<div dir="ltr">Nope... same behavior...<br><br><br>Jonathan<br><br><div class="gmail_quote">On Wed, Aug 6, 2008 at 1:09 PM, ROZA, Ariel <span dir="ltr"><<a href="mailto:Ariel.ROZA@la.logicalis.com">Ariel.ROZA@la.logicalis.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>
<div dir="ltr" align="left"><span><font size="2" color="#0000ff" face="Arial">Jonathan,</font></span></div>
<div dir="ltr" align="left"><span><font size="2" color="#0000ff" face="Arial"></font></span> </div>
<div dir="ltr" align="left"><span> <font size="2" color="#0000ff" face="Arial">I thinkyou need the command</font></span></div>
<div dir="ltr" align="left"><span></span> </div>
<div dir="ltr" align="left"><span> <font size="2" color="#0000ff" face="Arial">redirect ip2ip under voice service voip (Global)
or under each Dial¨-Peer (specific)</font></span></div>
<div dir="ltr" align="left"><span></span> </div>
<div dir="ltr" align="left"><span><font size="2" color="#0000ff" face="Arial">Regards,</font></span></div>
<div dir="ltr" align="left"><span></span> </div>
<div dir="ltr" align="left"><span> <font size="2" color="#0000ff" face="Arial">Ariel</font> </span></div><br>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><b>From:</b> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Jonathan
Charles<br><b>Sent:</b> Miércoles, 06 de Agosto de 2008 02:56 p.m.<div class="Ih2E3d"><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></div><b>Subject:</b> Re: [cisco-voip] [OSL |
CCIE_Voice] IPIPGW Sip to Sip<br></font><br></div><div><div></div><div class="Wj3C7c">
<div></div>
<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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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 vlink="purple" link="blue" lang="EN-US">
<div>
<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></div>
<div>
<div></div>
<div>
<p> </p>
<div>
<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>
<div>
<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></div>
<div>
<div>
<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></div>
<p> </p></div></div></div></div></div></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div>