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@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Broken silence with webrtc (Ken Rice)
---------- Forwarded message ---------- From: Ken Rice <krice@freeswitch.org> To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org> 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@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Gary Foreman *Sent:* Tuesday, September 1, 2015 10:13 AM *To:* freeswitch-users@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Ok so the issue has been superseded by intermittent one-way / no audio.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@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: Broken silence with webrtc (Stanislav Sinyagin)
---------- Forwarded message ---------- From: Stanislav Sinyagin <ssinyagin@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> 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 <gaz.foreman@gmail.com> 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
<extension name="Inbound_Routing">
<condition field="destination_number" expression="^2003$">
<action application="export" data="dialed_extension=$1"/>
<action application="export" data="transfer_ringback=$${uk-ring}"/>
<action application="set" data="RECORD_TITLE=Title goes here"/>
<action application="set" data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
<action application="export" data="send_silence_when_idle=false"/>
<action application="export" data="bridge_generate_comfort_noise=false"/>
<action application="export" data="rtp_enable_vad_in=false"/>
<action application="export" data="rtp_enable_vad_out=false"/>
<action application="export" data="fire_talk_events=true"/>
<action application="export" data="fire_not_talk_events=true"/>
<action application="export" data="recording_id=${uuid}"/>
<action application="set" data="media_bug_answer_req=true"/>
<action application="answer" />
<action application="playback" data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
</condition>
</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@lists.freeswitch.org> wrote:
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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 <alxpol@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> 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 <d.mordovin@dwide.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> 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.
<extension name="play_and_get_digits example">
<condition field="destination_number" expression="^(1888)$">
<action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/>
<action application="log" data="CRIT ${foobar}"/>
</condition>
</extension>
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 <gaz.foreman@gmail.com> To: freeswitch-users@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 <brian@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> 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 <gaz.foreman@gmail.com> 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!
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---------- Forwarded message ---------- From: Chris Young <Chris.Young@enghouse.com> To: "freeswitch-users@lists.freeswitch.org" < freeswitch-users@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 <d.mordovin@dwide.com> To: "freeswitch-users@lists.freeswitch.org" < freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 01 Sep 2015 12:56:08 +0400 Subject: [Freeswitch-users] Playing with conditions Hello
<section name="dialplan" description="Local call"> <context name="public"> <extension name="local_1">
<condition field="destination_number" expression="^(\d{4})$"> <action application="answer"/>
<action application="play_and_get_digits" data="1 1 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/>
<condition field="${foobar}" expression="1$"> <action application="log" data="CRIT --------------1------------------"/> <anti-action application="log" data="CRIT --------------not 1------------------"/> </condition>
<action application="log" data="CRIT ---- last ${foobar}"/> </condition>
</extension> </context> </section>
Why inner condition not works?
---------- Forwarded message ---------- From: Avi Marcus <avi@avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Date: Tue, 1 Sep 2015 10:30:00 +0000 Subject: Re: [Freeswitch-users] Playing with conditions
Short answer: Each extension only has 1 set of conditions.
The condition evaluating foobar is run *before* it gets set.
After the play_and_get_digits you should transfer/execute_extension to a new extension that will evaluate ${foobar}
-Avi Marcus BestFone
On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin@dwide.com> wrote:
Hello
<section name="dialplan" description="Local call"> <context name="public"> <extension name="local_1">
<condition field="destination_number" expression="^(\d{4})$"> <action application="answer"/>
<action application="play_and_get_digits" data="1 1 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/> <action application="log" data="CRIT ${foobar}"/>
<condition field="${foobar}" expression="1$"> <action application="log" data="CRIT --------------1------------------"/> <anti-action application="log" data="CRIT --------------not 1------------------"/> </condition>
<action application="log" data="CRIT ---- last ${foobar}"/> </condition>
</extension> </context> </section>
Why inner condition not works?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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