<div dir="ltr">It&#39;s negotiates INFO but sends rfc2833? <div>The liberal-dtmf option will accept rfc2833 always. See here:</div><div><meta http-equiv="content-type" content="text/html; charset=utf-8"><a href="http://wiki.freeswitch.org/wiki/DTMF#DTMF_Options">http://wiki.freeswitch.org/wiki/DTMF#DTMF_Options</a></div>

<div><br></div><div><br clear="all"><div dir="ltr"><span style="font-family:Verdana, Arial, Helvetica, sans-serif"><span style="font-size:small">-Avi</span></span></div><br>
<br><br><div class="gmail_quote">On Tue, Aug 23, 2011 at 11:02 AM, Mitja Thomas <span dir="ltr">&lt;<a href="mailto:mitja.thomas1@ewetel.de">mitja.thomas1@ewetel.de</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">


  

    
  
  <div bgcolor="#FFFFFF" text="#000000">
    Hi,<br>
    <br>
    we are experiencing problems with DTMF between Snom Sip Phones and
    FS. The DMTF Type is set to SIP INFO (sip info only on Snom Phones).
    We could nail it down to a missing &quot;telephone-event&quot; parameter in
    sip sdp:<br>
    <small><small><span>INVITE sip<span></span>:***6@spklw.x;user:phone

          SIP/2.0<br>
          Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport<br>
          From: &quot;Test&quot; <a href="mailto:sip:8000@spklw.x" target="_blank">&lt;sip:8000@spklw.x&gt;</a>;tag=qqnbrunjmq<br>
          To: &lt;<a>sip:***6@spklw.x;user:p</a><span></span>hone&gt;<br>
          Call-ID: 3c33e6c729b3-t9yol2613q6v<br>
          CSeq: 1 INVITE<br>
          Max-Forwards: 70<br>
          Contact: <a href="mailto:sip:8000@85.16.245.234:5060;line=9lwqdnrx" target="_blank">&lt;sip:8000@ip:5060;line=9lwqdnrx&gt;</a>;reg-id=1<br>
          X-Serialnumber: 000413262C33<br>
          P-Key-Flags: resolution=&quot;31x13&quot;, keys=&quot;4&quot;<br>
          User-Agent: snom370/8.4.32<br>
          Accept: application/sdp<br>
          Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
          SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE<br>
          Allow-Events: talk, hold, refer, call-info<br>
          Supported: timer, 100rel, replaces, from-change<br>
          Session-Expires: 3600;refresher=uas<br>
          Min-SE: 90<br>
          Content-Type: application/sdp<br>
          Content-Length: 244<br>
          <br>
          v=0<br>
          o=root 334284706 334284706 IN IP4 phoneip<br>
          s=call<br>
          c=IN IP4 phoneip<br>
          t=0 0<br>
          m=audio 12610 RTP/AVP 8 9<br>
          a=direction:b<span></span>oth<br>
          a=rtpmap:8 PCMA/8000<br>
          a=rtpmap:9 G722/8000<br>
          a:p<span></span>time:20<br>
          a=rtcp-xr:s<span></span>tat-summary=loss,dup,jitt<br>
          a=sendrecv</span></small></small><br>
    <br>
    That resulted in sofia_glue.c disabling DTMF. A slight change in
    sofia_glue.c (Line 4839 in commit
    71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us:<br>
    <small>if (tech_pvt-&gt;dtmf_type == DTMF_INFO) { //EWETEL<br>
      } else {                                //EWETEL<br>
                     
      switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
      SWITCH_LOG_DEBUG, &quot;Disable 2833 dtmf\n&quot;);<br>
                      switch_channel_set_variable(tech_pvt-&gt;channel,
      &quot;dtmf_type&quot;, &quot;none&quot;);<br>
                      tech_pvt-&gt;dtmf_type = DTMF_NONE;<br>
                      te = tech_pvt-&gt;recv_te = 0;<br>
      } //EWETEL</small><br>
    <br>
    Now the question is: <br>
    Is this a too restrictive behaviour in FS or a shortcoming from the
    Snom phones (which would mean we have to submit a bug, wait, wait,
    wait and then hopefully someday it gets fixed :) )?<br>
    <br>
    Regards,<br>
    Mitja<br>
    <br>
    <pre cols="72">-- 
Mitja Thomas
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<br></blockquote></div><br></div></div>