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    <div class="moz-cite-prefix">Do you have some options/variables in
      mind I should try using auto_outcall_prefix? I tried setting
      codec_string=silk@24000 and absolute_codec_string=silk@24000, but
      still FS offers only SILK/8000 to the b-leg.<br>
      <br>
      Am 19.09.2012 01:35, schrieb Seven Du:<br>
    </div>
    <blockquote cite="mid:D7E2D92BA89A47BBA28012CE29BEBCF4@gmail.com"
      type="cite">
      <div> maybe try to set&nbsp;<a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix">http://wiki.freeswitch.org/wiki/Variable_conference_auto_outcall_prefix</a>&nbsp;?</div>
      <div>
        <div><br>
        </div>
        <div>--&nbsp;</div>
        <div>Seven Du</div>
        <div>Sent with <a moz-do-not-send="true"
            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 moz-do-not-send="true"
                  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 moz-do-not-send="true"
                  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 moz-do-not-send="true"
                  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 moz-do-not-send="true"
                  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
                  moz-do-not-send="true"
                  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 moz-do-not-send="true"
                  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>
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        </span> </blockquote>
      <div> <br>
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      <br>
      <pre wrap="">_________________________________________________________________________
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</pre>
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