From Chris.Young at enghouse.com Tue Sep 1 11:31:00 2015
From: Chris.Young at enghouse.com (Chris Young)
Date: Tue, 1 Sep 2015 07:31:00 +0000
Subject: [Freeswitch-users] SIP profile not loading
Message-ID: <373842e4388c4eeb93fde54cfdbaea92@UK-MAIL-001.edge.local>
Hi all,
Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named (for example):
dummy.xml
external.xml
internal.xml
then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly so it appears to be valid.
Has anybody else seen this kind of behaviour or know what could be causing it?
Many thanks,
Chris
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From d.mordovin at dwide.com Tue Sep 1 12:56:08 2015
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Tue, 01 Sep 2015 12:56:08 +0400
Subject: [Freeswitch-users] Playing with conditions
Message-ID: <55E56828.102@dwide.com>
Hello
Why inner condition not works?
From avi at avimarcus.net Tue Sep 1 14:30:00 2015
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 1 Sep 2015 10:30:00 +0000
Subject: [Freeswitch-users] Playing with conditions
In-Reply-To: <55E56828.102@dwide.com>
References: <55E56828.102@dwide.com>
Message-ID: <0000014f8874ccc5-e8464c2a-7efa-4711-a201-5efec14fe0f2-000000@email.amazonses.com>
Short answer: Each extension only has 1 set of conditions.
The condition evaluating foobar is run *before* it gets set.
After the play_and_get_digits you should transfer/execute_extension to a
new extension that will evaluate ${foobar}
-Avi Marcus
BestFone
On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
wrote:
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gaz.foreman at gmail.com Tue Sep 1 14:49:54 2015
From: gaz.foreman at gmail.com (Gary Foreman)
Date: Tue, 1 Sep 2015 11:49:54 +0100
Subject: [Freeswitch-users] Broken silence with webrtc
Message-ID:
I've found that it occurs after any bridge, its not specific to the
originate command.
Would you require a wireshark trace or the output of the freeswitch console?
The scenario below reproduces the issue ...
Test extension
I originate a call from a polycom handset using g722 to the extension above.
I originate a call using the verto client to the extension above.
I get the uuids of the channels using show channels and use uuid_bridge
[uuid1] [uuid2] to merge the channels.
On Tue, Sep 1, 2015 at 11:30 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. How send in To header anonymous and in P-Asserted-Identity
> caller-id-number (Alex Polischuk)
> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
> 3. Broken silence with webrtc (Gary Foreman)
> 4. Re: Broken silence with webrtc (Brian West)
> 5. SIP profile not loading (Chris Young)
> 6. Playing with conditions (Dmitry Mordovin)
> 7. Re: Playing with conditions (Avi Marcus)
>
>
> ---------- Forwarded message ----------
> From: Alex Polischuk
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 17:02:25 +0300
> Subject: [Freeswitch-users] How send in To header anonymous and in
> P-Asserted-Identity caller-id-number
> Hi all,
>
> How I can define different users and domains in To and P-Asserted-Identity
> headers?
>
> Thanks,
> Alex
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 18:24:08 +0400
> Subject: [Freeswitch-users] Loop play while wait DTMF digit
> Hello
>
> This example play conf-pin.wav and wait DTMF.
>
>
>
>
>
>
>
>
>
> Is it possible to play WAV file in infinity loop and wait user DTMF?
>
> And how can I check DTMF input after user entered DTMF?
>
> In dialpeer, like.
> If ${DTMF} = 1 then bridge to XXX
> If ${DTMF} = 2 then play file and finish session
>
>
>
> Thank you.
> Dmitry
>
>
>
> ---------- Forwarded message ----------
> From: Gary Foreman
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Mon, 31 Aug 2015 20:36:59 +0100
> Subject: [Freeswitch-users] Broken silence with webrtc
> Hi all,
>
> Hoping someone can point me in the right direction because after several
> hours I'm out of ideas.
>
> I'm having an issue where the 2nd leg of a call that is bridged to a verto
> endpoint has broken silence. When there is no sound on the line the audio
> goes completely silent (all background noise is dropped) but around every
> third of a second it repeatedly cuts back in for a fraction, then goes
> completely silent again. This only happens when the call is created using
> the originate command.
>
> Can anyone give me an idea of where to look next? I have a wireshark trace
> that during playback shows the audio cutting out periodically on the 2nd
> leg during periods of silence. The 1st leg is using webrtc encryption and I
> cant decode the stream.
>
> Thanks in advance!
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 14:40:17 -0500
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
> Please provide logs and samples of how you originate this, sounds like
> Voice Activity Detection possibly.
>
> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
> wrote:
>
>> Hi all,
>>
>> Hoping someone can point me in the right direction because after several
>> hours I'm out of ideas.
>>
>> I'm having an issue where the 2nd leg of a call that is bridged to a
>> verto endpoint has broken silence. When there is no sound on the line the
>> audio goes completely silent (all background noise is dropped) but around
>> every third of a second it repeatedly cuts back in for a fraction, then
>> goes completely silent again. This only happens when the call is created
>> using the originate command.
>>
>> Can anyone give me an idea of where to look next? I have a wireshark
>> trace that during playback shows the audio cutting out periodically on the
>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>> and I cant decode the stream.
>>
>> Thanks in advance!
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> Got Bugs? Report them here ! | Reddit:
> /r/freeswitch
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> ---------- Forwarded message ----------
> From: Chris Young
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 1 Sep 2015 07:31:00 +0000
> Subject: [Freeswitch-users] SIP profile not loading
>
> Hi all,
>
>
>
> Recently, we've begun experiencing a strange problem whereby the first SIP
> profile to be loaded gets 'stuck' and never actually completes its
> initialisation. This always seems to affect the first profile only, so if I
> have profiles named (for example):
>
>
>
> dummy.xml
>
> external.xml
>
> internal.xml
>
>
>
> then 'dummy' would fail to load but 'external' and 'internal' would be
> fine. No error messages are output to the logs but preliminary
> investigation suggests that the profile thread is becoming blocked for some
> reason. The specified IP address is valid and available and there are no
> other processes using the requested port. FreeSWITCH comes up successfully
> but 'sofia status' shows only 'external' and 'internal'. At this point, I
> can use 'sofia profile dummy start' and the profile loads correctly so it
> appears to be valid.
>
>
>
> Has anybody else seen this kind of behaviour or know what could be causing
> it?
>
>
>
> Many thanks,
>
> Chris
>
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 01 Sep 2015 12:56:08 +0400
> Subject: [Freeswitch-users] Playing with conditions
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
>
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 10:30:00 +0000
> Subject: Re: [Freeswitch-users] Playing with conditions
> Short answer: Each extension only has 1 set of conditions.
> The condition evaluating foobar is run *before* it gets set.
> After the play_and_get_digits you should transfer/execute_extension to a
> new extension that will evaluate ${foobar}
>
> -Avi Marcus
> BestFone
>
> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
> wrote:
>
>> Hello
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Why inner condition not works?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ssinyagin at gmail.com Tue Sep 1 15:03:25 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 1 Sep 2015 13:03:25 +0200
Subject: [Freeswitch-users] Broken silence with webrtc
In-Reply-To:
References:
Message-ID:
is it running on a virtual machine?
I found a strange effect that I could only reproduce in a VM, and never on
physical hardware:
https://freeswitch.org/jira/browse/FS-7805
under certain load, an originate command triggers a continuous distortion
in another, running and unrelated, channel.
It seems to be triggered by insufficient CPU resource at the moment of the
origination.
On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman wrote:
> I've found that it occurs after any bridge, its not specific to the
> originate command.
>
> Would you require a wireshark trace or the output of the freeswitch
> console?
>
> The scenario below reproduces the issue ...
>
> Test extension
>
>
>
>
>
>
> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>
>
>
>
>
>
>
>
>
> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>
>
>
> I originate a call from a polycom handset using g722 to the extension
> above.
> I originate a call using the verto client to the extension above.
>
> I get the uuids of the channels using show channels and use uuid_bridge
> [uuid1] [uuid2] to merge the channels.
>
> On Tue, Sep 1, 2015 at 11:30 AM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>> Send FreeSWITCH-users mailing list submissions to
>> freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>> freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>> freeswitch-users-owner at lists.freeswitch.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>> Today's Topics:
>>
>> 1. How send in To header anonymous and in P-Asserted-Identity
>> caller-id-number (Alex Polischuk)
>> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
>> 3. Broken silence with webrtc (Gary Foreman)
>> 4. Re: Broken silence with webrtc (Brian West)
>> 5. SIP profile not loading (Chris Young)
>> 6. Playing with conditions (Dmitry Mordovin)
>> 7. Re: Playing with conditions (Avi Marcus)
>>
>>
>> ---------- Forwarded message ----------
>> From: Alex Polischuk
>> To: FreeSWITCH Users Help
>> Cc:
>> Date: Mon, 31 Aug 2015 17:02:25 +0300
>> Subject: [Freeswitch-users] How send in To header anonymous and in
>> P-Asserted-Identity caller-id-number
>> Hi all,
>>
>> How I can define different users and domains in To and P-Asserted-Identity
>> headers?
>>
>> Thanks,
>> Alex
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Dmitry Mordovin
>> To: FreeSWITCH Users Help
>> Cc:
>> Date: Mon, 31 Aug 2015 18:24:08 +0400
>> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>> Hello
>>
>> This example play conf-pin.wav and wait DTMF.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Is it possible to play WAV file in infinity loop and wait user DTMF?
>>
>> And how can I check DTMF input after user entered DTMF?
>>
>> In dialpeer, like.
>> If ${DTMF} = 1 then bridge to XXX
>> If ${DTMF} = 2 then play file and finish session
>>
>>
>>
>> Thank you.
>> Dmitry
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Gary Foreman
>> To: freeswitch-users at lists.freeswitch.org
>> Cc:
>> Date: Mon, 31 Aug 2015 20:36:59 +0100
>> Subject: [Freeswitch-users] Broken silence with webrtc
>> Hi all,
>>
>> Hoping someone can point me in the right direction because after several
>> hours I'm out of ideas.
>>
>> I'm having an issue where the 2nd leg of a call that is bridged to a
>> verto endpoint has broken silence. When there is no sound on the line the
>> audio goes completely silent (all background noise is dropped) but around
>> every third of a second it repeatedly cuts back in for a fraction, then
>> goes completely silent again. This only happens when the call is created
>> using the originate command.
>>
>> Can anyone give me an idea of where to look next? I have a wireshark
>> trace that during playback shows the audio cutting out periodically on the
>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>> and I cant decode the stream.
>>
>> Thanks in advance!
>>
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian West
>> To: FreeSWITCH Users Help
>> Cc:
>> Date: Mon, 31 Aug 2015 14:40:17 -0500
>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>> Please provide logs and samples of how you originate this, sounds like
>> Voice Activity Detection possibly.
>>
>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
>> wrote:
>>
>>> Hi all,
>>>
>>> Hoping someone can point me in the right direction because after several
>>> hours I'm out of ideas.
>>>
>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>> verto endpoint has broken silence. When there is no sound on the line the
>>> audio goes completely silent (all background noise is dropped) but around
>>> every third of a second it repeatedly cuts back in for a fraction, then
>>> goes completely silent again. This only happens when the call is created
>>> using the originate command.
>>>
>>> Can anyone give me an idea of where to look next? I have a wireshark
>>> trace that during playback shows the audio cutting out periodically on the
>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>> and I cant decode the stream.
>>>
>>> Thanks in advance!
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> Got Bugs? Report them here ! | Reddit:
>> /r/freeswitch
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>> ---------- Forwarded message ----------
>> From: Chris Young
>> To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Tue, 1 Sep 2015 07:31:00 +0000
>> Subject: [Freeswitch-users] SIP profile not loading
>>
>> Hi all,
>>
>>
>>
>> Recently, we've begun experiencing a strange problem whereby the first
>> SIP profile to be loaded gets 'stuck' and never actually completes its
>> initialisation. This always seems to affect the first profile only, so if I
>> have profiles named (for example):
>>
>>
>>
>> dummy.xml
>>
>> external.xml
>>
>> internal.xml
>>
>>
>>
>> then 'dummy' would fail to load but 'external' and 'internal' would be
>> fine. No error messages are output to the logs but preliminary
>> investigation suggests that the profile thread is becoming blocked for some
>> reason. The specified IP address is valid and available and there are no
>> other processes using the requested port. FreeSWITCH comes up successfully
>> but 'sofia status' shows only 'external' and 'internal'. At this point, I
>> can use 'sofia profile dummy start' and the profile loads correctly so it
>> appears to be valid.
>>
>>
>>
>> Has anybody else seen this kind of behaviour or know what could be
>> causing it?
>>
>>
>>
>> Many thanks,
>>
>> Chris
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Dmitry Mordovin
>> To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Tue, 01 Sep 2015 12:56:08 +0400
>> Subject: [Freeswitch-users] Playing with conditions
>> Hello
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Why inner condition not works?
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Avi Marcus
>> To: FreeSWITCH Users Help
>> Cc:
>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>> Subject: Re: [Freeswitch-users] Playing with conditions
>> Short answer: Each extension only has 1 set of conditions.
>> The condition evaluating foobar is run *before* it gets set.
>> After the play_and_get_digits you should transfer/execute_extension to a
>> new extension that will evaluate ${foobar}
>>
>> -Avi Marcus
>> BestFone
>>
>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
>> wrote:
>>
>>> Hello
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Why inner condition not works?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gaz.foreman at gmail.com Tue Sep 1 15:12:47 2015
From: gaz.foreman at gmail.com (Gary Foreman)
Date: Tue, 1 Sep 2015 12:12:47 +0100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 3
In-Reply-To:
References:
Message-ID:
I'm getting the issue on a physical server with no load unfortunately. I'm
trying to build the latest FS to test but its proving difficult on CentOS
6, looking like a Debian VM is the next step.
On Tue, Sep 1, 2015 at 12:04 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>
>
> ---------- Forwarded message ----------
> From: Stanislav Sinyagin
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 13:03:25 +0200
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
> is it running on a virtual machine?
>
> I found a strange effect that I could only reproduce in a VM, and never on
> physical hardware:
> https://freeswitch.org/jira/browse/FS-7805
> under certain load, an originate command triggers a continuous distortion
> in another, running and unrelated, channel.
>
> It seems to be triggered by insufficient CPU resource at the moment of the
> origination.
>
>
>
>
>
>
> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman
> wrote:
>
>> I've found that it occurs after any bridge, its not specific to the
>> originate command.
>>
>> Would you require a wireshark trace or the output of the freeswitch
>> console?
>>
>> The scenario below reproduces the issue ...
>>
>> Test extension
>>
>>
>>
>>
>>
>>
>> > data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> > data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>>
>>
>>
>> I originate a call from a polycom handset using g722 to the extension
>> above.
>> I originate a call using the verto client to the extension above.
>>
>> I get the uuids of the channels using show channels and use uuid_bridge
>> [uuid1] [uuid2] to merge the channels.
>>
>> On Tue, Sep 1, 2015 at 11:30 AM, <
>> freeswitch-users-request at lists.freeswitch.org> wrote:
>>
>>> Send FreeSWITCH-users mailing list submissions to
>>> freeswitch-users at lists.freeswitch.org
>>>
>>> To subscribe or unsubscribe via the World Wide Web, visit
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> or, via email, send a message with subject or body 'help' to
>>> freeswitch-users-request at lists.freeswitch.org
>>>
>>> You can reach the person managing the list at
>>> freeswitch-users-owner at lists.freeswitch.org
>>>
>>> When replying, please edit your Subject line so it is more specific
>>> than "Re: Contents of FreeSWITCH-users digest..."
>>>
>>> Today's Topics:
>>>
>>> 1. How send in To header anonymous and in P-Asserted-Identity
>>> caller-id-number (Alex Polischuk)
>>> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
>>> 3. Broken silence with webrtc (Gary Foreman)
>>> 4. Re: Broken silence with webrtc (Brian West)
>>> 5. SIP profile not loading (Chris Young)
>>> 6. Playing with conditions (Dmitry Mordovin)
>>> 7. Re: Playing with conditions (Avi Marcus)
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Alex Polischuk
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 17:02:25 +0300
>>> Subject: [Freeswitch-users] How send in To header anonymous and in
>>> P-Asserted-Identity caller-id-number
>>> Hi all,
>>>
>>> How I can define different users and domains in To and P-Asserted-Identity
>>> headers?
>>>
>>> Thanks,
>>> Alex
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 18:24:08 +0400
>>> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>>> Hello
>>>
>>> This example play conf-pin.wav and wait DTMF.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Is it possible to play WAV file in infinity loop and wait user DTMF?
>>>
>>> And how can I check DTMF input after user entered DTMF?
>>>
>>> In dialpeer, like.
>>> If ${DTMF} = 1 then bridge to XXX
>>> If ${DTMF} = 2 then play file and finish session
>>>
>>>
>>>
>>> Thank you.
>>> Dmitry
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Gary Foreman
>>> To: freeswitch-users at lists.freeswitch.org
>>> Cc:
>>> Date: Mon, 31 Aug 2015 20:36:59 +0100
>>> Subject: [Freeswitch-users] Broken silence with webrtc
>>> Hi all,
>>>
>>> Hoping someone can point me in the right direction because after several
>>> hours I'm out of ideas.
>>>
>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>> verto endpoint has broken silence. When there is no sound on the line the
>>> audio goes completely silent (all background noise is dropped) but around
>>> every third of a second it repeatedly cuts back in for a fraction, then
>>> goes completely silent again. This only happens when the call is created
>>> using the originate command.
>>>
>>> Can anyone give me an idea of where to look next? I have a wireshark
>>> trace that during playback shows the audio cutting out periodically on the
>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>> and I cant decode the stream.
>>>
>>> Thanks in advance!
>>>
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 14:40:17 -0500
>>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>>> Please provide logs and samples of how you originate this, sounds like
>>> Voice Activity Detection possibly.
>>>
>>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
>>> wrote:
>>>
>>>> Hi all,
>>>>
>>>> Hoping someone can point me in the right direction because after
>>>> several hours I'm out of ideas.
>>>>
>>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>>> verto endpoint has broken silence. When there is no sound on the line the
>>>> audio goes completely silent (all background noise is dropped) but around
>>>> every third of a second it repeatedly cuts back in for a fraction, then
>>>> goes completely silent again. This only happens when the call is created
>>>> using the originate command.
>>>>
>>>> Can anyone give me an idea of where to look next? I have a wireshark
>>>> trace that during playback shows the audio cutting out periodically on the
>>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>>> and I cant decode the stream.
>>>>
>>>> Thanks in advance!
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> Got Bugs? Report them here ! | Reddit:
>>> /r/freeswitch
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Chris Young
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 1 Sep 2015 07:31:00 +0000
>>> Subject: [Freeswitch-users] SIP profile not loading
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recently, we've begun experiencing a strange problem whereby the first
>>> SIP profile to be loaded gets 'stuck' and never actually completes its
>>> initialisation. This always seems to affect the first profile only, so if I
>>> have profiles named (for example):
>>>
>>>
>>>
>>> dummy.xml
>>>
>>> external.xml
>>>
>>> internal.xml
>>>
>>>
>>>
>>> then 'dummy' would fail to load but 'external' and 'internal' would be
>>> fine. No error messages are output to the logs but preliminary
>>> investigation suggests that the profile thread is becoming blocked for some
>>> reason. The specified IP address is valid and available and there are no
>>> other processes using the requested port. FreeSWITCH comes up successfully
>>> but 'sofia status' shows only 'external' and 'internal'. At this point, I
>>> can use 'sofia profile dummy start' and the profile loads correctly so it
>>> appears to be valid.
>>>
>>>
>>>
>>> Has anybody else seen this kind of behaviour or know what could be
>>> causing it?
>>>
>>>
>>>
>>> Many thanks,
>>>
>>> Chris
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 01 Sep 2015 12:56:08 +0400
>>> Subject: [Freeswitch-users] Playing with conditions
>>> Hello
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Why inner condition not works?
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Avi Marcus
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>>> Subject: Re: [Freeswitch-users] Playing with conditions
>>> Short answer: Each extension only has 1 set of conditions.
>>> The condition evaluating foobar is run *before* it gets set.
>>> After the play_and_get_digits you should transfer/execute_extension to
>>> a new extension that will evaluate ${foobar}
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
>>> wrote:
>>>
>>>> Hello
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Why inner condition not works?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From italorossib at gmail.com Tue Sep 1 15:53:06 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 1 Sep 2015 08:53:06 -0300
Subject: [Freeswitch-users] SIP profile not loading
In-Reply-To: <373842e4388c4eeb93fde54cfdbaea92@UK-MAIL-001.edge.local>
References: <373842e4388c4eeb93fde54cfdbaea92@UK-MAIL-001.edge.local>
Message-ID:
Which version are you using? Did you enable the debug logs? Can you
pastebin it?
On Tue, Sep 1, 2015 at 4:31 AM, Chris Young
wrote:
> Hi all,
>
>
>
> Recently, we've begun experiencing a strange problem whereby the first SIP
> profile to be loaded gets 'stuck' and never actually completes its
> initialisation. This always seems to affect the first profile only, so if I
> have profiles named (for example):
>
>
>
> dummy.xml
>
> external.xml
>
> internal.xml
>
>
>
> then 'dummy' would fail to load but 'external' and 'internal' would be
> fine. No error messages are output to the logs but preliminary
> investigation suggests that the profile thread is becoming blocked for some
> reason. The specified IP address is valid and available and there are no
> other processes using the requested port. FreeSWITCH comes up successfully
> but 'sofia status' shows only 'external' and 'internal'. At this point, I
> can use 'sofia profile dummy start' and the profile loads correctly so it
> appears to be valid.
>
>
>
> Has anybody else seen this kind of behaviour or know what could be causing
> it?
>
>
>
> Many thanks,
>
> Chris
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From italorossib at gmail.com Tue Sep 1 16:20:30 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 1 Sep 2015 09:20:30 -0300
Subject: [Freeswitch-users] Loop play while wait DTMF digit
In-Reply-To: <55E46388.7080103@dwide.com>
References: <55E46388.7080103@dwide.com>
Message-ID:
Why not use an ivr menu?
On Mon, Aug 31, 2015 at 11:24 AM, Dmitry Mordovin
wrote:
> Hello
>
> This example play conf-pin.wav and wait DTMF.
>
>
>
>
>
>
>
>
>
> Is it possible to play WAV file in infinity loop and wait user DTMF?
>
> And how can I check DTMF input after user entered DTMF?
>
> In dialpeer, like.
> If ${DTMF} = 1 then bridge to XXX
> If ${DTMF} = 2 then play file and finish session
>
>
>
> Thank you.
> Dmitry
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From Chris.Young at enghouse.com Tue Sep 1 16:23:34 2015
From: Chris.Young at enghouse.com (Chris Young)
Date: Tue, 1 Sep 2015 12:23:34 +0000
Subject: [Freeswitch-users] SIP profile not loading
In-Reply-To:
References: <373842e4388c4eeb93fde54cfdbaea92@UK-MAIL-001.edge.local>
Message-ID:
Hi ?talo,
Thanks for your reply.
The version is old (1.2.3) but it's similar to a problem which was reported earlier this year (not by me):
http://lists.freeswitch.org/pipermail/freeswitch-users/2015-April/112373.html
We too are using Windows and our problem only seems to occur on 2012 - the same old version of FreeSWITCH runs correctly on Windows Server 2008 R2, for example.
I enabled the debug logs but they didn't show anything unexpected. I ran the code through a debugger and the thread appears to stop after creating the profile's root object. The log statement, "Creating agent for " is never written the to log file and Process Explorer shows that the thread's context switch count is not increasing. By contrast, the threads for the profiles which do get created successfully increment their counts on a regular basis, as you would expect.
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: 01 September 2015 12:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP profile not loading
Which version are you using? Did you enable the debug logs? Can you pastebin it?
On Tue, Sep 1, 2015 at 4:31 AM, Chris Young > wrote:
Hi all,
Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named (for example):
dummy.xml
external.xml
internal.xml
then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly so it appears to be valid.
Has anybody else seen this kind of behaviour or know what could be causing it?
Many thanks,
Chris
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at 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
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From my.post at hotmail.com Tue Sep 1 16:25:04 2015
From: my.post at hotmail.com (Pavel)
Date: Tue, 1 Sep 2015 18:25:04 +0600
Subject: [Freeswitch-users] SIP headers storage in core or sofia database.
Message-ID:
Hello !
I've set up an ODBC connection for core and sofia profile like this:
Using PgAdmin I see some tables was created in Postgres DB. I want to retrieve all SIP invite headers from this DB. What table should I look for this information if, of course, it is stored at all ?
Thanks.
Regards,Pavel.
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From brian at freeswitch.org Tue Sep 1 17:31:51 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Sep 2015 08:31:51 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 3
In-Reply-To:
References:
Message-ID:
If you're going to respond to the mailing list could you change off the
Digest option?
When you say latest do you mean master? If so then you'll want to read up
on the Debian Jessie install instructions on confluence.
On Tue, Sep 1, 2015 at 6:12 AM, Gary Foreman wrote:
> I'm getting the issue on a physical server with no load unfortunately. I'm
> trying to build the latest FS to test but its proving difficult on CentOS
> 6, looking like a Debian VM is the next step.
>
> On Tue, Sep 1, 2015 at 12:04 PM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>> Send FreeSWITCH-users mailing list submissions to
>> freeswitch-users at lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>> freeswitch-users-request at lists.freeswitch.org
>>
>> You can reach the person managing the list at
>> freeswitch-users-owner at lists.freeswitch.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>> Today's Topics:
>>
>> 1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>>
>>
>> ---------- Forwarded message ----------
>> From: Stanislav Sinyagin
>> To: FreeSWITCH Users Help
>> Cc:
>> Date: Tue, 1 Sep 2015 13:03:25 +0200
>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>> is it running on a virtual machine?
>>
>> I found a strange effect that I could only reproduce in a VM, and never
>> on physical hardware:
>> https://freeswitch.org/jira/browse/FS-7805
>> under certain load, an originate command triggers a continuous distortion
>> in another, running and unrelated, channel.
>>
>> It seems to be triggered by insufficient CPU resource at the moment of
>> the origination.
>>
>>
>>
>>
>>
>>
>> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman
>> wrote:
>>
>>> I've found that it occurs after any bridge, its not specific to the
>>> originate command.
>>>
>>> Would you require a wireshark trace or the output of the freeswitch
>>> console?
>>>
>>> The scenario below reproduces the issue ...
>>>
>>> Test extension
>>>
>>>
>>>
>>>
>>>
>>>
>>> >> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> >> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>>>
>>>
>>>
>>> I originate a call from a polycom handset using g722 to the extension
>>> above.
>>> I originate a call using the verto client to the extension above.
>>>
>>> I get the uuids of the channels using show channels and use uuid_bridge
>>> [uuid1] [uuid2] to merge the channels.
>>>
>>> On Tue, Sep 1, 2015 at 11:30 AM, <
>>> freeswitch-users-request at lists.freeswitch.org> wrote:
>>>
>>>> Send FreeSWITCH-users mailing list submissions to
>>>> freeswitch-users at lists.freeswitch.org
>>>>
>>>> To subscribe or unsubscribe via the World Wide Web, visit
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> or, via email, send a message with subject or body 'help' to
>>>> freeswitch-users-request at lists.freeswitch.org
>>>>
>>>> You can reach the person managing the list at
>>>> freeswitch-users-owner at lists.freeswitch.org
>>>>
>>>> When replying, please edit your Subject line so it is more specific
>>>> than "Re: Contents of FreeSWITCH-users digest..."
>>>>
>>>> Today's Topics:
>>>>
>>>> 1. How send in To header anonymous and in P-Asserted-Identity
>>>> caller-id-number (Alex Polischuk)
>>>> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
>>>> 3. Broken silence with webrtc (Gary Foreman)
>>>> 4. Re: Broken silence with webrtc (Brian West)
>>>> 5. SIP profile not loading (Chris Young)
>>>> 6. Playing with conditions (Dmitry Mordovin)
>>>> 7. Re: Playing with conditions (Avi Marcus)
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Alex Polischuk
>>>> To: FreeSWITCH Users Help
>>>> Cc:
>>>> Date: Mon, 31 Aug 2015 17:02:25 +0300
>>>> Subject: [Freeswitch-users] How send in To header anonymous and in
>>>> P-Asserted-Identity caller-id-number
>>>> Hi all,
>>>>
>>>> How I can define different users and domains in To and P-Asserted-Identity
>>>> headers?
>>>>
>>>> Thanks,
>>>> Alex
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Dmitry Mordovin
>>>> To: FreeSWITCH Users Help
>>>> Cc:
>>>> Date: Mon, 31 Aug 2015 18:24:08 +0400
>>>> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>>>> Hello
>>>>
>>>> This example play conf-pin.wav and wait DTMF.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Is it possible to play WAV file in infinity loop and wait user DTMF?
>>>>
>>>> And how can I check DTMF input after user entered DTMF?
>>>>
>>>> In dialpeer, like.
>>>> If ${DTMF} = 1 then bridge to XXX
>>>> If ${DTMF} = 2 then play file and finish session
>>>>
>>>>
>>>>
>>>> Thank you.
>>>> Dmitry
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Gary Foreman
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Cc:
>>>> Date: Mon, 31 Aug 2015 20:36:59 +0100
>>>> Subject: [Freeswitch-users] Broken silence with webrtc
>>>> Hi all,
>>>>
>>>> Hoping someone can point me in the right direction because after
>>>> several hours I'm out of ideas.
>>>>
>>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>>> verto endpoint has broken silence. When there is no sound on the line the
>>>> audio goes completely silent (all background noise is dropped) but around
>>>> every third of a second it repeatedly cuts back in for a fraction, then
>>>> goes completely silent again. This only happens when the call is created
>>>> using the originate command.
>>>>
>>>> Can anyone give me an idea of where to look next? I have a wireshark
>>>> trace that during playback shows the audio cutting out periodically on the
>>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>>> and I cant decode the stream.
>>>>
>>>> Thanks in advance!
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Brian West
>>>> To: FreeSWITCH Users Help
>>>> Cc:
>>>> Date: Mon, 31 Aug 2015 14:40:17 -0500
>>>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>>>> Please provide logs and samples of how you originate this, sounds like
>>>> Voice Activity Detection possibly.
>>>>
>>>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
>>>> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> Hoping someone can point me in the right direction because after
>>>>> several hours I'm out of ideas.
>>>>>
>>>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>>>> verto endpoint has broken silence. When there is no sound on the line the
>>>>> audio goes completely silent (all background noise is dropped) but around
>>>>> every third of a second it repeatedly cuts back in for a fraction, then
>>>>> goes completely silent again. This only happens when the call is created
>>>>> using the originate command.
>>>>>
>>>>> Can anyone give me an idea of where to look next? I have a wireshark
>>>>> trace that during playback shows the audio cutting out periodically on the
>>>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>>>> and I cant decode the stream.
>>>>>
>>>>> Thanks in advance!
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> Got Bugs? Report them here ! | Reddit:
>>>> /r/freeswitch
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Chris Young
>>>> To: "freeswitch-users at lists.freeswitch.org" <
>>>> freeswitch-users at lists.freeswitch.org>
>>>> Cc:
>>>> Date: Tue, 1 Sep 2015 07:31:00 +0000
>>>> Subject: [Freeswitch-users] SIP profile not loading
>>>>
>>>> Hi all,
>>>>
>>>>
>>>>
>>>> Recently, we've begun experiencing a strange problem whereby the first
>>>> SIP profile to be loaded gets 'stuck' and never actually completes its
>>>> initialisation. This always seems to affect the first profile only, so if I
>>>> have profiles named (for example):
>>>>
>>>>
>>>>
>>>> dummy.xml
>>>>
>>>> external.xml
>>>>
>>>> internal.xml
>>>>
>>>>
>>>>
>>>> then 'dummy' would fail to load but 'external' and 'internal' would be
>>>> fine. No error messages are output to the logs but preliminary
>>>> investigation suggests that the profile thread is becoming blocked for some
>>>> reason. The specified IP address is valid and available and there are no
>>>> other processes using the requested port. FreeSWITCH comes up successfully
>>>> but 'sofia status' shows only 'external' and 'internal'. At this point, I
>>>> can use 'sofia profile dummy start' and the profile loads correctly so it
>>>> appears to be valid.
>>>>
>>>>
>>>>
>>>> Has anybody else seen this kind of behaviour or know what could be
>>>> causing it?
>>>>
>>>>
>>>>
>>>> Many thanks,
>>>>
>>>> Chris
>>>>
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Dmitry Mordovin
>>>> To: "freeswitch-users at lists.freeswitch.org" <
>>>> freeswitch-users at lists.freeswitch.org>
>>>> Cc:
>>>> Date: Tue, 01 Sep 2015 12:56:08 +0400
>>>> Subject: [Freeswitch-users] Playing with conditions
>>>> Hello
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Why inner condition not works?
>>>>
>>>>
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Avi Marcus
>>>> To: FreeSWITCH Users Help
>>>> Cc:
>>>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>>>> Subject: Re: [Freeswitch-users] Playing with conditions
>>>> Short answer: Each extension only has 1 set of conditions.
>>>> The condition evaluating foobar is run *before* it gets set.
>>>> After the play_and_get_digits you should transfer/execute_extension to
>>>> a new extension that will evaluate ${foobar}
>>>>
>>>> -Avi Marcus
>>>> BestFone
>>>>
>>>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
>>>> wrote:
>>>>
>>>>> Hello
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Why inner condition not works?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Got Bugs? Report them here ! | Reddit:
/r/freeswitch
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org Tue Sep 1 17:32:42 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Sep 2015 08:32:42 -0500
Subject: [Freeswitch-users] Playing with conditions
In-Reply-To: <0000014f8874ccc5-e8464c2a-7efa-4711-a201-5efec14fe0f2-000000@email.amazonses.com>
References: <55E56828.102@dwide.com>
<0000014f8874ccc5-e8464c2a-7efa-4711-a201-5efec14fe0f2-000000@email.amazonses.com>
Message-ID:
https://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Condition_Caveats_and_Notes
This isn't moved over yet, or I can't find it on confluence.
On Tue, Sep 1, 2015 at 5:30 AM, Avi Marcus wrote:
> Short answer: Each extension only has 1 set of conditions.
> The condition evaluating foobar is run *before* it gets set.
> After the play_and_get_digits you should transfer/execute_extension to a
> new extension that will evaluate ${foobar}
>
> -Avi Marcus
> BestFone
>
> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
> wrote:
>
>> Hello
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Why inner condition not works?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Got Bugs? Report them here ! | Reddit:
/r/freeswitch
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org Tue Sep 1 17:39:09 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Sep 2015 08:39:09 -0500
Subject: [Freeswitch-users] SIP headers storage in core or sofia
database.
In-Reply-To:
References:
Message-ID:
Its not stored. What problem are you trying to solve?
On Tue, Sep 1, 2015 at 7:25 AM, Pavel wrote:
> Hello !
>
> I've set up an ODBC connection for core and sofia profile like this:
>
>
>
>
> Using PgAdmin I see some tables was created in Postgres DB. I want to
> retrieve all SIP invite headers from this DB. What table should I look for
> this information if, of course, it is stored at all ?
>
> Thanks.
>
> Regards,
> Pavel.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
Got Bugs? Report them here ! | Reddit:
/r/freeswitch
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From d.mordovin at dwide.com Tue Sep 1 17:40:42 2015
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Tue, 01 Sep 2015 17:40:42 +0400
Subject: [Freeswitch-users] Playing with conditions
In-Reply-To:
References: <55E56828.102@dwide.com> <0000014f8874ccc5-e8464c2a-7efa-4711-a201-5efec14fe0f2-000000@email.amazonses.com>
Message-ID: <55E5AADA.5020609@dwide.com>
Great! Thanks all!
On 09/01/2015 05:32 PM, Brian West wrote:
> https://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Condition_Caveats_and_Notes
>
> This isn't moved over yet, or I can't find it on confluence.
>
> On Tue, Sep 1, 2015 at 5:30 AM, Avi Marcus > wrote:
>
> Short answer: Each extension only has 1 set of conditions.
> The condition evaluating foobar is run /before/ it gets set.
> After the play_and_get_digits you should
> transfer/execute_extension to a new extension that will evaluate
> ${foobar}
>
> -Avi Marcus
> BestFone
>
> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
> > wrote:
>
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> Got Bugs? Report them here ! | Reddit:
> /r/freeswitch
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brian at freeswitch.org Tue Sep 1 17:40:31 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Sep 2015 08:40:31 -0500
Subject: [Freeswitch-users] SIP profile not loading
In-Reply-To:
References: <373842e4388c4eeb93fde54cfdbaea92@UK-MAIL-001.edge.local>
Message-ID:
What ever the problem is it won't be fixed in 1.2.x, you should see if the
issue is gone in 1.4.21, are you able to build your own with MSVC on
Windows?
On Tue, Sep 1, 2015 at 7:23 AM, Chris Young
wrote:
> Hi ?talo,
>
>
>
> Thanks for your reply.
>
>
>
> The version is old (1.2.3) but it's similar to a problem which was
> reported earlier this year (not by me):
>
>
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-April/112373.html
>
>
>
> We too are using Windows and our problem only seems to occur on 2012 - the
> same old version of FreeSWITCH runs correctly on Windows Server 2008 R2,
> for example.
>
>
>
> I enabled the debug logs but they didn't show anything unexpected. I ran
> the code through a debugger and the thread appears to stop after creating
> the profile's root object. The log statement, "Creating agent for
> " is never written the to log file and Process Explorer shows that
> the thread's context switch count is not increasing. By contrast, the
> threads for the profiles which do get created successfully increment their
> counts on a regular basis, as you would expect.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *?talo Rossi
> *Sent:* 01 September 2015 12:53
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP profile not loading
>
>
>
> Which version are you using? Did you enable the debug logs? Can you
> pastebin it?
>
>
>
> On Tue, Sep 1, 2015 at 4:31 AM, Chris Young
> wrote:
>
> Hi all,
>
>
>
> Recently, we've begun experiencing a strange problem whereby the first SIP
> profile to be loaded gets 'stuck' and never actually completes its
> initialisation. This always seems to affect the first profile only, so if I
> have profiles named (for example):
>
>
>
> dummy.xml
>
> external.xml
>
> internal.xml
>
>
>
> then 'dummy' would fail to load but 'external' and 'internal' would be
> fine. No error messages are output to the logs but preliminary
> investigation suggests that the profile thread is becoming blocked for some
> reason. The specified IP address is valid and available and there are no
> other processes using the requested port. FreeSWITCH comes up successfully
> but 'sofia status' shows only 'external' and 'internal'. At this point, I
> can use 'sofia profile dummy start' and the profile loads correctly so it
> appears to be valid.
>
>
>
> Has anybody else seen this kind of behaviour or know what could be causing
> it?
>
>
>
> Many thanks,
>
> Chris
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
Got Bugs? Report them here ! | Reddit:
/r/freeswitch
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From d.mordovin at dwide.com Tue Sep 1 17:45:55 2015
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Tue, 01 Sep 2015 17:45:55 +0400
Subject: [Freeswitch-users] Delay and exec after answer
Message-ID: <55E5AC13.8080708@dwide.com>
Hello
Is it possible to execute some api with delay after answer received?
Trying
api_on_answer_1='sleep 5000',api_on_answer_2='log -------delayed 5sec
after answer------'
but 'api_on_answer_2' executed at once
Any ideas?
From my.post at hotmail.com Tue Sep 1 18:39:07 2015
From: my.post at hotmail.com (Pavel)
Date: Tue, 1 Sep 2015 20:39:07 +0600
Subject: [Freeswitch-users] SIP headers storage in core or sofia
database.
In-Reply-To:
References:
Message-ID:
Brian, thanks a lot for your reply. To put it short - I am willing to do some call visualization with simple web application based on some incoming INVITE sip header values.So I've been thinking, that watching for changes to sofia (or core) db is enough to represent my data.The ESL is probably a way to go, right ?Regards,Pavel.
From: my.post at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: SIP headers storage in core or sofia database.
Date: Tue, 1 Sep 2015 18:25:04 +0600
Hello !
I've set up an ODBC connection for core and sofia profile like this:
Using PgAdmin I see some tables was created in Postgres DB. I want to retrieve all SIP invite headers from this DB. What table should I look for this information if, of course, it is stored at all ?
Thanks.
Regards,Pavel.
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From mike at jerris.com Tue Sep 1 18:40:25 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 1 Sep 2015 10:40:25 -0400
Subject: [Freeswitch-users] Delay and exec after answer
In-Reply-To: <55E5AC13.8080708@dwide.com>
References: <55E5AC13.8080708@dwide.com>
Message-ID:
you could use api on answer to schedule an api command
On Tuesday, September 1, 2015, Dmitry Mordovin wrote:
> Hello
>
> Is it possible to execute some api with delay after answer received?
>
> Trying
>
> api_on_answer_1='sleep 5000',api_on_answer_2='log -------delayed 5sec
> after answer------'
>
>
> but 'api_on_answer_2' executed at once
>
> Any ideas?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From d.mordovin at dwide.com Tue Sep 1 18:59:36 2015
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Tue, 01 Sep 2015 18:59:36 +0400
Subject: [Freeswitch-users] Delay and exec after answer
In-Reply-To:
References: <55E5AC13.8080708@dwide.com>
Message-ID: <55E5BD58.2010000@dwide.com>
api_on_answer_1='sched_api +10 none send_dtmf 1234 at 100'
In Console
[DEBUG] switch_event.c:1698 Parsing variable
[api_on_answer_1]=[sched_api +10 none send_dtmf 1234 at 100]
Looks good,
...
But in 10 seconds:
[DEBUG] mod_commands.c:4575 Command send_dtmf(1234 at 100):
INVALID COMMAND!
What is wrong??
On 09/01/2015 06:40 PM, Michael Jerris wrote:
> you could use api on answer to schedule an api command
>
> On Tuesday, September 1, 2015, Dmitry Mordovin > wrote:
>
> Hello
>
> Is it possible to execute some api with delay after answer received?
>
> Trying
>
> api_on_answer_1='sleep 5000',api_on_answer_2='log -------delayed 5sec
> after answer------'
>
>
> but 'api_on_answer_2' executed at once
>
> Any ideas?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gaz.foreman at gmail.com Tue Sep 1 19:13:18 2015
From: gaz.foreman at gmail.com (Gary Foreman)
Date: Tue, 1 Sep 2015 16:13:18 +0100
Subject: [Freeswitch-users] Broken silence with webrtc
Message-ID:
Ok so the issue has been superseded by intermittent one-way / no audio. I'm
getting it very intermittently (1 in every 30 calls or so) but I'm
struggling to debug it as the traffic is encrypted and wireshark doesn't
see it as rtp stream.
Where is the best place to start debugging verto? I was previously using
sip.js without any audio issues so it seems to be verto specific.
On Tue, Sep 1, 2015 at 12:04 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>
>
> ---------- Forwarded message ----------
> From: Stanislav Sinyagin
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 13:03:25 +0200
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
> is it running on a virtual machine?
>
> I found a strange effect that I could only reproduce in a VM, and never on
> physical hardware:
> https://freeswitch.org/jira/browse/FS-7805
> under certain load, an originate command triggers a continuous distortion
> in another, running and unrelated, channel.
>
> It seems to be triggered by insufficient CPU resource at the moment of the
> origination.
>
>
>
>
>
>
> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman
> wrote:
>
>> I've found that it occurs after any bridge, its not specific to the
>> originate command.
>>
>> Would you require a wireshark trace or the output of the freeswitch
>> console?
>>
>> The scenario below reproduces the issue ...
>>
>> Test extension
>>
>>
>>
>>
>>
>>
>> > data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> > data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>>
>>
>>
>> I originate a call from a polycom handset using g722 to the extension
>> above.
>> I originate a call using the verto client to the extension above.
>>
>> I get the uuids of the channels using show channels and use uuid_bridge
>> [uuid1] [uuid2] to merge the channels.
>>
>> On Tue, Sep 1, 2015 at 11:30 AM, <
>> freeswitch-users-request at lists.freeswitch.org> wrote:
>>
>>> Send FreeSWITCH-users mailing list submissions to
>>> freeswitch-users at lists.freeswitch.org
>>>
>>> To subscribe or unsubscribe via the World Wide Web, visit
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> or, via email, send a message with subject or body 'help' to
>>> freeswitch-users-request at lists.freeswitch.org
>>>
>>> You can reach the person managing the list at
>>> freeswitch-users-owner at lists.freeswitch.org
>>>
>>> When replying, please edit your Subject line so it is more specific
>>> than "Re: Contents of FreeSWITCH-users digest..."
>>>
>>> Today's Topics:
>>>
>>> 1. How send in To header anonymous and in P-Asserted-Identity
>>> caller-id-number (Alex Polischuk)
>>> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
>>> 3. Broken silence with webrtc (Gary Foreman)
>>> 4. Re: Broken silence with webrtc (Brian West)
>>> 5. SIP profile not loading (Chris Young)
>>> 6. Playing with conditions (Dmitry Mordovin)
>>> 7. Re: Playing with conditions (Avi Marcus)
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Alex Polischuk
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 17:02:25 +0300
>>> Subject: [Freeswitch-users] How send in To header anonymous and in
>>> P-Asserted-Identity caller-id-number
>>> Hi all,
>>>
>>> How I can define different users and domains in To and P-Asserted-Identity
>>> headers?
>>>
>>> Thanks,
>>> Alex
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 18:24:08 +0400
>>> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>>> Hello
>>>
>>> This example play conf-pin.wav and wait DTMF.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Is it possible to play WAV file in infinity loop and wait user DTMF?
>>>
>>> And how can I check DTMF input after user entered DTMF?
>>>
>>> In dialpeer, like.
>>> If ${DTMF} = 1 then bridge to XXX
>>> If ${DTMF} = 2 then play file and finish session
>>>
>>>
>>>
>>> Thank you.
>>> Dmitry
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Gary Foreman
>>> To: freeswitch-users at lists.freeswitch.org
>>> Cc:
>>> Date: Mon, 31 Aug 2015 20:36:59 +0100
>>> Subject: [Freeswitch-users] Broken silence with webrtc
>>> Hi all,
>>>
>>> Hoping someone can point me in the right direction because after several
>>> hours I'm out of ideas.
>>>
>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>> verto endpoint has broken silence. When there is no sound on the line the
>>> audio goes completely silent (all background noise is dropped) but around
>>> every third of a second it repeatedly cuts back in for a fraction, then
>>> goes completely silent again. This only happens when the call is created
>>> using the originate command.
>>>
>>> Can anyone give me an idea of where to look next? I have a wireshark
>>> trace that during playback shows the audio cutting out periodically on the
>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>> and I cant decode the stream.
>>>
>>> Thanks in advance!
>>>
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Mon, 31 Aug 2015 14:40:17 -0500
>>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>>> Please provide logs and samples of how you originate this, sounds like
>>> Voice Activity Detection possibly.
>>>
>>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
>>> wrote:
>>>
>>>> Hi all,
>>>>
>>>> Hoping someone can point me in the right direction because after
>>>> several hours I'm out of ideas.
>>>>
>>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>>> verto endpoint has broken silence. When there is no sound on the line the
>>>> audio goes completely silent (all background noise is dropped) but around
>>>> every third of a second it repeatedly cuts back in for a fraction, then
>>>> goes completely silent again. This only happens when the call is created
>>>> using the originate command.
>>>>
>>>> Can anyone give me an idea of where to look next? I have a wireshark
>>>> trace that during playback shows the audio cutting out periodically on the
>>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>>> and I cant decode the stream.
>>>>
>>>> Thanks in advance!
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> Got Bugs? Report them here ! | Reddit:
>>> /r/freeswitch
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Chris Young
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 1 Sep 2015 07:31:00 +0000
>>> Subject: [Freeswitch-users] SIP profile not loading
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recently, we've begun experiencing a strange problem whereby the first
>>> SIP profile to be loaded gets 'stuck' and never actually completes its
>>> initialisation. This always seems to affect the first profile only, so if I
>>> have profiles named (for example):
>>>
>>>
>>>
>>> dummy.xml
>>>
>>> external.xml
>>>
>>> internal.xml
>>>
>>>
>>>
>>> then 'dummy' would fail to load but 'external' and 'internal' would be
>>> fine. No error messages are output to the logs but preliminary
>>> investigation suggests that the profile thread is becoming blocked for some
>>> reason. The specified IP address is valid and available and there are no
>>> other processes using the requested port. FreeSWITCH comes up successfully
>>> but 'sofia status' shows only 'external' and 'internal'. At this point, I
>>> can use 'sofia profile dummy start' and the profile loads correctly so it
>>> appears to be valid.
>>>
>>>
>>>
>>> Has anybody else seen this kind of behaviour or know what could be
>>> causing it?
>>>
>>>
>>>
>>> Many thanks,
>>>
>>> Chris
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 01 Sep 2015 12:56:08 +0400
>>> Subject: [Freeswitch-users] Playing with conditions
>>> Hello
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Why inner condition not works?
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Avi Marcus
>>> To: FreeSWITCH Users Help
>>> Cc:
>>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>>> Subject: Re: [Freeswitch-users] Playing with conditions
>>> Short answer: Each extension only has 1 set of conditions.
>>> The condition evaluating foobar is run *before* it gets set.
>>> After the play_and_get_digits you should transfer/execute_extension to
>>> a new extension that will evaluate ${foobar}
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
>>> wrote:
>>>
>>>> Hello
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Why inner condition not works?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org Tue Sep 1 19:25:10 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 1 Sep 2015 10:25:10 -0500
Subject: [Freeswitch-users] Broken silence with webrtc
In-Reply-To:
References:
Message-ID: <186c01d0e4ca$648d8f70$2da8ae50$@freeswitch.org>
Just enable verto debugging in verto.conf.xml in your configs? it?ll print it right to the screen
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gary Foreman
Sent: Tuesday, September 1, 2015 10:13 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Broken silence with webrtc
Ok so the issue has been superseded by intermittent one-way / no audio. I'm getting it very intermittently (1 in every 30 calls or so) but I'm struggling to debug it as the traffic is encrypted and wireshark doesn't see it as rtp stream.
Where is the best place to start debugging verto? I was previously using sip.js without any audio issues so it seems to be verto specific.
On Tue, Sep 1, 2015 at 12:04 PM, > wrote:
Send FreeSWITCH-users mailing list submissions to
freeswitch-users at lists.freeswitch.org
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request at lists.freeswitch.org
You can reach the person managing the list at
freeswitch-users-owner at lists.freeswitch.org
When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:
1. Re: Broken silence with webrtc (Stanislav Sinyagin)
---------- Forwarded message ----------
From: Stanislav Sinyagin >
To: FreeSWITCH Users Help >
Cc:
Date: Tue, 1 Sep 2015 13:03:25 +0200
Subject: Re: [Freeswitch-users] Broken silence with webrtc
is it running on a virtual machine?
I found a strange effect that I could only reproduce in a VM, and never on physical hardware:
https://freeswitch.org/jira/browse/FS-7805
under certain load, an originate command triggers a continuous distortion in another, running and unrelated, channel.
It seems to be triggered by insufficient CPU resource at the moment of the origination.
On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman > wrote:
I've found that it occurs after any bridge, its not specific to the originate command.
Would you require a wireshark trace or the output of the freeswitch console?
The scenario below reproduces the issue ...
Test extension
I originate a call from a polycom handset using g722 to the extension above.
I originate a call using the verto client to the extension above.
I get the uuids of the channels using show channels and use uuid_bridge [uuid1] [uuid2] to merge the channels.
On Tue, Sep 1, 2015 at 11:30 AM, > wrote:
Send FreeSWITCH-users mailing list submissions to
freeswitch-users at lists.freeswitch.org
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request at lists.freeswitch.org
You can reach the person managing the list at
freeswitch-users-owner at lists.freeswitch.org
When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:
1. How send in To header anonymous and in P-Asserted-Identity
caller-id-number (Alex Polischuk)
2. Loop play while wait DTMF digit (Dmitry Mordovin)
3. Broken silence with webrtc (Gary Foreman)
4. Re: Broken silence with webrtc (Brian West)
5. SIP profile not loading (Chris Young)
6. Playing with conditions (Dmitry Mordovin)
7. Re: Playing with conditions (Avi Marcus)
---------- Forwarded message ----------
From: Alex Polischuk >
To: FreeSWITCH Users Help >
Cc:
Date: Mon, 31 Aug 2015 17:02:25 +0300
Subject: [Freeswitch-users] How send in To header anonymous and in P-Asserted-Identity caller-id-number
Hi all,
How I can define different users and domains in To and P-Asserted-Identity headers?
Thanks,
Alex
---------- Forwarded message ----------
From: Dmitry Mordovin >
To: FreeSWITCH Users Help >
Cc:
Date: Mon, 31 Aug 2015 18:24:08 +0400
Subject: [Freeswitch-users] Loop play while wait DTMF digit
Hello
This example play conf-pin.wav and wait DTMF.
Is it possible to play WAV file in infinity loop and wait user DTMF?
And how can I check DTMF input after user entered DTMF?
In dialpeer, like.
If ${DTMF} = 1 then bridge to XXX
If ${DTMF} = 2 then play file and finish session
Thank you.
Dmitry
---------- Forwarded message ----------
From: Gary Foreman >
To: freeswitch-users at lists.freeswitch.org
Cc:
Date: Mon, 31 Aug 2015 20:36:59 +0100
Subject: [Freeswitch-users] Broken silence with webrtc
Hi all,
Hoping someone can point me in the right direction because after several hours I'm out of ideas.
I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command.
Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream.
Thanks in advance!
---------- Forwarded message ----------
From: Brian West >
To: FreeSWITCH Users Help >
Cc:
Date: Mon, 31 Aug 2015 14:40:17 -0500
Subject: Re: [Freeswitch-users] Broken silence with webrtc
Please provide logs and samples of how you originate this, sounds like Voice Activity Detection possibly.
On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman > wrote:
Hi all,
Hoping someone can point me in the right direction because after several hours I'm out of ideas.
I'm having an issue where the 2nd leg of a call that is bridged to a verto endpoint has broken silence. When there is no sound on the line the audio goes completely silent (all background noise is dropped) but around every third of a second it repeatedly cuts back in for a fraction, then goes completely silent again. This only happens when the call is created using the originate command.
Can anyone give me an idea of where to look next? I have a wireshark trace that during playback shows the audio cutting out periodically on the 2nd leg during periods of silence. The 1st leg is using webrtc encryption and I cant decode the stream.
Thanks in advance!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at 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
Got Bugs? Report them here ! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
---------- Forwarded message ----------
From: Chris Young >
To: "freeswitch-users at lists.freeswitch.org " >
Cc:
Date: Tue, 1 Sep 2015 07:31:00 +0000
Subject: [Freeswitch-users] SIP profile not loading
Hi all,
Recently, we've begun experiencing a strange problem whereby the first SIP profile to be loaded gets 'stuck' and never actually completes its initialisation. This always seems to affect the first profile only, so if I have profiles named (for example):
dummy.xml
external.xml
internal.xml
then 'dummy' would fail to load but 'external' and 'internal' would be fine. No error messages are output to the logs but preliminary investigation suggests that the profile thread is becoming blocked for some reason. The specified IP address is valid and available and there are no other processes using the requested port. FreeSWITCH comes up successfully but 'sofia status' shows only 'external' and 'internal'. At this point, I can use 'sofia profile dummy start' and the profile loads correctly so it appears to be valid.
Has anybody else seen this kind of behaviour or know what could be causing it?
Many thanks,
Chris
---------- Forwarded message ----------
From: Dmitry Mordovin >
To: "freeswitch-users at lists.freeswitch.org " >
Cc:
Date: Tue, 01 Sep 2015 12:56:08 +0400
Subject: [Freeswitch-users] Playing with conditions
Hello
Why inner condition not works?
---------- Forwarded message ----------
From: Avi Marcus >
To: FreeSWITCH Users Help >
Cc:
Date: Tue, 1 Sep 2015 10:30:00 +0000
Subject: Re: [Freeswitch-users] Playing with conditions
Short answer: Each extension only has 1 set of conditions.
The condition evaluating foobar is run before it gets set.
After the play_and_get_digits you should transfer/execute_extension to a new extension that will evaluate ${foobar}
-Avi Marcus
BestFone
On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin > wrote:
Hello
Why inner condition not works?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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-------------- next part --------------
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From anthony.minessale at gmail.com Tue Sep 1 19:35:30 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 1 Sep 2015 10:35:30 -0500
Subject: [Freeswitch-users] Broken silence with webrtc
In-Reply-To: <186c01d0e4ca$648d8f70$2da8ae50$@freeswitch.org>
References:
<186c01d0e4ca$648d8f70$2da8ae50$@freeswitch.org>
Message-ID:
Your findings contradict each other so much, I recommend you start over
from scratch.
Backup your configs. update to the latest master version of FS. If you
are on 1.4, nothing new will be done to mitigate webrtc issues. Set up a
box with the default configurations and retest.
you may also want to try putting in vars.xml
On Tue, Sep 1, 2015 at 10:25 AM, Ken Rice wrote:
> Just enable verto debugging in verto.conf.xml in your configs? it?ll print
> it right to the screen
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gary Foreman
> *Sent:* Tuesday, September 1, 2015 10:13 AM
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Subject:* Re: [Freeswitch-users] Broken silence with webrtc
>
>
>
> Ok so the issue has been superseded by intermittent one-way / no audio.
> I'm getting it very intermittently (1 in every 30 calls or so) but I'm
> struggling to debug it as the traffic is encrypted and wireshark doesn't
> see it as rtp stream.
>
>
>
> Where is the best place to start debugging verto? I was previously using
> sip.js without any audio issues so it seems to be verto specific.
>
>
>
> On Tue, Sep 1, 2015 at 12:04 PM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>
>
> ---------- Forwarded message ----------
> From: Stanislav Sinyagin
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 13:03:25 +0200
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> is it running on a virtual machine?
>
> I found a strange effect that I could only reproduce in a VM, and never on
> physical hardware:
> https://freeswitch.org/jira/browse/FS-7805
>
> under certain load, an originate command triggers a continuous distortion
> in another, running and unrelated, channel.
>
> It seems to be triggered by insufficient CPU resource at the moment of the
> origination.
>
>
>
>
>
>
>
> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman
> wrote:
>
> I've found that it occurs after any bridge, its not specific to the
> originate command.
>
>
>
> Would you require a wireshark trace or the output of the freeswitch
> console?
>
>
>
> The scenario below reproduces the issue ...
>
>
>
> Test extension
>
>
>
>
>
>
>
>
>
>
>
>
>
> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>
>
>
>
>
>
>
> I originate a call from a polycom handset using g722 to the extension
> above.
>
> I originate a call using the verto client to the extension above.
>
>
>
> I get the uuids of the channels using show channels and use uuid_bridge
> [uuid1] [uuid2] to merge the channels.
>
>
>
> On Tue, Sep 1, 2015 at 11:30 AM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. How send in To header anonymous and in P-Asserted-Identity
> caller-id-number (Alex Polischuk)
> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
> 3. Broken silence with webrtc (Gary Foreman)
> 4. Re: Broken silence with webrtc (Brian West)
> 5. SIP profile not loading (Chris Young)
> 6. Playing with conditions (Dmitry Mordovin)
> 7. Re: Playing with conditions (Avi Marcus)
>
>
> ---------- Forwarded message ----------
> From: Alex Polischuk
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 17:02:25 +0300
> Subject: [Freeswitch-users] How send in To header anonymous and in
> P-Asserted-Identity caller-id-number
>
> Hi all,
>
>
>
> How I can define different users and domains in To and P-Asserted-Identity
> headers?
>
>
>
> Thanks,
>
> Alex
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 18:24:08 +0400
> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>
> Hello
>
> This example play conf-pin.wav and wait DTMF.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Is it possible to play WAV file in infinity loop and wait user DTMF?
>
> And how can I check DTMF input after user entered DTMF?
>
> In dialpeer, like.
> If ${DTMF} = 1 then bridge to XXX
> If ${DTMF} = 2 then play file and finish session
>
>
>
> Thank you.
> Dmitry
>
>
>
> ---------- Forwarded message ----------
> From: Gary Foreman
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Mon, 31 Aug 2015 20:36:59 +0100
> Subject: [Freeswitch-users] Broken silence with webrtc
>
> Hi all,
>
>
>
> Hoping someone can point me in the right direction because after several
> hours I'm out of ideas.
>
>
>
> I'm having an issue where the 2nd leg of a call that is bridged to a verto
> endpoint has broken silence. When there is no sound on the line the audio
> goes completely silent (all background noise is dropped) but around every
> third of a second it repeatedly cuts back in for a fraction, then goes
> completely silent again. This only happens when the call is created using
> the originate command.
>
>
>
> Can anyone give me an idea of where to look next? I have a wireshark trace
> that during playback shows the audio cutting out periodically on the 2nd
> leg during periods of silence. The 1st leg is using webrtc encryption and I
> cant decode the stream.
>
>
>
> Thanks in advance!
>
>
>
>
>
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 14:40:17 -0500
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> Please provide logs and samples of how you originate this, sounds like
> Voice Activity Detection possibly.
>
>
>
> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
> wrote:
>
> Hi all,
>
>
>
> Hoping someone can point me in the right direction because after several
> hours I'm out of ideas.
>
>
>
> I'm having an issue where the 2nd leg of a call that is bridged to a verto
> endpoint has broken silence. When there is no sound on the line the audio
> goes completely silent (all background noise is dropped) but around every
> third of a second it repeatedly cuts back in for a fraction, then goes
> completely silent again. This only happens when the call is created using
> the originate command.
>
>
>
> Can anyone give me an idea of where to look next? I have a wireshark trace
> that during playback shows the audio cutting out periodically on the 2nd
> leg during periods of silence. The 1st leg is using webrtc encryption and I
> cant decode the stream.
>
>
>
> Thanks in advance!
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> Got Bugs? Report them here ! | Reddit:
> /r/freeswitch
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> ---------- Forwarded message ----------
> From: Chris Young
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 1 Sep 2015 07:31:00 +0000
> Subject: [Freeswitch-users] SIP profile not loading
>
> Hi all,
>
>
>
> Recently, we've begun experiencing a strange problem whereby the first SIP
> profile to be loaded gets 'stuck' and never actually completes its
> initialisation. This always seems to affect the first profile only, so if I
> have profiles named (for example):
>
>
>
> dummy.xml
>
> external.xml
>
> internal.xml
>
>
>
> then 'dummy' would fail to load but 'external' and 'internal' would be
> fine. No error messages are output to the logs but preliminary
> investigation suggests that the profile thread is becoming blocked for some
> reason. The specified IP address is valid and available and there are no
> other processes using the requested port. FreeSWITCH comes up successfully
> but 'sofia status' shows only 'external' and 'internal'. At this point, I
> can use 'sofia profile dummy start' and the profile loads correctly so it
> appears to be valid.
>
>
>
> Has anybody else seen this kind of behaviour or know what could be causing
> it?
>
>
>
> Many thanks,
>
> Chris
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 01 Sep 2015 12:56:08 +0400
> Subject: [Freeswitch-users] Playing with conditions
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
>
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 10:30:00 +0000
> Subject: Re: [Freeswitch-users] Playing with conditions
>
> Short answer: Each extension only has 1 set of conditions.
>
> The condition evaluating foobar is run *before* it gets set.
>
> After the play_and_get_digits you should transfer/execute_extension to a
> new extension that will evaluate ${foobar}
>
>
> -Avi Marcus
> BestFone
>
>
>
> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
> wrote:
>
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
Anthony Minessale II ? @anthmfs ? @FreeSWITCH ?
? http://freeswitch.org/ ? http://cluecon.com/ ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
*
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org ? +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
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From gaz.foreman at gmail.com Tue Sep 1 19:36:21 2015
From: gaz.foreman at gmail.com (Gary Foreman)
Date: Tue, 1 Sep 2015 16:36:21 +0100
Subject: [Freeswitch-users] Broken silence with webrtc
Message-ID:
I have that running. Unfortunately info reported in the console output
appears to be the same whether the call has audio or not.
The only message I've seen that has given a clue was "no audio stun for a
long time". The setup is purely on a LAN so I'm not sure why stun is
causing an issue.
On Tue, Sep 1, 2015 at 4:25 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Ken Rice)
>
>
> ---------- Forwarded message ----------
> From: Ken Rice
> To: "'FreeSWITCH Users Help'"
> Cc:
> Date: Tue, 1 Sep 2015 10:25:10 -0500
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> Just enable verto debugging in verto.conf.xml in your configs? it?ll print
> it right to the screen
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gary Foreman
> *Sent:* Tuesday, September 1, 2015 10:13 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Broken silence with webrtc
>
>
>
> Ok so the issue has been superseded by intermittent one-way / no audio.
> I'm getting it very intermittently (1 in every 30 calls or so) but I'm
> struggling to debug it as the traffic is encrypted and wireshark doesn't
> see it as rtp stream.
>
>
>
> Where is the best place to start debugging verto? I was previously using
> sip.js without any audio issues so it seems to be verto specific.
>
>
>
> On Tue, Sep 1, 2015 at 12:04 PM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Broken silence with webrtc (Stanislav Sinyagin)
>
>
> ---------- Forwarded message ----------
> From: Stanislav Sinyagin
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 13:03:25 +0200
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> is it running on a virtual machine?
>
> I found a strange effect that I could only reproduce in a VM, and never on
> physical hardware:
> https://freeswitch.org/jira/browse/FS-7805
>
> under certain load, an originate command triggers a continuous distortion
> in another, running and unrelated, channel.
>
> It seems to be triggered by insufficient CPU resource at the moment of the
> origination.
>
>
>
>
>
>
>
> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman
> wrote:
>
> I've found that it occurs after any bridge, its not specific to the
> originate command.
>
>
>
> Would you require a wireshark trace or the output of the freeswitch
> console?
>
>
>
> The scenario below reproduces the issue ...
>
>
>
> Test extension
>
>
>
>
>
>
>
>
>
>
>
>
>
> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>
>
>
>
>
>
>
> I originate a call from a polycom handset using g722 to the extension
> above.
>
> I originate a call using the verto client to the extension above.
>
>
>
> I get the uuids of the channels using show channels and use uuid_bridge
> [uuid1] [uuid2] to merge the channels.
>
>
>
> On Tue, Sep 1, 2015 at 11:30 AM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. How send in To header anonymous and in P-Asserted-Identity
> caller-id-number (Alex Polischuk)
> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
> 3. Broken silence with webrtc (Gary Foreman)
> 4. Re: Broken silence with webrtc (Brian West)
> 5. SIP profile not loading (Chris Young)
> 6. Playing with conditions (Dmitry Mordovin)
> 7. Re: Playing with conditions (Avi Marcus)
>
>
> ---------- Forwarded message ----------
> From: Alex Polischuk
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 17:02:25 +0300
> Subject: [Freeswitch-users] How send in To header anonymous and in
> P-Asserted-Identity caller-id-number
>
> Hi all,
>
>
>
> How I can define different users and domains in To and P-Asserted-Identity
> headers?
>
>
>
> Thanks,
>
> Alex
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 18:24:08 +0400
> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>
> Hello
>
> This example play conf-pin.wav and wait DTMF.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Is it possible to play WAV file in infinity loop and wait user DTMF?
>
> And how can I check DTMF input after user entered DTMF?
>
> In dialpeer, like.
> If ${DTMF} = 1 then bridge to XXX
> If ${DTMF} = 2 then play file and finish session
>
>
>
> Thank you.
> Dmitry
>
>
>
> ---------- Forwarded message ----------
> From: Gary Foreman
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Date: Mon, 31 Aug 2015 20:36:59 +0100
> Subject: [Freeswitch-users] Broken silence with webrtc
>
> Hi all,
>
>
>
> Hoping someone can point me in the right direction because after several
> hours I'm out of ideas.
>
>
>
> I'm having an issue where the 2nd leg of a call that is bridged to a verto
> endpoint has broken silence. When there is no sound on the line the audio
> goes completely silent (all background noise is dropped) but around every
> third of a second it repeatedly cuts back in for a fraction, then goes
> completely silent again. This only happens when the call is created using
> the originate command.
>
>
>
> Can anyone give me an idea of where to look next? I have a wireshark trace
> that during playback shows the audio cutting out periodically on the 2nd
> leg during periods of silence. The 1st leg is using webrtc encryption and I
> cant decode the stream.
>
>
>
> Thanks in advance!
>
>
>
>
>
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West
> To: FreeSWITCH Users Help
> Cc:
> Date: Mon, 31 Aug 2015 14:40:17 -0500
> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>
> Please provide logs and samples of how you originate this, sounds like
> Voice Activity Detection possibly.
>
>
>
> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman
> wrote:
>
> Hi all,
>
>
>
> Hoping someone can point me in the right direction because after several
> hours I'm out of ideas.
>
>
>
> I'm having an issue where the 2nd leg of a call that is bridged to a verto
> endpoint has broken silence. When there is no sound on the line the audio
> goes completely silent (all background noise is dropped) but around every
> third of a second it repeatedly cuts back in for a fraction, then goes
> completely silent again. This only happens when the call is created using
> the originate command.
>
>
>
> Can anyone give me an idea of where to look next? I have a wireshark trace
> that during playback shows the audio cutting out periodically on the 2nd
> leg during periods of silence. The 1st leg is using webrtc encryption and I
> cant decode the stream.
>
>
>
> Thanks in advance!
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> Got Bugs? Report them here ! | Reddit:
> /r/freeswitch
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
> ---------- Forwarded message ----------
> From: Chris Young
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 1 Sep 2015 07:31:00 +0000
> Subject: [Freeswitch-users] SIP profile not loading
>
> Hi all,
>
>
>
> Recently, we've begun experiencing a strange problem whereby the first SIP
> profile to be loaded gets 'stuck' and never actually completes its
> initialisation. This always seems to affect the first profile only, so if I
> have profiles named (for example):
>
>
>
> dummy.xml
>
> external.xml
>
> internal.xml
>
>
>
> then 'dummy' would fail to load but 'external' and 'internal' would be
> fine. No error messages are output to the logs but preliminary
> investigation suggests that the profile thread is becoming blocked for some
> reason. The specified IP address is valid and available and there are no
> other processes using the requested port. FreeSWITCH comes up successfully
> but 'sofia status' shows only 'external' and 'internal'. At this point, I
> can use 'sofia profile dummy start' and the profile loads correctly so it
> appears to be valid.
>
>
>
> Has anybody else seen this kind of behaviour or know what could be causing
> it?
>
>
>
> Many thanks,
>
> Chris
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Dmitry Mordovin
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Tue, 01 Sep 2015 12:56:08 +0400
> Subject: [Freeswitch-users] Playing with conditions
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
>
>
>
> ---------- Forwarded message ----------
> From: Avi Marcus
> To: FreeSWITCH Users Help
> Cc:
> Date: Tue, 1 Sep 2015 10:30:00 +0000
> Subject: Re: [Freeswitch-users] Playing with conditions
>
> Short answer: Each extension only has 1 set of conditions.
>
> The condition evaluating foobar is run *before* it gets set.
>
> After the play_and_get_digits you should transfer/execute_extension to a
> new extension that will evaluate ${foobar}
>
>
> -Avi Marcus
> BestFone
>
>
>
> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin
> wrote:
>
> Hello
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Why inner condition not works?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shakumarsoftware at gmail.com Tue Sep 1 00:52:31 2015
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Mon, 31 Aug 2015 14:52:31 -0600
Subject: [Freeswitch-users] Freeswitch 1.4.21 from source and Speex
Message-ID:
Hi,
I'm trying to cross-compile for an embedded target and building the latest
production build from source but it fails trying to build speex.
I also noticed in the libs directory there is no speex folder, so I had to
copy it from some other place.
How do I disable speex from compiling ? is it with --without-speex in the
configure script ?
thanks,
Shaks
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From ankitt.sharma1 at outlook.com Tue Sep 1 17:08:35 2015
From: ankitt.sharma1 at outlook.com (Ankit Sharma)
Date: Tue, 1 Sep 2015 13:08:35 +0000
Subject: [Freeswitch-users] Freeswitch 1.6 video recording issue using
uuid_record
Message-ID:
We are using freeswitch as media server in integration with kamailio. So
the purpose of using freeswitch is to record the two-way video calls between the legs
so
without any recording freeswich application or API being used my video
and audio is working just perfect (running on default media handling
mode i.e. proxy_media & bypass media not used), late-negotiation is
disabled, zrtp-passthru is set to false and disable-transcoding is set
to true
after exploring all over web over my recording concern i got to know about the following APP/APIs:
record, record_session, record_fsv, record_mp4, uuid_record (however i want to record in .mp4 format)
so
following the very first step I looked for "video-media-bug" branch and
got to know it is merged in freeSWITCH 1.6 master, so used the 1.6
master , and now the vide-media-bug branch is contained within my
current repo
now using all above mentioned applications i'am not
getting my recording done, here i'am sharing what's happening with
uuid_record being used as
my kbridge dialplan is:
and the fs_cli log says:
2015-08-21 13:19:50.980274 [NOTICE] switch_channel.c:1089 New Channel sofia/internal/102 at 192.168.5.99 [c1d25c8d-4e6e-458c-abe0-c40b89bc3d68]
2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 <102>->kb-101 in context public
2015-08-21 13:19:50.990233 [NOTICE] switch_ivr.c:1877 Transfer sofia/internal/102 at 192.168.5.99 to XML[kb-101 at default]
2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 <102>->kb-101 in context default
2015-08-21 13:19:51.000223 [INFO] switch_ivr_async.c:2339 Sending early media
2015-08-21 13:19:51.000223 [INFO] switch_core_media.c:6543 Activating VIDEO RTCP PORT 5244 interval 2000 mux 1
2015-08-21 13:19:51.000223 [NOTICE] mod_sofia.c:2296 Pre-Answer sofia/internal/102 at 192.168.5.99!
2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-in.G722] 16000hz
2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-out.G722] 16000hz
2015-08-21 13:19:51.010223 [NOTICE] switch_channel.c:1089 New Channel sofia/internal/101 at 192.168.5.99 [5127cf3a-c7ec-4038-a033-f287b2999a79]
2015-08-21 13:19:51.220289 [NOTICE] sofia.c:6817 Ring-Ready sofia/internal/101 at 192.168.5.99!
2015-08-21 13:19:55.240230 [INFO] switch_core_media.c:6543 Activating VIDEO RTCP PORT 0 interval 2000 mux 0
2015-08-21 13:19:55.240230 [NOTICE] sofia.c:7576 Channel [sofia/internal/101 at 192.168.5.99] has been answered
2015-08-21 13:19:55.240230 [NOTICE] switch_ivr_originate.c:3526 Channel [sofia/internal/102 at 192.168.5.99] has been answered
2015-08-21 13:20:04.180275 [NOTICE] sofia.c:952 Hangup sofia/internal/102 at 192.168.5.99 [CS_EXECUTE] [NORMAL_CLEARING]
2015-08-21 13:20:04.290279 [NOTICE] switch_ivr_bridge.c:874 Hangup sofia/internal/101 at 192.168.5.99 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1657 Session 29 (sofia/internal/101 at 192.168.5.99) Ended
2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/101 at 192.168.5.99 [CS_DESTROY]
2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1657 Session 28 (sofia/internal/102 at 192.168.5.99) Ended
2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/102 at 192.168.5.99 [CS_DESTROY]
now
in the recordings directory i got 2 files(can be seen in the log ) the
files hav the extension .G722, which is not supposed to be the required
output
since uuid_record is an API I tried using uuid_record in console
so started the call (after commenting the record api in dialplan) and it gave me this error:
-ERR Cannot locate session!
so help me out over this concern
thanks in advance :)
Ankit
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From haunma at keteu.org Tue Sep 1 19:44:01 2015
From: haunma at keteu.org (Mark Haun)
Date: Tue, 1 Sep 2015 08:44:01 -0700
Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue
In-Reply-To: <20150826153850.GA6402@hau.nz>
References: <20150826153850.GA6402@hau.nz>
Message-ID: <20150901154401.GA7510@hau.nz>
Anyone? I thought this might elicit some interest from the developers. Is
no one using mod_portaudio anymore?
Mark
Mark Haun [haunma at keteu.org] wrote:
> I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for
> my ARM-based home server (an NVIDIA Jetson board), with the intention of
> using mod_portaudio with a wireless USB headset for my main home phone. It
> works well when running freeswitch directly as root, but when I configured
> freeswitch to run at system startup, mod_portaudio could no longer find any
> audio devices.
>
> No problem, I thought, it's a simple permissions issue. I am using the
> example debian init script from the wiki (this is a Jessie install with SysV
> init rather than systemd). It runs freeswitch as user freeswitch, group
> daemon using the -u and -g command-line options. So I added "freeswitch" to
> the audio group, then used sudo to verify that I could play sound files with
> "aplay" as the freeswitch user. That worked, but it did not solve the
> mod_portaudio failure.
>
> Thinking there must be an issue with portaudio separate from ALSA, I fetched
> and built the "pa_devs" utility which enumerates and prints the available
> audio devices. (This is basically what mod_portaudio is doing on startup.)
> Running as the freeswitch user, that worked too. Hmmmm.
>
> I tried running freeswitch from the command line using sudo rather than the
> -u and -g options, i.e.
>
> $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc
>
> and that worked as well! Then I started experimenting with -u and -g and
> discovered that
>
> $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch
>
> allows mod_portaudio to load correctly, whereas
>
> $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon
>
> causes mod_portaudio to fail.
>
> Can anyone explain what's happening here? In both cases the process UID,
> GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as
> displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a
> bug in freeswitch? There are a couple of threads from 2-3 years ago in the
> archives, where mod_portaudio permissions issues were never resolved and the
> solution remained a mystery.
>
> For the time being I guess this (not using -g) is a workaround, but I'd sure
> like to understand what is going on.
>
> Mark
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
From krice at freeswitch.org Tue Sep 1 19:47:16 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 1 Sep 2015 10:47:16 -0500
Subject: [Freeswitch-users] Python based Event Socket Library
doesn't connect to remote FreeSWITCH
In-Reply-To:
References:
Message-ID: <18e301d0e4cd$7ae263b0$70a72b10$@freeswitch.org>
Did you make the eventsocket on the freeswitch server bind to something other than localhost which is its default configuration?
If you do this you should also ensure that this connection is firewalled off and not passed over the public internet as there is absolutely no encryption on this connection and the password is passed in the clear
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Fererro
Sent: Sunday, August 30, 2015 12:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Python based Event Socket Library doesn't connect to remote FreeSWITCH
To all Event Socket Library FreeSWITCH gurus!
Python implementation of the Event Socket Library doesn't seem to work with FreeSWITCH running on remote host. Only "localhost" or "127.0.0.1" has to be specified as the first parameter to the ESLconnection constructor to let client to successfully connect to the FreeSWITCH. Event if FreeSWITCH is running on the SAME machine (not even remote host) as the client and the "host" is specified with the IP address of the machine in the ESLconnection - no valid connection is getting established.
I am pretty sure that such a fundamental thing can't not to work and I am just missing something simple. Any help is greatly appreciated.
Thanks
Carlos
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From brian at freeswitch.org Tue Sep 1 19:56:56 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 1 Sep 2015 10:56:56 -0500
Subject: [Freeswitch-users] Delay and exec after answer
In-Reply-To: <55E5BD58.2010000@dwide.com>
References: <55E5AC13.8080708@dwide.com>
<55E5BD58.2010000@dwide.com>
Message-ID:
send_dtmf isn't an API call, you'll probably want to use an api call like
uuid_send_dtmf.
On Tue, Sep 1, 2015 at 9:59 AM, Dmitry Mordovin
wrote:
>
>
> api_on_answer_1='sched_api +10 none send_dtmf 1234 at 100'
>
> In Console
>
> [DEBUG] switch_event.c:1698 Parsing variable [api_on_answer_1]=[sched_api
> +10 none send_dtmf 1234 at 100]
>
> Looks good,
> ...
> But in 10 seconds:
>
> [DEBUG] mod_commands.c:4575 Command send_dtmf(1234 at 100):
> INVALID COMMAND!
>
> What is wrong??
>
>
>
>
> On 09/01/2015 06:40 PM, Michael Jerris wrote:
>
> you could use api on answer to schedule an api command
>
> On Tuesday, September 1, 2015, Dmitry Mordovin
> wrote:
>
>> Hello
>>
>> Is it possible to execute some api with delay after answer received?
>>
>> Trying
>>
>> api_on_answer_1='sleep 5000',api_on_answer_2='log -------delayed 5sec
>> after answer------'
>>
>>
>> but 'api_on_answer_2' executed at once
>>
>> Any ideas?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
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>>
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>>
>
>
> _________________________________________________________________________
> 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
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>
--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
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