expires: 1800;refresher=uac
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220

   v=0
   o=FreeSWITCH 1484904625 1484904628 IN IP4 172.16.10.11
   s=FreeSWITCH
   c=IN IP4 172.16.10.11
   t=0 0
   m=audio 22522 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 540 bytes from udp/[217.0.23.100]:5060 at 16:46:49.837652:
   ------------------------------------------------------------------------
   ACK sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P SIP/2.0
   Max-Forwards: 66
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ishs6rygvv7pvgg3qpcfprg24g
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127512 ACK
   Contact:
<sip:sgc_c at 217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 577 bytes from udp/[217.0.23.100]:5060 at 16:47:03.694027:
   ------------------------------------------------------------------------
   BYE sip:gw+P at 172.16.10.11:5080;transport=udp;gw=P SIP/2.0
   Max-Forwards: 66
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i241fgoep8n7wg8m5jfzdz6q59
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127513 BYE
   Reason: Q.850;cause=16;eri-location=0
   Content-Length: 0
   Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH,
INFO, INVITE, ACK, OPTIONS, CANCEL, BYE

   ------------------------------------------------------------------------
send 541 bytes to udp/[217.0.23.100]:5060 at 16:47:03.706495:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i241fgoep8n7wg8m5jfzdz6q59
   From:
<sip:022029****@tel.t-online.de>;tag=h7g4Esbg_p65566t1484927148m25502c952768733s1_2334977008-822422466
   To: "PhonerLite" <sip:0220224****@tel.t-online.de>;tag=5e8p22t558t9N
   Call-ID: 5a8f1143-59ca-1235-1283-408d5c1beb53
   CSeq: 102127513 BYE
   User-Agent: H6G Callserver
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
send 538 bytes to udp/[10.0.0.2]:5060 at 16:47:03.709373:
   ------------------------------------------------------------------------
   BYE sip:204 at 10.0.0.2:5060 SIP/2.0
   Via: SIP/2.0/UDP 172.16.10.11;rport;branch=z9hG4bK7rN6peF2S5pUB
   Max-Forwards: 70
   From: <sip:022029****@172.16.10.11>;tag=8XZN8mgHrg9aB
   To: "PhonerLite" <sip:204 at 172.16.10.11>;tag=2258694084
   Call-ID: 00DE6A2E-95DD-E611-A220-B1B68E4749C1 at 10.0.0.2
   CSeq: 102127547 BYE
   User-Agent: H6G Callserver
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;eri-location=0
   Content-Length: 0



From mike at jerris.com  Fri Jan 20 20:58:30 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 20 Jan 2017 12:58:30 -0500
Subject: [Freeswitch-users] leg-B cuts off connection while transfer
In-Reply-To: <2c70a7a9-eafe-5b8a-2129-64c24dc6b4ab@paulzillmann.de>
References: <2c70a7a9-eafe-5b8a-2129-64c24dc6b4ab@paulzillmann.de>
Message-ID: <595AF1E4-BA71-4ADB-B0C1-B7B7470A2A4F@jerris.com>

You would have to ask whatever is sending you the BYE.  Sometimes misconfigured things can hang up when on hold due to media timeout, sometimes other issues.  From this information its impossible to know.


> On Jan 20, 2017, at 12:42 PM, Paul Zillmann <mail at paulzillmann.de> wrote:
> 
> Hello Folks,
> 
> a user hooked me on, that whenever he calls certain numbers, and was put
> into hold (call is transfered), that legB sends a BYE after all.
> Is this a misconfigure on the user's side - or is legB misconfigured?
> 
> The siptrace attatched seems to be okay - but what I don't understand
> is, why is the second reinvite marked as recvonly.
> I guess the second reinvited comes from legB because of the transfer was
> accepted - but why reconly - and why does legB sends a BYE after that one?
> 
> Thanks a lot



From mail at paulzillmann.de  Sat Jan 21 02:55:13 2017
From: mail at paulzillmann.de (Paul Zillmann)
Date: Sat, 21 Jan 2017 00:55:13 +0100
Subject: [Freeswitch-users] leg-B cuts off connection while transfer
In-Reply-To: <595AF1E4-BA71-4ADB-B0C1-B7B7470A2A4F@jerris.com>
References: <2c70a7a9-eafe-5b8a-2129-64c24dc6b4ab@paulzillmann.de>
	<595AF1E4-BA71-4ADB-B0C1-B7B7470A2A4F@jerris.com>
Message-ID: <582a3f50-14f5-4174-a9e5-a1dc556a265f@paulzillmann.de>

I've contacted legB operator about the issue.
Anything I could do on legA side to adress the issue?

Am 20.01.2017 um 18:58 schrieb Michael Jerris:
> You would have to ask whatever is sending you the BYE.  Sometimes misconfigured things can hang up when on hold due to media timeout, sometimes other issues.  From this information its impossible to know.
>
>
>> On Jan 20, 2017, at 12:42 PM, Paul Zillmann <mail at paulzillmann.de> wrote:
>>
>> Hello Folks,
>>
>> a user hooked me on, that whenever he calls certain numbers, and was put
>> into hold (call is transfered), that legB sends a BYE after all.
>> Is this a misconfigure on the user's side - or is legB misconfigured?
>>
>> The siptrace attatched seems to be okay - but what I don't understand
>> is, why is the second reinvite marked as recvonly.
>> I guess the second reinvited comes from legB because of the transfer was
>> accepted - but why reconly - and why does legB sends a BYE after that one?
>>
>> Thanks a lot
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat Jan 21 03:03:56 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 20 Jan 2017 19:03:56 -0500
Subject: [Freeswitch-users] leg-B cuts off connection while transfer
In-Reply-To: <582a3f50-14f5-4174-a9e5-a1dc556a265f@paulzillmann.de>
References: <2c70a7a9-eafe-5b8a-2129-64c24dc6b4ab@paulzillmann.de>
	<595AF1E4-BA71-4ADB-B0C1-B7B7470A2A4F@jerris.com>
	<582a3f50-14f5-4174-a9e5-a1dc556a265f@paulzillmann.de>
Message-ID: <BE358B03-7A65-461C-A2EF-713EBE6F48D0@jerris.com>

Hard to answer without knowing WHY they are hanging up.  Are we doing DTX to them?  If so, you could make sure constant RTP is going to them.  Thats just a wild guess based on what i see here.  They should be able to tell you why they are hanging up the call.



> On Jan 20, 2017, at 6:55 PM, Paul Zillmann <mail at paulzillmann.de> wrote:
> 
> I've contacted legB operator about the issue.
> Anything I could do on legA side to adress the issue?
> 
> Am 20.01.2017 um 18:58 schrieb Michael Jerris:
>> You would have to ask whatever is sending you the BYE.  Sometimes misconfigured things can hang up when on hold due to media timeout, sometimes other issues.  From this information its impossible to know.
>> 
>> 
>>> On Jan 20, 2017, at 12:42 PM, Paul Zillmann <mail at paulzillmann.de> wrote:
>>> 
>>> Hello Folks,
>>> 
>>> a user hooked me on, that whenever he calls certain numbers, and was put
>>> into hold (call is transfered), that legB sends a BYE after all.
>>> Is this a misconfigure on the user's side - or is legB misconfigured?
>>> 
>>> The siptrace attatched seems to be okay - but what I don't understand
>>> is, why is the second reinvite marked as recvonly.
>>> I guess the second reinvited comes from legB because of the transfer was
>>> accepted - but why reconly - and why does legB sends a BYE after that one?
>>> 
>>> Thanks a lot



From findmeinwland at gmail.com  Sun Jan 22 15:20:00 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Sun, 22 Jan 2017 17:20:00 +0500
Subject: [Freeswitch-users] centos 7 service freeswitch start
Message-ID: <CAOLEQ7fxiM9bhoOe25mjgZ0BdtpHVdW5zrV3H5gF8FrYv1fvYw@mail.gmail.com>

Just wanted to say, that when you do fresh install Freeswitch on Centos 7,
`service freeswitch start` command may not work (for me it not worked
twice). In this case you need to allow access for user `freeswitch` or
group `daemon` access /etc/sysconfig/freeswitch

fro example, run `chown root:daemon /etc/sysconfig/freeswitch`


-- 
?Arthur?
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From rw at panorgan.ch  Mon Jan 23 10:20:36 2017
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Mon, 23 Jan 2017 08:20:36 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
Message-ID: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>

Hi,

Is it possible to set "sip_invite_params='user=phone'" for all outgoing 
calls / all outgoing calls using a specific gateway?

I'm searching a solution which work with the dialplan and also with
calls over mod_event_socket and "originate".

I tried to set the variable in "conf/sip_profiles/external.xml" and
"conf/sip_profiles/external/outgoing.xml" but I don't know if this was 
the wrong place or if I was just using the wrong syntax.

Thanks,
Ren?


From ksrigo at gmail.com  Mon Jan 23 12:42:27 2017
From: ksrigo at gmail.com (KSrigo)
Date: Mon, 23 Jan 2017 10:42:27 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
Message-ID: <9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>

hi,

In your dialplan use:

<action application="bridge" data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>

In your ESL script:

originate {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212 &park()

PS: sip_invite_params only apply to the RURI, If you wish to modify the To param son the invite you must use sip_invite_to_params the same way.

Srigo

> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch> wrote:
> 
> Hi,
> 
> Is it possible to set "sip_invite_params='user=phone'" for all outgoing 
> calls / all outgoing calls using a specific gateway?
> 
> I'm searching a solution which work with the dialplan and also with
> calls over mod_event_socket and "originate".
> 
> I tried to set the variable in "conf/sip_profiles/external.xml" and
> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was 
> the wrong place or if I was just using the wrong syntax.
> 
> Thanks,
> Ren?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rw at panorgan.ch  Mon Jan 23 13:03:23 2017
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Mon, 23 Jan 2017 11:03:23 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
	<9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
Message-ID: <2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>

Hi Srigo

That is what I am doing at the moment.

What I would like to know is if there is a way to globaly set this 
variable in one (XML) file so that I don't have to remeber to always add 
it and also can change at only one place if I ever need to.


Am 23.01.17 um 10:42 schrieb KSrigo:
> hi,
>
> In your dialplan use:
>
> <action application="bridge"
> data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>
>
> In your ESL script:
>
> originate
> {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212 &park()
>
> PS: sip_invite_params only apply to the RURI, If you wish to modify the
> To param son the invite you must use sip_invite_to_params the same way.
>
> Srigo
>
>> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch
>> <mailto:rw at panorgan.ch>> wrote:
>>
>> Hi,
>>
>> Is it possible to set "sip_invite_params='user=phone'" for all outgoing
>> calls / all outgoing calls using a specific gateway?
>>
>> I'm searching a solution which work with the dialplan and also with
>> calls over mod_event_socket and "originate".
>>
>> I tried to set the variable in "conf/sip_profiles/external.xml" and
>> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was
>> the wrong place or if I was just using the wrong syntax.
>>
>> Thanks,
>> Ren?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ksrigo at gmail.com  Mon Jan 23 13:03:51 2017
From: ksrigo at gmail.com (KSrigo)
Date: Mon, 23 Jan 2017 11:03:51 +0100
Subject: [Freeswitch-users] Replace disposition cause
In-Reply-To: <CAGOm0PJdLn01W4jKiaUTf4Q5tRv0uPGHccYO9u313B7Ox50yWw@mail.gmail.com>
References: <CAGOm0PKEd7hQN4JCF_RTt2s4izVYeq9xSTkWgziiL0PnGr+snw@mail.gmail.com>
	<CANsY4DJyCGFu83Vyo_qFSng3C8u3EQUT2p1PK-VyV90RXoOkqA@mail.gmail.com>
	<CAL+DTgLPA9dbX6Wq6juH7KfG8r8QboSLKwHKT6zUPjed-rVqqw@mail.gmail.com>
	<CAGOm0P+Ld6kDthxmYbcnPmPjnXfQbNQtdnwqoMreZDTeUF9V1A@mail.gmail.com>
	<CAGOm0PKeJSJpqKvcQVYY_C0WNneofdiUBuiVbvJRzekrZdqFdQ@mail.gmail.com>
	<CAGOm0PJdLn01W4jKiaUTf4Q5tRv0uPGHccYO9u313B7Ox50yWw@mail.gmail.com>
Message-ID: <7EE8952B-B6E9-416A-84FF-B8D27D0B20CB@gmail.com>

Hi,

try this:
<action application="set" data="continue_on_fail=true?/>
<action application="set" data="hangup_after_bridge=true?/>
<action application="bridge" data="sofia/gateway/outgoing/12135551212?/>

<action application="hangup" data="NORMAL_CIRCUIT_CONGESTION?/>
or 
<action application=?respond" data="'503 SERVICE UNAVAILABLE?/>

Srigo


> On Jan 12, 2017, at 5:44 PM, devang nathwani <devang.nathwani31589 at gmail.com> wrote:
> 
> Can i have any suggestion on this issue?
> i have tried creating new dialplan as shown here
> https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause <https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause>
> as well as i have tried external lua to replace the cause code, suggest here,
> https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause <https://wiki.freeswitch.org/wiki/Variable_proto_specific_hangup_cause>
> 
> i am also attaching the sip log for the 480 
> https://pastebin.freeswitch.org/view/0dbbf7d7 <https://pastebin.freeswitch.org/view/0dbbf7d7>
> 
> On Fri, Jan 6, 2017 at 3:45 PM, devang nathwani <devang.nathwani31589 at gmail.com <mailto:devang.nathwani31589 at gmail.com>> wrote:
> yeah, i am getting reference from here,
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables <https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables>
> 
> On Fri, Jan 6, 2017 at 3:38 PM, devang nathwani <devang.nathwani31589 at gmail.com <mailto:devang.nathwani31589 at gmail.com>> wrote:
> Hello,
> 
> no i haven't read when i drafted the previous mail!
> 
> The scenario is something, gateway is completely down suppose because of hardware failure or electricity issue.
> 
> so freeswitch trying to send caller's request to gateway, as gateway is completely down freeswitch continuously trying to send 'INVITE' request to gateway, getting nothing in response so freeswitch sending 480 TEMPORARILY_UNAVAILABLE to caller but i want to change with '503 SERVICE_UNAVAILABLE'
> 
> and fscli showing, [CS_EXECUTE] [ALLOTTED_TIMEOUT] and [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 
> On Fri, Jan 6, 2017 at 3:18 PM, Gonzalo Gasca Meza <gascagonzalo at gmail.com <mailto:gascagonzalo at gmail.com>> wrote:
> 
> Can you share more details about the call flow and debug if possible? Are you planning to replace an incoming disconnect code or you want to send an specific disconnect code?
> 
> <action application="set" data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/>
> 
> continue_on_fail "Controls what happens when the called party can not be reached (busy/offline). If "true" the dialplan continues to be processed. If "false" the dialplan will stop processing. Can contain the return messages that will continue on fail also."
> 
> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/>
> 
> failure_causes  Controls which failure causes will be considered as a failure to the bridge(s). This will change the values for which continue_on_fail will fail by default unless continue_on_fail is set to true.
> 
> Depending of your flow you can use:
> 
> <action application="hangup" data="SERVICE_UNAVAILABLE"/>
> 
> 
> On Fri, Jan 6, 2017 at 1:07 AM, Vladyslav Zakhozhai <v.zakhozhai at gmail.com <mailto:v.zakhozhai at gmail.com>> wrote:
> Devang, I'm no sure but this config should not work.
> I think that you need to handle failure_causes after the bridge and use respond command to respond with cause you need.
> 
> Maybe I am wrong but this approach will do the work.
> 
> 2017-01-06 10:43 GMT+02:00 devang nathwani <devang.nathwani31589 at gmail.com <mailto:devang.nathwani31589 at gmail.com>>:
> Hello,
> 
> I want to replace disposition cause '480 TEMPORARILY_UNAVAILABLE' with '503 SERVICE_UNAVAILABLE'; 
> 
> tried;
> <action application="set"data="continue_on_fail=TEMPORARILY_UNAVAILABLE"/> <action application="set" data="failure_causes=SERVICE_UNAVAILABLE"/> before bridge application
> But its not working
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> ? ?????????,
> ????????? ???????
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ksrigo at gmail.com  Mon Jan 23 13:39:30 2017
From: ksrigo at gmail.com (KSrigo)
Date: Mon, 23 Jan 2017 11:39:30 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
	<9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
	<2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>
Message-ID: <59142575-4AF4-4880-83BB-F5DAF3DB018B@gmail.com>

In this case, put this at the top in your dialplan (it ll match everything):

<extension name="Default_parameters" continue="true">
  <condition field="destination_number" expression="^.*$" break="never">
    <action application=?export" data="sip_invite_params=user=phone? inline=?true"/>
  </condition>
</extension>

Srigo

> On Jan 23, 2017, at 11:03 AM, Ren? Weiss <rw at panorgan.ch> wrote:
> 
> Hi Srigo
> 
> That is what I am doing at the moment.
> 
> What I would like to know is if there is a way to globaly set this 
> variable in one (XML) file so that I don't have to remeber to always add 
> it and also can change at only one place if I ever need to.
> 
> 
> Am 23.01.17 um 10:42 schrieb KSrigo:
>> hi,
>> 
>> In your dialplan use:
>> 
>> <action application="bridge"
>> data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>
>> 
>> In your ESL script:
>> 
>> originate
>> {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212 &park()
>> 
>> PS: sip_invite_params only apply to the RURI, If you wish to modify the
>> To param son the invite you must use sip_invite_to_params the same way.
>> 
>> Srigo
>> 
>>> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch
>>> <mailto:rw at panorgan.ch <mailto:rw at panorgan.ch>>> wrote:
>>> 
>>> Hi,
>>> 
>>> Is it possible to set "sip_invite_params='user=phone'" for all outgoing
>>> calls / all outgoing calls using a specific gateway?
>>> 
>>> I'm searching a solution which work with the dialplan and also with
>>> calls over mod_event_socket and "originate".
>>> 
>>> I tried to set the variable in "conf/sip_profiles/external.xml" and
>>> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was
>>> the wrong place or if I was just using the wrong syntax.
>>> 
>>> Thanks,
>>> Ren?
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://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 <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 rw at panorgan.ch  Mon Jan 23 14:29:50 2017
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Mon, 23 Jan 2017 12:29:50 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <59142575-4AF4-4880-83BB-F5DAF3DB018B@gmail.com>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
	<9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
	<2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>
	<59142575-4AF4-4880-83BB-F5DAF3DB018B@gmail.com>
Message-ID: <9d4b7485-0d9e-9bd6-bb14-03e7efda0ca5@panorgan.ch>

Does a ESL command like
"originate sofia/gateway/outgoing/12135551212 &park()" even run through 
the dialplan? My impression was that I would have to use the "loopback" 
endpoint for this, which doesn't work with some of our scripts.

Is there really no way to simple set or export "sip_invite_params"
in a file like "vars.xml", "sip_profiles/external.xml" or 
"sip_profiles/external/outgoing.xml"?

Am 23.01.17 um 11:39 schrieb KSrigo:
> In this case, put this at the top in your dialplan (it ll match everything):
>
> <extension name="Default_parameters" continue="true">
>   <condition field="destination_number" expression="^.*$" break="never">
>     <action application=?export" data="sip_invite_params=user=phone?
> inline=?true"/>
>   </condition>
> </extension>
>
> Srigo
>
>> On Jan 23, 2017, at 11:03 AM, Ren? Weiss <rw at panorgan.ch
>> <mailto:rw at panorgan.ch>> wrote:
>>
>> Hi Srigo
>>
>> That is what I am doing at the moment.
>>
>> What I would like to know is if there is a way to globaly set this
>> variable in one (XML) file so that I don't have to remeber to always add
>> it and also can change at only one place if I ever need to.
>>
>>
>> Am 23.01.17 um 10:42 schrieb KSrigo:
>>> hi,
>>>
>>> In your dialplan use:
>>>
>>> <action application="bridge"
>>> data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>
>>>
>>> In your ESL script:
>>>
>>> originate
>>> {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212 &park()
>>>
>>> PS: sip_invite_params only apply to the RURI, If you wish to modify the
>>> To param son the invite you must use sip_invite_to_params the same way.
>>>
>>> Srigo
>>>
>>>> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch
>>>> <mailto:rw at panorgan.ch>
>>>> <mailto:rw at panorgan.ch>> wrote:
>>>>
>>>> Hi,
>>>>
>>>> Is it possible to set "sip_invite_params='user=phone'" for all outgoing
>>>> calls / all outgoing calls using a specific gateway?
>>>>
>>>> I'm searching a solution which work with the dialplan and also with
>>>> calls over mod_event_socket and "originate".
>>>>
>>>> I tried to set the variable in "conf/sip_profiles/external.xml" and
>>>> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was
>>>> the wrong place or if I was just using the wrong syntax.
>>>>
>>>> Thanks,
>>>> Ren?
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <http://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 ksrigo at gmail.com  Mon Jan 23 15:22:46 2017
From: ksrigo at gmail.com (KSrigo)
Date: Mon, 23 Jan 2017 13:22:46 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <9d4b7485-0d9e-9bd6-bb14-03e7efda0ca5@panorgan.ch>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
	<9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
	<2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>
	<59142575-4AF4-4880-83BB-F5DAF3DB018B@gmail.com>
	<9d4b7485-0d9e-9bd6-bb14-03e7efda0ca5@panorgan.ch>
Message-ID: <2616B237-8DB7-42ED-A10F-302E67D464DB@gmail.com>

put this in your "sip_profiles/external/outgoing.xml?:

<include>
    <gateway name="outgoing">
      <variables>
        <variable name="sip_invite_params" value="user=phone"/>
      </variables>
      ...
    </gateway>
 </include>

Srigo


> On Jan 23, 2017, at 12:29 PM, Ren? Weiss <rw at panorgan.ch> wrote:
> 
> Does a ESL command like
> "originate sofia/gateway/outgoing/12135551212 &park()" even run through 
> the dialplan? My impression was that I would have to use the "loopback" 
> endpoint for this, which doesn't work with some of our scripts.
> 
> Is there really no way to simple set or export "sip_invite_params"
> in a file like "vars.xml", "sip_profiles/external.xml" or 
> "sip_profiles/external/outgoing.xml"?
> 
> Am 23.01.17 um 11:39 schrieb KSrigo:
>> In this case, put this at the top in your dialplan (it ll match everything):
>> 
>> <extension name="Default_parameters" continue="true">
>>  <condition field="destination_number" expression="^.*$" break="never">
>>    <action application=?export" data="sip_invite_params=user=phone?
>> inline=?true"/>
>>  </condition>
>> </extension>
>> 
>> Srigo
>> 
>>> On Jan 23, 2017, at 11:03 AM, Ren? Weiss <rw at panorgan.ch <mailto:rw at panorgan.ch>
>>> <mailto:rw at panorgan.ch <mailto:rw at panorgan.ch>>> wrote:
>>> 
>>> Hi Srigo
>>> 
>>> That is what I am doing at the moment.
>>> 
>>> What I would like to know is if there is a way to globaly set this
>>> variable in one (XML) file so that I don't have to remeber to always add
>>> it and also can change at only one place if I ever need to.
>>> 
>>> 
>>> Am 23.01.17 um 10:42 schrieb KSrigo:
>>>> hi,
>>>> 
>>>> In your dialplan use:
>>>> 
>>>> <action application="bridge"
>>>> data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>
>>>> 
>>>> In your ESL script:
>>>> 
>>>> originate
>>>> {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212 &park()
>>>> 
>>>> PS: sip_invite_params only apply to the RURI, If you wish to modify the
>>>> To param son the invite you must use sip_invite_to_params the same way.
>>>> 
>>>> Srigo
>>>> 
>>>>> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch <mailto:rw at panorgan.ch>
>>>>> <mailto:rw at panorgan.ch <mailto:rw at panorgan.ch>>
>>>>> <mailto:rw at panorgan.ch <mailto:rw at panorgan.ch>>> wrote:
>>>>> 
>>>>> Hi,
>>>>> 
>>>>> Is it possible to set "sip_invite_params='user=phone'" for all outgoing
>>>>> calls / all outgoing calls using a specific gateway?
>>>>> 
>>>>> I'm searching a solution which work with the dialplan and also with
>>>>> calls over mod_event_socket and "originate".
>>>>> 
>>>>> I tried to set the variable in "conf/sip_profiles/external.xml" and
>>>>> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was
>>>>> the wrong place or if I was just using the wrong syntax.
>>>>> 
>>>>> Thanks,
>>>>> Ren?
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>> <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>> <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/ <http://www.freeswitchsolutions.com/>>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>>
>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/ <http://confluence.freeswitch.org/>>
>>>>> http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/ <http://www.cluecon.com/>>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <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://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>>
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>>
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/ <http://www.freeswitchsolutions.com/>>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/ <http://confluence.freeswitch.org/>>
>>> http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/ <http://www.cluecon.com/>>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <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://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>>
>>> http://www.freeswitch.org <http://www.freeswitch.org/> <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

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From rw at panorgan.ch  Mon Jan 23 15:52:36 2017
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Mon, 23 Jan 2017 13:52:36 +0100
Subject: [Freeswitch-users] sip_invite_params for all outgoing calls
In-Reply-To: <2616B237-8DB7-42ED-A10F-302E67D464DB@gmail.com>
References: <654f4ad3-348a-5fb2-1e78-0f07f601168f@panorgan.ch>
	<9A4BE6F6-8C75-43F4-889E-FF5940F26CF2@gmail.com>
	<2c0e7a5f-d020-6e33-7d2f-a025876eab5e@panorgan.ch>
	<59142575-4AF4-4880-83BB-F5DAF3DB018B@gmail.com>
	<9d4b7485-0d9e-9bd6-bb14-03e7efda0ca5@panorgan.ch>
	<2616B237-8DB7-42ED-A10F-302E67D464DB@gmail.com>
Message-ID: <1d0786e4-a76c-773a-dd83-610dd0e9e601@panorgan.ch>

Thanks, it works now.

I think I must have had something wrong with the syntax last time I 
tried to put the variable into outgoing.xml.

Am 23.01.17 um 13:22 schrieb KSrigo:
> put this in your "sip_profiles/external/outgoing.xml?:
>
> <include>
>     <gateway name="outgoing">
>       <variables>
>         <variable name="sip_invite_params" value="user=phone"/>
>       </variables>
>       ...
>     </gateway>
>  </include>
>
> Srigo
>
>
>> On Jan 23, 2017, at 12:29 PM, Ren? Weiss <rw at panorgan.ch
>> <mailto:rw at panorgan.ch>> wrote:
>>
>> Does a ESL command like
>> "originate sofia/gateway/outgoing/12135551212 &park()" even run through
>> the dialplan? My impression was that I would have to use the "loopback"
>> endpoint for this, which doesn't work with some of our scripts.
>>
>> Is there really no way to simple set or export "sip_invite_params"
>> in a file like "vars.xml", "sip_profiles/external.xml" or
>> "sip_profiles/external/outgoing.xml"?
>>
>> Am 23.01.17 um 11:39 schrieb KSrigo:
>>> In this case, put this at the top in your dialplan (it ll match
>>> everything):
>>>
>>> <extension name="Default_parameters" continue="true">
>>>  <condition field="destination_number" expression="^.*$" break="never">
>>>    <action application=?export" data="sip_invite_params=user=phone?
>>> inline=?true"/>
>>>  </condition>
>>> </extension>
>>>
>>> Srigo
>>>
>>>> On Jan 23, 2017, at 11:03 AM, Ren? Weiss <rw at panorgan.ch
>>>> <mailto:rw at panorgan.ch>
>>>> <mailto:rw at panorgan.ch>> wrote:
>>>>
>>>> Hi Srigo
>>>>
>>>> That is what I am doing at the moment.
>>>>
>>>> What I would like to know is if there is a way to globaly set this
>>>> variable in one (XML) file so that I don't have to remeber to always add
>>>> it and also can change at only one place if I ever need to.
>>>>
>>>>
>>>> Am 23.01.17 um 10:42 schrieb KSrigo:
>>>>> hi,
>>>>>
>>>>> In your dialplan use:
>>>>>
>>>>> <action application="bridge"
>>>>> data="{sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212"/>
>>>>>
>>>>> In your ESL script:
>>>>>
>>>>> originate
>>>>> {sip_invite_params=user=phone}sofia/gateway/outgoing/12135551212
>>>>> &park()
>>>>>
>>>>> PS: sip_invite_params only apply to the RURI, If you wish to modify the
>>>>> To param son the invite you must use sip_invite_to_params the same way.
>>>>>
>>>>> Srigo
>>>>>
>>>>>> On Jan 23, 2017, at 8:20 AM, Ren? Weiss <rw at panorgan.ch
>>>>>> <mailto:rw at panorgan.ch>
>>>>>> <mailto:rw at panorgan.ch>
>>>>>> <mailto:rw at panorgan.ch>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Is it possible to set "sip_invite_params='user=phone'" for all
>>>>>> outgoing
>>>>>> calls / all outgoing calls using a specific gateway?
>>>>>>
>>>>>> I'm searching a solution which work with the dialplan and also with
>>>>>> calls over mod_event_socket and "originate".
>>>>>>
>>>>>> I tried to set the variable in "conf/sip_profiles/external.xml" and
>>>>>> "conf/sip_profiles/external/outgoing.xml" but I don't know if this was
>>>>>> the wrong place or if I was just using the wrong syntax.
>>>>>>
>>>>>> Thanks,
>>>>>> Ren?
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>> <mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org>
>>>>>> http://www.freeswitchsolutions.com
>>>>>> <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/>
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>>>>>> http://confluence.freeswitch.org
>>>>>> <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/>
>>>>>> http://www.cluecon.com
>>>>>> <http://www.cluecon.com/> <http://www.cluecon.com/>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> 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://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://www.freeswitch.org
>>>>>> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>> <mailto:consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/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> <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org
>>>> <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/>
>>>> http://www.cluecon.com
>>>> <http://www.cluecon.com/> <http://www.cluecon.com/>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> 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://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org
>>>> <http://www.freeswitch.org/> <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Mon Jan 23 18:26:48 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 23 Jan 2017 09:26:48 -0600
Subject: [Freeswitch-users] BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS IF
	YOU USE FAXING)
Message-ID: <CAEJMVkB7mnOrgwoqk2Tb37uHj1=iB5s-H_m51x7qng4Qj3XEkQ@mail.gmail.com>

FreeSWITCHers,

In 1.6.14 we did a slight behavior change related to T.38 re-invite
acceptance.  Before 1.6.14 we would by default accept the re-invite even if
t38 wasn't enabled and switch to UDPTL mode.  This was an incorrect
behavior to do by default.

But one caveat in 1.6.14, We by default 488 the fax but switch the media
mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for
1.6.15.

If you wish to retain the previous behavior you can set fax_enable_t38=true
in your vars.xml, You can also not set that and just set refuse_t38=true if
you wish to never do t.38.

These default behaviors are fixed in our next release 1.6.15.

Thanks,

-- 

*Brian West*
brian at freeswitch.org


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

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From brian at freeswitch.org  Mon Jan 23 20:03:09 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 23 Jan 2017 11:03:09 -0600
Subject: [Freeswitch-users] BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS
	IF YOU USE FAXING)
In-Reply-To: <CAEJMVkB7mnOrgwoqk2Tb37uHj1=iB5s-H_m51x7qng4Qj3XEkQ@mail.gmail.com>
References: <CAEJMVkB7mnOrgwoqk2Tb37uHj1=iB5s-H_m51x7qng4Qj3XEkQ@mail.gmail.com>
Message-ID: <CAEJMVkBDATJU599hbQ58ARGiWxNnDp9vQF77ed0DaVLuVYEmZA@mail.gmail.com>

To clarify this work around will work on 1.6.14:

If you wish to retain the previous behavior you can set fax_enable_t38=true
in your vars.xml, You can also not set that and just set refuse_t38=true if
you wish to never do t.38.

/b

On Mon, Jan 23, 2017 at 9:26 AM, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
> In 1.6.14 we did a slight behavior change related to T.38 re-invite
> acceptance.  Before 1.6.14 we would by default accept the re-invite even if
> t38 wasn't enabled and switch to UDPTL mode.  This was an incorrect
> behavior to do by default.
>
> But one caveat in 1.6.14, We by default 488 the fax but switch the media
> mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for
> 1.6.15.
>
> If you wish to retain the previous behavior you can set
> fax_enable_t38=true in your vars.xml, You can also not set that and just
> set refuse_t38=true if you wish to never do t.38.
>
> These default behaviors are fixed in our next release 1.6.15.
>
> Thanks,
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>



-- 

*Brian West*
brian at freeswitch.org


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

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From haus.voip at gmail.com  Sun Jan 22 01:35:05 2017
From: haus.voip at gmail.com (abc1)
Date: Sat, 21 Jan 2017 22:35:05 +0000
Subject: [Freeswitch-users] mod_radius_cdr
Message-ID: <CAGWmP0ZqVjFX+h2m7QE2YZZ3nrXiifOFcUNEVs87GNXS=t44Xw@mail.gmail.com>

Hi List

Is there a DB schema for generating the radacct table in freeradius to work
with mod_radius_cdr?

Thanks!
Alex
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From raman.chv at gmail.com  Mon Jan 23 08:32:17 2017
From: raman.chv at gmail.com (Ram Anji)
Date: Mon, 23 Jan 2017 11:02:17 +0530
Subject: [Freeswitch-users] Fwd: NOTIFYs not received for conference
 SUBSCRIBE events
In-Reply-To: <CAEJMVkBO30Gt=5-_yMB-1jexfxp47tFO5LvyQhS=LuU5Cn+01A@mail.gmail.com>
References: <CAArNKuc16hnERYcbH9-C7atw-zpry4s0zUkJdAfwhfaAAZ+=KQ@mail.gmail.com>
	<CAArNKue7QrSUbg-9qV6DHNAq-9ceAAUL8Y=m4XGM95y5io6quQ@mail.gmail.com>
	<CAEJMVkBO30Gt=5-_yMB-1jexfxp47tFO5LvyQhS=LuU5Cn+01A@mail.gmail.com>
Message-ID: <CAArNKueHG4CaJCqgnrBhOiYVubGmtxYKZLhWo3-VTy6MJ9N2Eg@mail.gmail.com>

Yes Brain,

Enabled CDR in conference.conf.xml

<profile name="default">
      <param name="cdr-log-dir" value="auto"/>
.....

And i am seeing the cdr after end of
conference /var/log/freeswitch/conference_cdr

enclosed the cdr for reference.



On Fri, Jan 20, 2017 at 9:02 PM, Brian West <brian at freeswitch.org> wrote:

> Do you have conference CDR's enabled and have you set the flag to enable
> this?
>
> On Thu, Jan 19, 2017 at 10:21 AM, Ram Anji <raman.chv at gmail.com> wrote:
>
>> Hi,
>>
>> I am testing conference on freeswitch with jitsi. Jitsi sending SUBSCRIBE
>> event after receiving 200 ok of call with focus parameter along with
>> contact header.
>> Server is responding 202 of subscription but no NOTIFY is observed for
>> the subscription.
>>
>> I am testing with FreeSWITCH version: 1.6.14~64bit ( 64bit) where it is
>> installed from freeswitch rpms and os is centos.
>>
>> Enclosed the console logs for the reference
>>
>> Regards,
>> Raman
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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 raman.chv at gmail.com  Mon Jan 23 20:08:39 2017
From: raman.chv at gmail.com (Ram)
Date: Mon, 23 Jan 2017 22:38:39 +0530
Subject: [Freeswitch-users] Fwd: NOTIFYs not received for conference
 SUBSCRIBE events
In-Reply-To: <CAEJMVkBO30Gt=5-_yMB-1jexfxp47tFO5LvyQhS=LuU5Cn+01A@mail.gmail.com>
References: <CAArNKuc16hnERYcbH9-C7atw-zpry4s0zUkJdAfwhfaAAZ+=KQ@mail.gmail.com>
	<CAArNKue7QrSUbg-9qV6DHNAq-9ceAAUL8Y=m4XGM95y5io6quQ@mail.gmail.com>
	<CAEJMVkBO30Gt=5-_yMB-1jexfxp47tFO5LvyQhS=LuU5Cn+01A@mail.gmail.com>
Message-ID: <CAArNKuc0pCJ8q_pO+e6zqFVpvKdsRh6UwkgJzVoiR_pAk=AWHw@mail.gmail.com>

Yes Enabled CDR in conference.conf.xml

<profile name="default">
      <param name="cdr-log-dir" value="auto"/>
.....

And i am seeing the cdr after end of conference /var/log/
freeswitch/conference_cdr

Following is the cdr[25a1fc00-e12c-11e6-ae85-f546e975fa69.cdr.xml] for
reference

<?xml version="1.0"?>
<cdr>
  <conference>
    <name>30000-10.90.111.196</name>
    <hostname>localhost.localdomain</hostname>
    <rate>8000</rate>
    <interval>20</interval>
    <start_time type="UNIX-epoch">1485149040</start_time>
    <end_time endconference_forced="false"
type="UNIX-epoch">1485149209</end_time>
    <members>
      <member type="caller">
        <join_time type="UNIX-epoch">1485149158</join_time>
        <leave_time type="UNIX-epoch">1485149209</leave_time>
        <flags>
          <is_moderator>false</is_moderator>
          <end_conference>false</end_conference>
          <was_kicked>false</was_kicked>
          <is_ghost>false</is_ghost>
        </flags>
        <caller_profile>
          <username>1000</username>
          <dialplan>XML</dialplan>
          <caller_id_name>1000</caller_id_name>
          <caller_id_number>1000</caller_id_number>
          <callee_id_name></callee_id_name>
          <callee_id_number></callee_id_number>
          <ani>1000</ani>
          <aniii></aniii>
          <network_addr>10.90.111.21</network_addr>
          <rdnis></rdnis>
          <destination_number>30000</destination_number>
          <uuid>65ca3446-e12c-11e6-ae88-f546e975fa69</uuid>
          <source>mod_sofia</source>
          <context>default</context>
          <chan_name>sofia/internal/1000 at 10.90.111.196</chan_name>
        </caller_profile>
      </member>
      <member type="caller">
        <join_time type="UNIX-epoch">1485149040</join_time>
        <leave_time type="UNIX-epoch">1485149196</leave_time>
        <flags>
          <is_moderator>false</is_moderator>
          <end_conference>false</end_conference>
          <was_kicked>false</was_kicked>
          <is_ghost>false</is_ghost>
        </flags>
        <caller_profile>
          <username>1001</username>
          <dialplan>XML</dialplan>
          <caller_id_name>1001</caller_id_name>
          <caller_id_number>1001</caller_id_number>
          <callee_id_name></callee_id_name>
          <callee_id_number></callee_id_number>
          <ani>1001</ani>
          <aniii></aniii>
          <network_addr>10.90.111.21</network_addr>
          <rdnis></rdnis>
          <destination_number>30000</destination_number>
          <uuid>1f09eae2-e12c-11e6-ae76-f546e975fa69</uuid>
          <source>mod_sofia</source>
          <context>default</context>
          <chan_name>sofia/internal/1001 at 10.90.111.196</chan_name>
        </caller_profile>
      </member>
    </members>
    <rejected></rejected>
  </conference>
</cdr>



On Fri, Jan 20, 2017 at 9:02 PM, Brian West <brian at freeswitch.org> wrote:

> Do you have conference CDR's enabled and have you set the flag to enable
> this?
>
> On Thu, Jan 19, 2017 at 10:21 AM, Ram Anji <raman.chv at gmail.com> wrote:
>
>> Hi,
>>
>> I am testing conference on freeswitch with jitsi. Jitsi sending SUBSCRIBE
>> event after receiving 200 ok of call with focus parameter along with
>> contact header.
>> Server is responding 202 of subscription but no NOTIFY is observed for
>> the subscription.
>>
>> I am testing with FreeSWITCH version: 1.6.14~64bit ( 64bit) where it is
>> installed from freeswitch rpms and os is centos.
>>
>> Enclosed the console logs for the reference
>>
>> Regards,
>> Raman
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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 yehavi.bourvine at gmail.com  Mon Jan 23 21:55:25 2017
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 23 Jan 2017 20:55:25 +0200
Subject: [Freeswitch-users] BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS
 IF YOU USE FAXING)
In-Reply-To: <CAEJMVkBDATJU599hbQ58ARGiWxNnDp9vQF77ed0DaVLuVYEmZA@mail.gmail.com>
References: <CAEJMVkB7mnOrgwoqk2Tb37uHj1=iB5s-H_m51x7qng4Qj3XEkQ@mail.gmail.com>
	<CAEJMVkBDATJU599hbQ58ARGiWxNnDp9vQF77ed0DaVLuVYEmZA@mail.gmail.com>
Message-ID: <CAAtWKeH+owTquy1e4t6u4h-TVpmf5FfKwyqADuzXSfKuQ5GBDg@mail.gmail.com>

Hi Brian,

  Thanks for informing us. I just switched back to 1.6.13 an hour ago due
to users complaints (and did not have the time yet to dig more). However, I
have the following settings for all users:

          <action application="set" data="fax_enable_t38_request=true"/>
          <action application="set" data="fax_enable_t38=true"/>

which, according to your message above, should work ok. Anyway, I'll wait
to 1.6.15...

                              Thanks, __Yehavi:


2017-01-23 19:03 GMT+02:00 Brian West <brian at freeswitch.org>:

> To clarify this work around will work on 1.6.14:
>
> If you wish to retain the previous behavior you can set
> fax_enable_t38=true in your vars.xml, You can also not set that and just
> set refuse_t38=true if you wish to never do t.38.
>
> /b
>
> On Mon, Jan 23, 2017 at 9:26 AM, Brian West <brian at freeswitch.org> wrote:
>
>> FreeSWITCHers,
>>
>> In 1.6.14 we did a slight behavior change related to T.38 re-invite
>> acceptance.  Before 1.6.14 we would by default accept the re-invite even if
>> t38 wasn't enabled and switch to UDPTL mode.  This was an incorrect
>> behavior to do by default.
>>
>> But one caveat in 1.6.14, We by default 488 the fax but switch the media
>> mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for
>> 1.6.15.
>>
>> If you wish to retain the previous behavior you can set
>> fax_enable_t38=true in your vars.xml, You can also not set that and just
>> set refuse_t38=true if you wish to never do t.38.
>>
>> These default behaviors are fixed in our next release 1.6.15.
>>
>> Thanks,
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>> https://www.gofundme.com/freeswitch_ubuntu
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *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 brian at freeswitch.org  Mon Jan 23 22:59:58 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 23 Jan 2017 13:59:58 -0600
Subject: [Freeswitch-users] BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS
 IF YOU USE FAXING)
In-Reply-To: <CAAtWKeH+owTquy1e4t6u4h-TVpmf5FfKwyqADuzXSfKuQ5GBDg@mail.gmail.com>
References: <CAEJMVkB7mnOrgwoqk2Tb37uHj1=iB5s-H_m51x7qng4Qj3XEkQ@mail.gmail.com>
	<CAEJMVkBDATJU599hbQ58ARGiWxNnDp9vQF77ed0DaVLuVYEmZA@mail.gmail.com>
	<CAAtWKeH+owTquy1e4t6u4h-TVpmf5FfKwyqADuzXSfKuQ5GBDg@mail.gmail.com>
Message-ID: <CAEJMVkCtCp87U8PLHjG=gB4oJ3O68paTTL4XyHcipOxjkx_y_Q@mail.gmail.com>

You do NOT want 1.6.13 PERIOD if you're doing faxes or anything t.38
related.  You're in for some pain.

Use 1.6.14 or v1.6 branch which also has the final patches for 1.6.15.


/b


On Mon, Jan 23, 2017 at 12:55 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> Hi Brian,
>
>   Thanks for informing us. I just switched back to 1.6.13 an hour ago due
> to users complaints (and did not have the time yet to dig more). However, I
> have the following settings for all users:
>
>           <action application="set" data="fax_enable_t38_request=true"/>
>           <action application="set" data="fax_enable_t38=true"/>
>
> which, according to your message above, should work ok. Anyway, I'll wait
> to 1.6.15...
>
>                               Thanks, __Yehavi:
>
>
> 2017-01-23 19:03 GMT+02:00 Brian West <brian at freeswitch.org>:
>
>> To clarify this work around will work on 1.6.14:
>>
>> If you wish to retain the previous behavior you can set
>> fax_enable_t38=true in your vars.xml, You can also not set that and just
>> set refuse_t38=true if you wish to never do t.38.
>>
>> /b
>>
>> On Mon, Jan 23, 2017 at 9:26 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> FreeSWITCHers,
>>>
>>> In 1.6.14 we did a slight behavior change related to T.38 re-invite
>>> acceptance.  Before 1.6.14 we would by default accept the re-invite even if
>>> t38 wasn't enabled and switch to UDPTL mode.  This was an incorrect
>>> behavior to do by default.
>>>
>>> But one caveat in 1.6.14, We by default 488 the fax but switch the media
>>> mode to UDPTL which is also a bad behavior fixed in (FS-9943) slated for
>>> 1.6.15.
>>>
>>> If you wish to retain the previous behavior you can set
>>> fax_enable_t38=true in your vars.xml, You can also not set that and just
>>> set refuse_t38=true if you wish to never do t.38.
>>>
>>> These default behaviors are fixed in our next release 1.6.15.
>>>
>>> Thanks,
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>> https://www.gofundme.com/freeswitch_ubuntu
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>> https://www.gofundme.com/freeswitch_ubuntu
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *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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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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-- 

*Brian West*
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Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
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From findmeinwland at gmail.com  Tue Jan 24 09:16:25 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Tue, 24 Jan 2017 11:16:25 +0500
Subject: [Freeswitch-users] mod_radius_cdr
In-Reply-To: <CAGWmP0ZqVjFX+h2m7QE2YZZ3nrXiifOFcUNEVs87GNXS=t44Xw@mail.gmail.com>
References: <CAGWmP0ZqVjFX+h2m7QE2YZZ3nrXiifOFcUNEVs87GNXS=t44Xw@mail.gmail.com>
Message-ID: <CAOLEQ7eqsVkLCXad2cDX9G-95rzLz_y2k-=k2FnOL_hrjuhiQQ@mail.gmail.com>

?as i remeber, first you need to install DB backend for radius, for example
radius-mysql. When you install this package, in /etc/ridius/sql/mysql will
be a file named like 'schema.sql', this file is what you need?

2017-01-22 3:35 GMT+05:00 abc1 <haus.voip at gmail.com>:

> Hi List
>
> Is there a DB schema for generating the radacct table in freeradius to
> work with mod_radius_cdr?
>
> Thanks!
> Alex
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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-- 
?????
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From Chris.Young at enghouse.com  Tue Jan 24 11:51:12 2017
From: Chris.Young at enghouse.com (Chris Young)
Date: Tue, 24 Jan 2017 08:51:12 +0000
Subject: [Freeswitch-users] Random number generation
Message-ID: <1a76b473362b4768b0e85c5db09b5e03@UK-MAIL-001.edge.local>

Hello all,

On the wiki, there is an example of using mod_expr to generate a random number:

                <!-- generate random number between 1 and 100 -->
                <action application="set" data="rand_val=${expr(randomize(&x);ceil(random(0,100,&x)))}"/>
                <action application="log" data="INFO Random value is ${rand_val}"/>

This works well the first time but if the same dialplan instructions are executed subsequently, the same number is produced each time.

Is this by design? Is there any other way to seed the random number generator so it will produce different results, as randomize() doesn't seem to be doing quite what I thought it would?

Many thanks,
Chris

Chris Young
Software Engineer
[cid:image7482a0.PNG at dc00f514.4fb88205]
t:  +44 118 943 9249
e:  chris.young at enghouse.com<mailto:chris.young at enghouse.com>
w: www.enghouseinteractive.co.uk<http://www.enghouseinteractive.co.uk>
[cid:image6c1ba6.PNG at b8963a35.449174f5]
Enghouse Interactive (UK) Ltd is a company registered in England and Wales. Registered number: 04230977. Registered office: Imperium, Imperial Way, Reading, Berkshire, RG2 0TD

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From ksrigo at gmail.com  Tue Jan 24 12:14:19 2017
From: ksrigo at gmail.com (KSrigo)
Date: Tue, 24 Jan 2017 10:14:19 +0100
Subject: [Freeswitch-users] Random number generation
In-Reply-To: <1a76b473362b4768b0e85c5db09b5e03@UK-MAIL-001.edge.local>
References: <1a76b473362b4768b0e85c5db09b5e03@UK-MAIL-001.edge.local>
Message-ID: <1841435E-D7C1-4DAE-9C3D-D69FAC286B90@gmail.com>

Hi Chris,

I?m not sure if I understood exactly your needs. This is how I use randomize function:

Define Random Number Generator extension:

<extension name="random_number_gen" continue="true">
  <condition field="destination_number" expression="^random_number_gen$" break="never">
    <!-- generate random number between 1 and 100 -->
    <action application="set" data="rand_val=${expr(randomize(&x);ceil(random(0,100,&x)))}" inline="true"/>
    <action application="log" data="INFO Random value is ${rand_val}"/>
  </condition>
</extension>


then call it in your diaplan each time you need to get a random number set:

<action application="execute_extension" data="random_number_gen XML from-core"/>


Regards,
Srigo


> On Jan 24, 2017, at 9:51 AM, Chris Young <Chris.Young at enghouse.com> wrote:
> 
>