I'm at a loss. I have completely removed the route pattern and SIP trunk from Call Manager, then added them again, and I get the same results. <br><br>Any ideas anyone?<br><br><div class="gmail_quote">On Tue, Mar 11, 2008 at 8:15 AM, Aaron Fransen <<a href="mailto:aaron.fransen@gmail.com">aaron.fransen@gmail.com</a>> 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 class="Wj3C7c"><br><div class="gmail_quote">On Tue, Mar 11, 2008 at 6:11 AM, Aaron Fransen <<a href="mailto:aaron.fransen@gmail.com" target="_blank">aaron.fransen@gmail.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk.<br><br>Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM.<br>
<br>Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one point, but I suspect that our Telco may have pooched the config somehow, since they're in the process of connecting us to another CCM site).<br>
<br>I have verified: Media Termination point exists, Calling Search Space exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc), and a route pattern exists to take calls to the right trunk.<br><br>
The system will let me complete the dialing sequence to the Asterisk server, but as soon as I enter the last digit I get a busy signal.<br><br>Thoughts anyone?<br><br>Here's my sip.conf if that helps...<br><br>[callman]<br>
type=peer<br>context=incoming<br>insecure=very<br>host=(ip of my call manager server)<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>nat=no<br>canreinvite=yes<br>qualify=yes<br><br>Thanks! Aaron<br>
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