Using alaw or ulaw doesn't appear to make a difference. I did make a screw up on the sip.conf, and it's fixed now but still nothing.<br><br>I do in fact have Astra 57i SIP phones attached to Asterisk, and they can call both Call Manager extensions and external phones (via Call Manager!) no problem.<br>
<br><div class="gmail_quote">On Tue, Mar 11, 2008 at 6:38 AM, Peter Pauly <<a href="mailto:ppauly@gmail.com">ppauly@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;">
I've noticed two differences in what you described and my working CM setup:<br>
<br>
1. My sip trunk in CM is defined as 711alaw, you have ulaw.<br>
<br>
2. My sip.conf defines CM as a type=friend instead of a peer.<br>
<br>
Do you have any SIP phones connected to Asterisk (you could use a<br>
softphone like the free xten)? Can you call the phone from<br>
CallManager?<br>
<font color="#888888"><br>
Peter Pauly<br>
<a href="http://www.usbtests.com" target="_blank">http://www.usbtests.com</a><br>
</font><div><div></div><div class="Wj3C7c"><br>
<br>
On 3/11/08, Aaron Fransen <<a href="mailto:aaron.fransen@gmail.com">aaron.fransen@gmail.com</a>> wrote:<br>
><br>
> Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1,<br>
> Asterisk is running strictly VoIP over the network and using CCM as the<br>
> trunk.<br>
><br>
> Calls from the SIP phones connected to Asterisk work fine. They can call<br>
> 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<br>
> DID have this working at one point, but I suspect that our Telco may have<br>
> pooched the config somehow, since they're in the process of connecting us to<br>
> another CCM site).<br>
><br>
> I have verified: Media Termination point exists, Calling Search Space<br>
> exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc),<br>
> 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<br>
> 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>
><br>
</div></div></blockquote></div><br>