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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Cause code 16 is normal call clearing. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Jonathan Charles<br>
<b>Sent:</b> Wednesday, August 06, 2008 3:48 PM<br>
<b>To:</b> Chris Ward<br>
<b>Cc:</b> OSL CCIE Voice Lab Exam; cisco voip; Stephen Collinson<br>
<b>Subject:</b> Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>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<o:p></o:p></p>
<div>
<p class=MsoNormal>On Wed, Aug 6, 2008 at 2:45 PM, Chris Ward <<a
href="mailto:chrward@cisco.com">chrward@cisco.com</a>> wrote:<o:p></o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif"'>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:</span><o:p></o:p></p>
<ol start=1 type=1>
<li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
mso-list:l0 level1 lfo1'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>What
is the version of CUCM? </span><o:p></o:p></li>
<li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
mso-list:l0 level1 lfo1'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Do
calls fail in both directions? </span><o:p></o:p></li>
<li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
mso-list:l0 level1 lfo1'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>What
is the region setting (what codec) between the CUCM IP Phone and the SIP
trunk on CUCM? </span><o:p></o:p></li>
<li class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
mso-list:l0 level1 lfo1'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Is
the CUCM SIP trunk doing early media (is the "MTP required"
checkbox checked)? </span><o:p></o:p></li>
</ol>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif"'><br>
-Chris<o:p></o:p></span></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
<hr size=3 width="95%" align=center>
</span></div>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:
</span></b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>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<o:p></o:p></span></p>
<div>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><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:<o:p></o:p></span></p>
</div>
</div>
<div>
<div>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:navy'>SIP to SIP should work fine, when
configured correctly.<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>I was just trying to give you a scenario where we may need to use
it. Apologies if this was not helpful<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
<br>
</span><o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'>
<hr size=2 width="100%" align=center>
</div>
<p style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> 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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span>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 style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span>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 style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Perhaps worth looking at your config.<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>You will need sip to sip, say to access CUE VM from a CCM SIP
trunk.<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Check all G711 etc.<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Debug CCSIP<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'><br>
<br>
</span><o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'>
<hr size=2 width="100%" align=center>
</div>
<p style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> <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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span>So, I was playing with an IPIPGW<br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><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>
<br>
<o:p></o:p></p>
</blockquote>
</div>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif"'><o:p> </o:p></span></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
<hr size=3 width="95%" align=center>
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<p class=MsoNormal><span style='font-size:10.0pt;font-family:Consolas'>_______________________________________________<br>
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