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

Thu Oct 7 11:10:42 UTC 2021

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211007/18749e8a/attachment.html>