<div dir="ltr">CUCM is CCM 4.1.3(sr7)<br>Yes, the same behavior is exhibited in both directions.<br>G711ulaw all the way thru... if I change it to SIP to H323, it works fine in both directions. No transcoders in the network.<br>
Region is 711<br>The MTP required check box is set on the SIP trunk on CCM.<br><br><br><br>Jonathan<br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 2:45 PM, Chris Ward <span dir="ltr"><<a href="mailto:chrward@cisco.com">chrward@cisco.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">If the call works until the phone is answered, then it is probably media/capability related.<br>
<br>
First, is this the call flow?<br>
<br>
IP Phone --- CUCM --- SIP Trunk --- IPIPGW --- SIP --- CUCME --- IP Phone<br>
<br>
Now these questions:<br>
<br>
</span></font><ol><li><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">What is the version of CUCM?
</span></font></li><li><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Do calls fail in both directions?
</span></font></li><li><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">What is the region setting (what codec) between the CUCM IP Phone and the SIP trunk on CUCM?
</span></font></li><li><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;">Is the CUCM SIP trunk doing early media (is the "MTP required" checkbox checked)? <br>
</span></font></li></ol><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
-Chris<br>
<br>
<hr size="3" width="95%" align="center"><b>From: </b>Jonathan Charles <<a href="http://jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>><br>
<b>Date: </b>Wed, 6 Aug 2008 14:32:31 -0500<br>
<b>To: </b>Stephen Collinson <<a href="http://scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>><br>
<b>Cc: </b>OSL CCIE Voice Lab Exam <<a href="http://ccie_voice@onlinestudylist.com" target="_blank">ccie_voice@onlinestudylist.com</a>>, cisco voip <<a href="http://cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>><br>
<b>Subject: </b>Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip<div><div></div><div class="Wj3C7c"><br>
<br>
Right, the question is, how do you configure it correctly?<br>
<br>
What would cuz the audio to not cut thru and the call to drop... I was suspecting codec, but it is G711 all the way thru (hard coded on each dial peer)<br>
<br>
<br>
<br>
Jonathan<br>
<br>
On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson <<a href="http://scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>> wrote:<br>
</div></div></span></font><div><div></div><div class="Wj3C7c"><blockquote><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">SIP to SIP should work fine, when configured correctly.<br>
</span></font></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
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</span></font><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">I was just trying to give you a scenario where we may need to use it. Apologies if this was not helpful<br>
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<font size="2"><font face="Tahoma, Verdana, Helvetica, Arial"><span style="font-size: 10pt;"><b>From:</b> Jonathan Charles [<a href="mailto:jonvoip@gmail.com" target="_blank">mailto:jonvoip@gmail.com</a>] <br>
<b>Sent:</b> 06 August 2008 19:55<br>
<b>To:</b> Stephen Collinson<br>
<b>Cc:</b> cisco voip; OSL CCIE Voice Lab Exam<br>
<b>Subject:</b> Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip<br>
</span></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;"> <br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;">Perhaps I wasn't clear...<br>
<br>
<br>
There is no CUE.<br>
<br>
This is a SCCP phone on a CCME, and a SCCP phone on CCM with a SIP trunk to an IPIPGW, and a SIP dial-peer to CCME...<br>
<br>
<br>
Jonathan<br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;">On Wed, Aug 6, 2008 at 1:52 PM, Stephen Collinson <<a href="http://scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>> wrote:<br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">Perhaps worth looking at your config.<br>
</span></font></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
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</span></font></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">You will need sip to sip, say to access CUE VM from a CCM SIP trunk.<br>
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</span></font><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">Check all G711 etc.<br>
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</span></font><font color="#000080"><font size="2"><font face="Arial"><span style="font-size: 10pt;">Debug CCSIP<br>
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<font size="2"><font face="Tahoma, Verdana, Helvetica, Arial"><span style="font-size: 10pt;"><b>From:</b> <a href="http://ccie_voice-bounces@onlinestudylist.com" target="_blank">ccie_voice-bounces@onlinestudylist.com</a> [<a href="mailto:ccie_voice-bounces@onlinestudylist.com" target="_blank">mailto:ccie_voice-bounces@onlinestudylist.com</a>] <b>On Behalf Of </b>Jonathan Charles<br>
<b>Sent:</b> 06 August 2008 18:41<br>
</span></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font size="2"><font face="Tahoma, Verdana, Helvetica, Arial"><span style="font-size: 10pt;"><br>
<b>To:</b> OSL CCIE Voice Lab Exam; <a href="http://cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> [OSL | CCIE_Voice] IPIPGW Sip to Sip<br>
</span></font></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;"> <br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;">So, I was playing with an IPIPGW<br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;"><br>
<br>
CCM on one side (SIP trunk) and CCME on the other (SIP dial-peer)... call worked, but as soon as you answered it dropped.<br>
<br>
I changed the SIP dial-peer from the IPIPGW to H.323 (no session protocol) and RTP cuts thru fine...<br>
<br>
Am I misreading something, is SIP to SIP not supported, or is my config retarded?<br>
<br>
<br>
<br>
Jonathan<br>
</span></font><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
</span></font><font face="Times New Roman"><span style="font-size: 12pt;"> <br>
</span></font></p></blockquote></div></div><font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size: 11pt;"><br>
<br>
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