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maybe try to set <a href="http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix">http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix</a> ?</div>
<div><div><br></div><div>-- </div><div>Seven Du</div><div>Sent with <a href="http://www.sparrowmailapp.com/?sig">Sparrow</a></div><div><br></div></div>
<p style="color: #A0A0A8;">On Tuesday, September 18, 2012 at 9:22 PM, Oliver Saggau | Sysvision GmbH wrote:</p>
<blockquote type="cite" style="border-left-style:solid;border-width:1px;margin-left:0px;padding-left:10px;">
<span><div><div><div>Hi everyone,</div><div>I got a strange problem with mod_conference and auto_outcall taking the </div><div>wrong codec for b-leg. For what I can see from the log everything seems </div><div>fine. The a-leg calls with the following SDP:</div><div><br></div><div>2012-09-18 08:57:47.410023 [DEBUG] sofia.c:6293 Remote SDP:</div><div>v=0</div><div>o=- 3556961565 3556961565 IN IP4 10.20.30.125</div><div>s=pjmedia</div><div>t=0 0</div><div>m=audio 4000 RTP/AVP 99 101</div><div>c=IN IP4 10.20.30.125</div><div>a=rtpmap:99 SILK/24000</div><div>a=fmtp:99 useinbandfec=0</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>a=rtcp:4001 IN IP4 10.20.30.125</div><div><br></div><div>And the SILK/24000 codec is accepted according to the log file:</div><div>2012-09-18 08:57:47.650018 [DEBUG] sofia_glue.c:3077 Set Codec </div><div>sofia/internal/<a href="mailto:1000004@10.20.30.240">1000004@10.20.30.240</a> SILK/24000 20 ms 480 samples 40000 bits</div><div>2012-09-18 08:57:47.650018 [DEBUG] switch_core_codec.c:111 </div><div>sofia/internal/<a href="mailto:1000004@10.20.30.240">1000004@10.20.30.240</a> Original read codec set to SILK:120</div><div>2012-09-18 08:57:47.650018 [DEBUG] mod_sofia.c:836 Local SDP </div><div>sofia/internal/<a href="mailto:1000004@10.20.30.240">1000004@10.20.30.240</a>:</div><div>v=0</div><div>o=FreeSWITCH 1347951445 1347951446 IN IP4 10.20.30.240</div><div>s=FreeSWITCH</div><div>c=IN IP4 10.20.30.240</div><div>t=0 0</div><div>m=audio 21622 RTP/AVP 99 101</div><div>a=rtpmap:99 SILK/24000</div><div>a=fmtp:99 useinbandfec=0; usedtx=0; maxaveragebitrate=40000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=sendrecv</div><div><br></div><div>The conference get's created and activates the read/write codecs for the </div><div>first member:</div><div>2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7092 Raw Codec </div><div>Activation Success L16@24000hz 1 channel 20ms</div><div>2012-09-18 08:57:47.990021 [DEBUG] mod_conference.c:7137 Raw Codec </div><div>Activation Success L16@48000hz 1 channel 20ms</div><div><br></div><div>After that he's trying to do the auto_outcall and that's where things </div><div>get weird:</div><div>EXECUTE sofia/internal/<a href="mailto:1000004@freeswitch.aws.hosts.corpex.de">1000004@freeswitch.aws.hosts.corpex.de</a> </div><div>conference_set_auto_outcall(sofia/internal/sip:<a href="mailto:1000003@10.20.30.108">1000003@10.20.30.108</a>:63741;transport=UDP;ob)</div><div>2012-09-18 08:57:47.990021 [NOTICE] switch_channel.c:951 New Channel </div><div>sofia/internal/sip:<a href="mailto:1000003@10.20.30.108">1000003@10.20.30.108</a>:63741 </div><div>[0236d005-1ff0-4f13-9370-bf7736087dd0]</div><div>2012-09-18 08:57:47.990021 [DEBUG] sofia_glue.c:2637 Local SDP:</div><div>v=0</div><div>o=FreeSWITCH 1347952749 1347952750 IN IP4 10.20.30.240</div><div>s=FreeSWITCH</div><div>c=IN IP4 10.20.30.240</div><div>t=0 0</div><div>m=audio 20318 RTP/AVP 98 0 8 101 13</div><div>a=rtpmap:98 SILK/8000</div><div>a=fmtp:98 useinbandfec=1; usedtx=0</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=sendrecv</div><div>2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7092 Raw Codec </div><div>Activation Success L16@8000hz 1 channel 20ms</div><div>2012-09-18 08:57:52.590016 [DEBUG] mod_conference.c:7137 Raw Codec </div><div>Activation Success L16@48000hz 1 channel 20ms</div><div><br></div><div>Why is FS only offering SILK/8000 for b-leg? My outbound_codec_prefs is </div><div>configured like this:</div><div>outbound_codec_prefs=silk@24000,silk@16000,silk@8000,PCMU,PCMA,GSM</div><div><br></div><div>Thanks for your help,</div><div>Oliver</div><div><br></div><div>_________________________________________________________________________</div><div>Professional FreeSWITCH Consulting Services:</div><div><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a></div><div><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a></div><div><br></div><div>FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div><div><a href="http://www.cudatel.com">http://www.cudatel.com</a></div><div><br></div><div>Official FreeSWITCH Sites</div><div><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></div><div><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a></div><div><a href="http://www.cluecon.com">http://www.cluecon.com</a></div><div><br></div><div>FreeSWITCH-users mailing list</div><div><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a></div><div><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div><div>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div><div><a href="http://www.freeswitch.org">http://www.freeswitch.org</a></div></div></div></span>
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