[Freeswitch-users] Bridge to other FS server has no audio until DTMF

Brian West brian at freeswitch.com
Fri Nov 12 20:38:44 UTC 2021 

First off, don't use export and don't use nolocal, just set it and see what
happens.

/b


On Mon, Nov 8, 2021 at 12:36 PM Avi Marcus <avi at avimarcus.net> wrote:

> Is there a way to "fix" the standoff? Perhaps a header to send or a
> channel variable to set?
>
> I'd like to do bypass_media to cut one of my servers out of the media
> path, but then I won't be able to queue an rfc2833 digit press.
>
> Thanks,
> -Avi Marcus
> BestFone
>
>
> On Thu, Oct 7, 2021 at 7:50 PM Avi Marcus <avi at avimarcus.net> wrote:
>
>> I had to do this to get it to execute on the B leg:
>> <action application="export" data="nolocal:execute_on_answer=playback
>> silence_stream://100"/>
>>
>> ... but it didn't help. Only DTMF worked... either manually dialed or via
>> queue_dtmf
>> Freeswitch A waited for my DTMF to actually send the silence.
>> Version 1.10.6 -release-18-1ff9d0a60e 64bit
>>
>>
>>  2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort
>> noise payload to 13
>>  2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/
>> JOIN_CLASS_7229999 at voip.bestfone.com] has been answered
>>  EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com
>> playback(silence_stream://100)
>>  2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>>  -- 20 seconds later when I pressed a button --
>>
>>  2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done
>> playing file silence_stream://100
>>  2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/
>> JOIN_CLASS_7229999 at voip.bestfone.com) Callstate Change DOWN -> ACTIVE
>>  2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793
>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>>  2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Running State
>> Change CS_EXCHANGE_MEDIA (Cur 12 Tot 351090)
>>  2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654
>> (sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State
>> EXCHANGE_MEDIA
>>  2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA
>>  2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet
>> for [5] ts=960 dur=160/160/2000 seq=26795 lw=960
>>
>>
>>
>> This seemingly shouldn't be an issue. FS1 already has active media from
>> the A leg, so it should initiate to the B leg. The B leg has been
>> instructed to play a file, so it should initiate to the A leg...
>> But if this is somehow unavoidable, perhaps we need a workaround config,
>> where we have a simple variable in the bridge string to avoid the standoff?
>>
>> -Avi Marcus
>>
>>
>>
>> On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian at freeswitch.com> wrote:
>>
>>> execure_on_answer=playback::silence_stream://100 should solve it.
>>>
>>> /b
>>> PS, the non pc term that this has been said to be is
>>> https://en.wikipedia.org/wiki/Mexican_standoff
>>>
>>> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> I meant there's audio from pstn to fs1, but indeed I'm observing no
>>>> audio between fs1 and fs2.
>>>>
>>>> What api should I call with api on answer..?
>>>>
>>>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> If you see rtp glowing both ways, then this is not the stalemate I was
>>>>> talking about. The scenario I’m referring to is about FS not starting
>>>>> sending rtp waiting for the other side to start sending, and the other side
>>>>> doing the same thing, thus going into a stalemate. This is solved by
>>>>> injecting a silence (I would do api_on_answer).
>>>>>
>>>>> What you’re describing seems different to me.
>>>>>
>>>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>>> I'm using dialplan bridge, so then the dialplan is over. How do I
>>>>>> send silence after the bridge...? An api_on_answer with a uuid_broadcast..
>>>>>> seems overly complicated.
>>>>>>
>>>>>> <action application="bridge" data="sofia/external/
>>>>>> number at yyy.bestfone.com"/>
>>>>>>
>>>>>>
>>>>>> (And I don't know why there isn't audio - I had to set up an audio to
>>>>>> get to this options in the IVR... so there's already audio. And Server B
>>>>>> also started a file playback so should have initiated audio.)
>>>>>>
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>>> I seem to remember Brian saying this was because FS is waiting for
>>>>>>> the remote end to send audio before starting itself. I believe he
>>>>>>> recommended sending an empty (silence) to force the audio stream to be sent
>>>>>>> even if fs hasn’t received anything.
>>>>>>>
>>>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>>>
>>>>>>>> I started a new thread in case anyone muted it... it wasn't simply
>>>>>>>> a network issue.
>>>>>>>>
>>>>>>>> It seems the bridging occurs and dialplan processes, but no media
>>>>>>>> flows - until DTMF from the A-leg.
>>>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>>>>>> freeswitch B.
>>>>>>>>
>>>>>>>> Calls directly from carrier to Freeswitch B are fine.
>>>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>>>>>> Freeswitch B are also fine.
>>>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is
>>>>>>>> in the media path, it's an FS issue...
>>>>>>>>
>>>>>>>>
>>>>>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>>>>>> bridge, to force the audio stream to update.
>>>>>>>>
>>>>>>>> Here's the log on freeswitch B:
>>>>>>>>
>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>>>  log(DEBUG class chosen: 1234567)
>>>>>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>>>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>>>  playback(class/hold-wait-teacher.wav)
>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>>>  entering state [completed][200]
>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>>>  entering state [ready][200]
>>>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz
>>>>>>>>  1 channels 20ms
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>>>>>> 2021-10-07 09:16:34.923190 [INFO
>>>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080
>>>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>>>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>>>>>
>>>>>>>>
>>>>>>>> You can see a 10 second gap between call ready 200 and correct
>>>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't
>>>>>>>> happen automatically, only when I choose to press something.
>>>>>>>>
>>>>>>>>
>>>>>>>> Any ideas as to the root cause of this?
>>>>>>>>
>>>>>>>>
>>>>>>>> -Avi Marcus
>>>>>>>>
>>>>>>>> ---------- Forwarded message ---------
>>>>>>>> From: Avi Marcus <avi at avimarcus.net>
>>>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>>>>>> Subject: Bridge to other FS server has no audio ???
>>>>>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>
>>>>>>>>
>>>>>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>>>>>> think my upstream changed his SDP again.
>>>>>>>>
>>>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to
>>>>>>>> FS server B. Media works.
>>>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B
>>>>>>>> (not on bypass). Media works.
>>>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges
>>>>>>>> to FS server B. Call looks OK, but no media is flowing (I don't hear
>>>>>>>> anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All
>>>>>>>> the same codecs are set in the json cdrs (PCMU).
>>>>>>>>
>>>>>>>> FS server B is to join a conference if that matters.
>>>>>>>>
>>>>>>>> I was assuming it had to do with codecs, but setting
>>>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>>>>>> it's already always PCMU.
>>>>>>>>
>>>>>>>> I have NO clue what further could cause this other than codecs,
>>>>>>>> which seem to be fine. Any ideas please?
>>>>>>>>
>>>>>>>>
>>>>>>>> -Avi Marcus
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>> https://signalwire.com
>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>>> services.
>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>
>>>>>>>> Join our online community to chat in real time
>>>>>>>> https://signalwire.community
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Services
>>>>>>>> sales at freeswitch.com
>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> https://freeswitch.com/oss
>>>>>>>> https://freeswitch.org/confluence
>>>>>>>> https://cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>>
>>>>>>> David Villasmil
>>>>>>> email: david.villasmil.work at gmail.com
>>>>>>> phone: +34669448337
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>> --
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>>
>>> Brian West | Co-founder and Developer
>>>
>>> Need Commercial support? email sales at freeswitch.com
>>>
>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>> Email: brian at freeswitch.com
>>>
>>> Mobile: 918-424-9378
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>> <https://www.facebook.com/freeswitch> [image:
>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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