<div dir="ltr">Call is from x1008 (on CCM) to x3003 (on CCME)<br><br>On the IPIPGW:<br><br><br>voice service voip <br> allow-connections h323 to h323<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br>
h323<br> modem passthrough nse codec g711ulaw<br> sip<br><br>dial-peer voice 500 voip<br> destination-pattern 1008<br> session protocol sipv2<br> session target ipv4:<a href="http://10.0.0.124">10.0.0.124</a><br> dtmf-relay sip-notify rtp-nte<br>
codec g711ulaw<br>!<br>dial-peer voice 501 voip<br> destination-pattern 3...<br> session target ipv4:<a href="http://10.0.0.131">10.0.0.131</a><br> incoming called-number 3003<br> codec g711ulaw<br>!<br><br>On CCME:<br><br>
<br>dial-peer voice 201 voip<br> destination-pattern 1008<br> session protocol sipv2<br> session target ipv4:<a href="http://10.0.0.63">10.0.0.63</a><br> incoming called-number 3003<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>
<br><br><br><br>Jonathan<br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 12:54 PM, Nguyen Le <span dir="ltr"><<a href="mailto:nguyenbot@gmail.com">nguyenbot@gmail.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;">
<div link="blue" vlink="purple" lang="EN-US">
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<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">You also have under</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Voice service voip</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Allow connections sip to sip ?</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Also, just double check and make sure your SIP Trunk is in a
region that is set to G711 to all other sites</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Nguyen</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> Jonathan Charles
[mailto:<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>] <br>
<b>Sent:</b> Wednesday, August 06, 2008 12:52 PM<br>
<b>To:</b> Nguyen Le<br>
<b>Cc:</b> OSL CCIE Voice Lab Exam; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip</span></p>
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<p> </p>
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<p style="margin-bottom: 12pt;">Yeah, all dial-peers have the
codec hard set to 711ulaw<br>
<br>
<br>
Jonathan</p>
<div>
<p>On Wed, Aug 6, 2008 at 12:43 PM, Nguyen Le <<a href="mailto:nguyenbot@gmail.com" target="_blank">nguyenbot@gmail.com</a>> wrote:</p>
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<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Jonathan – </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Make sure your call codec is
g711</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Nguyen</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div style="border-style: solid none none; border-color: -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <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>On Behalf Of </b>Jonathan Charles<br>
<b>Sent:</b> Wednesday, August 06, 2008 12:41 PM<br>
<b>To:</b> OSL CCIE Voice Lab Exam; <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> [OSL | CCIE_Voice] IPIPGW Sip to Sip</span></p>
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<p> </p>
<div>
<p>So, I was playing with an IPIPGW<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</p>
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<p> </p>
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</blockquote></div><br></div>