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

Avi Marcus avi at avimarcus.net
Mon Nov 8 20:18:42 UTC 2021 

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
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>>>>>>> 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
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>>> 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
>>
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>>
>> [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
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