<div dir="ltr">Yeah, all dial-peers have the codec hard set to 711ulaw<br><br><br>Jonathan<br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 12:43 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">
<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: 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;"> <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>
</div><div><div></div><div class="Wj3C7c">
<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>
</div>
</div></div></div>
</div>
</blockquote></div><br></div>