<div>Yep, have that already. Gig0/1.110 has the IP address configured on the SIP trunk in CUCM. </div>
<div> </div>
<div>voice service voip<br> sip<br> bind control source-interface GigabitEthernet0/1.110<br> bind media source-interface GigabitEthernet0/1.110<br></div>
<div> </div>
<div>I also have the following:</div>
<div> </div>
<div>voice class codec 1<br> codec preference 1 g711ulaw<br> codec preference 2 g729r8<br></div>
<div>dial-peer voice 100 voip<br> destination-pattern ....<br> session protocol sipv2<br> session target ipv4:172.21.20.10<br> voice-class codec 1<br> dtmf-relay rtp-nte<br> no vad<br>dial-peer voice 1 pots<br> incoming called-number .<br>
direct-inward-dial<br></div>
<div><br><br> </div>
<div class="gmail_quote">On Mon, Apr 26, 2010 at 3:58 PM, Ahmed Elnagar <span dir="ltr"><<a href="mailto:ahmed_elnagar@rayacorp.com">ahmed_elnagar@rayacorp.com</a>></span> wrote:<br>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">You have to bind signal and media traffic out of the router to the CUCM with the IP address you have configured on CUCM “by default CUCM reject calls with source address other than the configured”</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">Try the below on the gateway:</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">voice service voip</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">sip</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">bind all source-interface “interface configured on CUCM”</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<div>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 10pt"> </span><span style="COLOR: #1f497d; FONT-SIZE: 10.5pt">Best Regards;</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 10.5pt"> Ahmed Elnagar</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 10.5pt"> Senior Network PS Engineer</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 10.5pt"> Mob: +2019-0016211</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 10.5pt"> <img alt="ccie_voice_large.gif" src="cid:image001.jpg@01CAE592.A86AADF0" width="63" height="63"><img alt="ccvp_voice_large.gif" src="cid:image002.jpg@01CAE592.A86AADF0" width="63" height="63"></span></p>
</div>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
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<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt"> <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>Brian Schultz<br>
<b>Sent:</b> Monday, April 26, 2010 10:34 PM<br><b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> [cisco-voip] SIP gateway config</span></p></div>
<div class="im">
<p class="MsoNormal"> </p>
<div>
<p class="MsoNormal">Does anyone happen to have an example SIP gateway config for an ISR? CUCM 8.0(2) with a SIP trunk to a 2921 gateway (15.0.M1.12) with a standard PRI for PSTN access. I have outbound working, but inbound gives a fast busy with a 401 Unauthorized in the SIP debug.</p>
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<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal">Thanks,</p></div>
<div>
<p class="MsoNormal">Brian</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"> </p></div></div></div>
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