<div dir="ltr">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><div class="gmail_quote">On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson <span dir="ltr"><<a href="mailto:scollinson@capewave.co.uk">scollinson@capewave.co.uk</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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<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">SIP to SIP should work fine, when
configured correctly.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">I was just trying to give you a scenario
where we may need to use it. Apologies if this was not helpful</span></font></p>
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<p><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"> Jonathan Charles
[mailto:<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>] <br>
<b><span style="font-weight: bold;">Sent:</span></b> 06 August 2008 19:55<br>
<b><span style="font-weight: bold;">To:</span></b> Stephen Collinson<br>
<b><span style="font-weight: bold;">Cc:</span></b> cisco voip; OSL CCIE Voice Lab
Exam<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [OSL | CCIE_Voice]
IPIPGW Sip to Sip</span></font></p>
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<p style="margin-bottom: 12pt;"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">Perhaps I wasn't clear...<br>
<br>
<br>
There is no CUE.<br>
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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</span></font></p>
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<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;">On Wed, Aug 6, 2008 at 1:52 PM, Stephen Collinson <<a href="mailto:scollinson@capewave.co.uk" target="_blank">scollinson@capewave.co.uk</a>>
wrote:</span></font></p>
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<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">Perhaps worth looking at your config.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">You will need sip to sip, say to access CUE VM from a CCM SIP
trunk.</span></font></p>
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<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">Check all G711 etc.</span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">Debug CCSIP</span></font></p>
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<p><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"> <a href="mailto:ccie_voice-bounces@onlinestudylist.com" target="_blank">ccie_voice-bounces@onlinestudylist.com</a>
[mailto:<a href="mailto:ccie_voice-bounces@onlinestudylist.com" target="_blank">ccie_voice-bounces@onlinestudylist.com</a>]
<b><span style="font-weight: bold;">On Behalf Of </span></b>Jonathan Charles<br>
<b><span style="font-weight: bold;">Sent:</span></b> 06 August 2008 18:41</span></font></p>
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<p><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"><br>
<b><span style="font-weight: bold;">To:</span></b> OSL CCIE Voice Lab Exam; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b><span style="font-weight: bold;">Subject:</span></b> [OSL | CCIE_Voice] IPIPGW
Sip to Sip</span></font></p>
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<p><font size="3" face="Times New Roman"><span style="font-size: 12pt;">So, I was
playing with an IPIPGW</span></font></p>
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<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</span></font></p>
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