<div class="gmail_quote"><br><br>thanks, I don't need 729, 711 is fine. going to configure an external MTP anyway just to rule it out.<div><br></div><div><font color="#888888">David</font><div><div></div><div class="h5">
<br><br><div class="gmail_quote">On 26 July 2011 01:05, Nicholas Samios <span dir="ltr"><<a href="mailto:nsamios@staff.iinet.net.au" target="_blank">nsamios@staff.iinet.net.au</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New";color:#1F497D">The software MTP on UCM only supports 711, if you need to use 729 you’ll need to configure an external MTP/transcoder or CUBE.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New";color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New";color:#1F497D">Using a CUBE in the middle as a demarc is best practice, gives you all the features of SIP & SDP manipulation (e.g delayed to early, message normalization), transcoding, etc and also avoids a direct SIP trunk to your UCM.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New";color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Cristobal Priego<br>
<b>Sent:</b> Tuesday, July 26, 2011 1:28 AM<br><b>To:</b> David Martin<br><b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> Re: [cisco-voip] CUCM 6.1 and SIP Trunking<u></u><u></u></span></p>
<div><div></div><div><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal" style="margin-bottom:12.0pt">on version 6.X if you want to send a early offer on your sip trunk you need MTP and the codec supported is g.711 only. if you're trying to use g.729 is not going to work <u></u><u></u></p>
<div><p class="MsoNormal">2011/7/25 David Martin <<a href="mailto:adavidm@gmail.com" target="_blank">adavidm@gmail.com</a>><u></u><u></u></p><p class="MsoNormal">All,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">I am trying to set up a SIP trunk between CUCM 6.1 and Freeswitch. I only need calls outbound from Cisco to Freeswitch at the moment but am having some difficulties.<u></u><u></u></p></div>
<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">The Cisco side never sends an SDP, either in the INVITE or the ACK, and Freeswitch needs this to operate. I have tried using "require media termination point" and "disable early media" options but there appears to be no difference, which is strange. I include a pastebin of the conversation:<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><span style="font-size:10.0pt;background:white"><a href="http://pastebin.com/T3eqms8N" target="_blank"><span style="color:#0658B5">http://pastebin.com/T3eqms8N</span></a></span><u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">and a couple of screenshots of the CUCM config for the trunk.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div>
<p class="MsoNormal"><a href="http://img200.imageshack.us/img200/9405/siptrunk.png" target="_blank">http://img200.imageshack.us/img200/9405/siptrunk.png</a><u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal"><a href="http://img191.imageshack.us/img191/3863/sipprofile.png" target="_blank">http://img191.imageshack.us/img191/3863/sipprofile.png</a><u></u><u></u></p></div><div><p class="MsoNormal">
<u></u> <u></u></p></div><div><p class="MsoNormal">Any thoughts on where to start looking? I was wondering if CUCM will continue to attempt the call if there is a problem with the MTP( I am using software MTP on CUCM at the moment). If this is the case then I can see why the SDP does not appear in the INVITE (assuming the MTP is broken), but not why it does not appear in the ACK.<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks in advance.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><span style="color:#888888">David<u></u><u></u></span></p>
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