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    <div class="moz-cite-prefix">I think problem is in function
      inherit_codec in switch_ivr_originate.c<br>
      It checks codecs from both sides but not taking in account 3pcc
      mode and obviously can't find any matches.<br>
      <br>
      On 06/11/15 21:38, Michael Jerris wrote:<br>
    </div>
    <blockquote
      cite="mid:6FF04EB3-0CCB-473A-BBB4-A70DD95A5659@jerris.com"
      type="cite">
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      do you have the mod loaded for g729?
      <div class=""><br class="">
        <div>
          <blockquote type="cite" class="">
            <div class="">On Nov 6, 2015, at 12:23 PM, Serge S. Yuriev
              &lt;<a moz-do-not-send="true" href="mailto:me@nevian.org"
                class="">me@nevian.org</a>&gt; wrote:</div>
            <br class="Apple-interchange-newline">
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                <div class="moz-cite-prefix">But why? I won't any
                  transcoding, both we and peer supporting g.729.<br
                    class="">
                  But in case of completely independent negotiation we
                  will choose g.722 and they g.729.<br class="">
                  <br class="">
                  On 06/11/15 19:31, Sergey Safarov wrote:<br class="">
                </div>
                <blockquote
cite="mid:CAHtxdDd5FJg1HyYCjv3yRXKBB498FAx6-WhMjaeOR-5RP1tc2w@mail.gmail.com"
                  type="cite" class="">
                  <p dir="ltr" class="">Try with PCM codec. G729 may
                    require license in your case.</p>
                  <br class="">
                  <div class="gmail_quote">
                    <div dir="ltr" class="">On Fri, Nov 6, 2015,
                      19:19 Serge S. Yuriev &lt;<a
                        moz-do-not-send="true"
                        class="moz-txt-link-abbreviated"
                        href="mailto:me@nevian.org"><a class="moz-txt-link-abbreviated" href="mailto:me@nevian.org">me@nevian.org</a></a>&gt;

                      wrote:<br class="">
                    </div>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br
                        class="">
                      <br class="">
                      Our PBX sending INVITEs w/o SDP so we set enable
                      3pcc=proxy<br class="">
                      In dialplan I have set inherit_codec=true<br
                        class="">
                      <br class="">
                      We receive from peer:<br class="">
                      <br class="">
                          m=audio 43512 RTP/AVP 18 101<br class="">
                          a=rtpmap:18 G729/8000<br class="">
                          a=fmtp:18 annexb=no<br class="">
                          a=rtpmap:101 telephone-event/8000<br class="">
                          a=fmtp:101 0-15<br class="">
                          a=ptime:20<br class="">
                      <br class="">
                      <br class="">
                      But FS refuses to send this to Caller<br class="">
                      <br class="">
                      2015-11-06 19:00:06.331944 [DEBUG]
                      switch_ivr_originate.c:416 Codec<br class="">
                      string G729@8000h@20i not supported on
                      sofia/internal/<a moz-do-not-send="true"
                        class="moz-txt-link-abbreviated"
                        href="mailto:12550@10.0.64.63">12550@10.0.64.63</a>,<br
                        class="">
                      skipping inheritance<br class="">
                      <br class="">
                      But why while there is no SDP from A side? PBX
                      definitely supports G729.<br class="">
                    </blockquote>
                  </div>
                </blockquote>
                <br class="">
                <pre class="moz-signature" cols="72">-- 
Serge S. Yuriev
Lead VoIP engineer

</pre>
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          </blockquote>
        </div>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Serge S. Yuriev
Lead VoIP engineer

</pre>
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