<div dir="ltr"><div dir="ltr">I started a new thread in case anyone muted it... it wasn't simply a network issue.<div><br></div><div>It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg.</div><div>Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B.</div><div><br></div><div>Calls directly from carrier to Freeswitch B are fine.</div><div>Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine.</div><div>So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue...</div><div><br></div><div><br></div><div>I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update.</div><div><br></div><div>Here's the log on freeswitch B:</div><div><br></div><div><div style="color:rgb(212,212,212);background-color:rgb(30,30,30);font-family:Consolas,"Cascadia Code",Hack,"Courier New",monospace,Consolas,"Courier New",monospace;font-size:14px;line-height:19px;white-space:pre-wrap"><div>EXECUTE [<span style="color:rgb(206,145,120)">depth=0</span>] sofia/external/<a href="mailto:972581234567@172.123.123.123" target="_blank">972581234567@172.123.123.123</a> log(DEBUG class chosen: 1234567)</div><div>2021-10-07 09:16:24.343175 [<span style="color:rgb(206,145,120)">DEBUG</span>] mod_dptools.c:1879 class chosen: 1234567</div><div>EXECUTE [<span style="color:rgb(206,145,120)">depth=0</span>] sofia/external/<a href="mailto:972581234567@172.123.123.123" target="_blank">972581234567@172.123.123.123</a> javascript(conference/lookupAndJoinConference.js 1234567)</div><div>EXECUTE [<span style="color:rgb(206,145,120)">depth=0</span>] sofia/external/<a href="mailto:972581234567@172.123.123.123" target="_blank">972581234567@172.123.123.123</a> playback(class/hold-wait-teacher.wav)</div><div>2021-10-07 09:16:24.363379 [<span style="color:rgb(206,145,120)">DEBUG</span>] sofia.c:7406 Channel sofia/external/<a href="mailto:972581234567@172.123.123.123" target="_blank">972581234567@172.123.123.123</a> entering state [<span style="color:rgb(206,145,120)">completed</span>][200]</div><div>2021-10-07 09:16:24.363379 [<span style="color:rgb(206,145,120)">DEBUG</span>] sofia.c:7406 Channel sofia/external/<a href="mailto:972581234567@172.123.123.123" target="_blank">972581234567@172.123.123.123</a> entering state [<span style="color:rgb(206,145,120)">ready</span>][200]</div><div>2021-10-07 09:16:24.363379 [<span style="color:rgb(206,145,120)">DEBUG</span>] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms</div><br><br><br><br><div>2021-10-07 09:16:34.903283 [<span style="color:rgb(206,145,120)">DEBUG</span>] switch_rtp.c:7793 Correct audio ip/port confirmed.</div><div>2021-10-07 09:16:34.923190 [<span style="color:rgb(206,145,120)">DEBUG</span>] switch_rtp.c:8038 RTP RECV DTMF 3:2080</div><div>2021-10-07 09:16:34.923190 [<span style="color:rgb(206,145,120)">INFO</span>] switch_channel.c:522 RECV DTMF 3:2080</div><div>2021-10-07 09:16:34.923190 [<span style="color:rgb(206,145,120)">DEBUG</span>] mod_dptools.c:2389 Digit 3</div><div>2021-10-07 09:16:37.143169 [<span style="color:rgb(206,145,120)">DEBUG</span>] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav</div></div></div><div><br></div><div><br></div><div>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.</div><div><br></div><div><br></div><div>Any ideas as to the root cause of this?</div><div><br></div><div><br clear="all"><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr">-Avi Marcus<br></div></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">---------- Forwarded message ---------<br>From: <strong class="gmail_sendername" dir="auto">Avi Marcus</strong> <span dir="auto"><<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>></span><br>Date: Wed, Oct 6, 2021 at 3:32 PM<br>Subject: Bridge to other FS server has no audio ???<br>To: FreeSWITCH Users Help <<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a>><br></div><br><br><div dir="ltr">Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again.<div><br>- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works.<br>- FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works.<br>- 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).<br><br></div><div>FS server B is to join a conference if that matters.<br><br></div><div>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.</div><div><br></div><div>I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please?<br><div><br></div><div> <br clear="all"><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr">-Avi Marcus<br><br></div></div></div></div></div></div></div></div>
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