Hello All,<div> I am having a bit peculiar problem with Asterisk 11 for a carrier. This carrier shares quite some information in SDP header, which should not be the problem, however what happen is as follow:</div>
<div><br></div><div><br></div><div>Carrier----> (INVITE) -> <b>SIP Proxy -> Asterisk 11 -> Answer()</b> -> right after answering call drops... Carrier send a BYE with (cause 79: service or option not implemented).</div>
<div><br></div><div><b>NOTE: Please refer to complete SIP traces attached. </b></div><div><b><br></b></div><div><b>Also Note:</b></div><div><div><u>Carrier</u>: 62.61.147.214</div><div><u>Proxy</u>: 77.X.X.X:5060</div><div>
<u>Asterisk11</u>: 77.X.X.X:5080</div></div><div><span style="font-size:x-small"><br></span></div><div><b><u>Here is Invite SDP from Carrier -> Proxy -> Asterisk 11</u></b></div><div><br></div><div><div><font size="1">INVITE sip:69609000@77.X.X.X SIP/2.0</font></div>
<div><font size="1">v=0</font></div><div><font size="1">o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214</font></div><div><font size="1">s=Phone-Call</font></div><div><font size="1">c=IN IP4 77.X.X.X</font></div>
<div><font size="1">t=0 0</font></div><div><font size="1">m=audio 53372 RTP/AVP 8 118 18</font></div><div><font size="1">a=rtpmap:8 PCMA/8000</font></div><div><font size="1">a=rtpmap:118 PCMA/8000</font></div><div><font size="1">a=gpmd:118 vbd=yes</font></div>
<div><font size="1">a=rtpmap:18 G729/8000</font></div><div><font size="1">a=fmtp:18 annexb=no</font></div><div><font size="1">a=ptime:20</font></div><div><font size="1">a=sendrecv</font></div><div><font size="1">a=rtcp:53373 IN IP4 77.X.X.X</font></div>
<div><font size="1">m=image 56854 udptl t38</font></div><div><font size="1">a=T38FaxVersion:0</font></div><div><font size="1">a=T38MaxBitRate:14400</font></div><div><font size="1">a=T38FaxMaxBuffer:1024</font></div><div><font size="1">a=T38FaxMaxDatagram:122</font></div>
<div><font size="1">a=T38FaxRateManagement:transferredTCF</font></div><div><font size="1">a=T38FaxUdpEC:t38UDPRedundancy</font></div></div><div><br clear="all"><div><div><i><b><u>SDP:After Answered by Asterisk 11</u></b></i></div>
<div><font size="1">v=0</font></div><div><font size="1">o=root 164966782 164966782 IN IP4 77.X.X.X</font></div><div><font size="1">s=Asterisk v11.0.1</font></div><div><font size="1">c=IN IP4 77.X.X.X</font></div><div><font size="1">t=0 0</font></div>
<div><font size="1">m=audio 12636 RTP/AVP 18 8</font></div><div><font size="1">a=rtpmap:18 G729/8000</font></div><div><font size="1">a=fmtp:18 annexb=no</font></div><div><font size="1">a=rtpmap:8 PCMA/8000</font></div><div>
<font size="1">a=ptime:20</font></div><div><font size="1">a=sendrecv</font></div><div><font size="1"><b><u>m=image 0 udptl t38</u></b></font></div></div><div><br></div><div>I have tired by disabling/unloading fax modules as <b>I am not using</b> them but no results. Secondly, also tried tweaking of udptl ever-odd nothing worked.</div>
<div><br></div><div>The same carrier works for Asterisk 1.6.X and the only difference I have notice so far is the above underlined line in Answered SDP -> m=image 0 udptl t38. I think if I some how do not advertise udptl here i would be able to avoid this scenario. I have tried multiple ways to strip off SDP from incoming INVITE at SIP Proxy level but it is not SDP wise enough. </div>
<div><br></div><div><br></div><div><b>Note:</b></div><div><br></div><div>In Asterisk 1.6 => WARNING[32671]: chan_sip.c:8833 process_sdp: Unsupported SDP media type in offer: image 59978 udptl t38</div><div>In Asterisk 11 => WARNING[18748][C-0000002f]: chan_sip.c:10277 process_sdp: Failed to initialize UDPTL, declining image stream</div>
<div><br></div><div><br></div>-- <br><font face="'times new roman', serif">Regards</font><div><font face="'times new roman', serif"><br></font><div><pre cols="72"><font face="'times new roman', serif">
<i>Muhammad Salman Zafar</i></font></pre><pre cols="72"></pre></div></div>
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