Hello - <br><br>Thanks for the suggestions, I actually learned since my post that the problem is related to the formation of the initial SIP invite sent from our Sonus gateway through our NexTone SBC. Using your idea regarding the variable I think i've found a work-around.
<br><br>When the Sonus sends the inital invite the format is:<br><br>INVITE sip:315579xxxx;npdi=<a href="mailto:yes@66.218.x.xx">yes@66.218.x.xx</a> SIP/2.0<br><br>It appears the Asterisk server is able to parse this message if I am only using one context, it appears to fail when I use multiple contexts to route my ingress calls.
<br><br>From what I can tell Asterisk parses the invite and looks to send the call to s in the default context (domain 315579xxxx).<br><br>Looking for s in default (domain 315579xxxx)<br><br>If 's' doesn't exist in [315579xxxx] the call chokes.
<br><br>If I create a variable based on the ingress 'domain' using:<br><br>exten => s,1,Goto(${SIPDOMAIN},s,1)<br><br>The call is sent to the correct context.<br><br>I think it's a work-around, but it seems to do the trick.
<br><br> -- Executing Goto("SIP/15241-08198868", "315579xxxx|s|1") in new stack<br> -- Goto (315579xxxx,s,1)<br> -- Executing Answer("SIP/15241-08198868", "") in new stack<br>
<br><br>Notes on Digium:<br><br><a href="http://bugs.digium.com/view.php?id=7208&nbn=24">http://bugs.digium.com/view.php?id=7208&nbn=24</a><br><br><br><br><br><br><br><br><br><br><br><br><div><span class="gmail_quote">
On 6/28/06, <b class="gmail_sendername">El Flynn</b> <<a href="mailto:el_flynn@lanvik-icu.com">el_flynn@lanvik-icu.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Christopher Aloi wrote:<br>> Hello -<br>><br>> I currently have 10 DID's coming into one Asterisk server, I seem to be<br>> having some difficulty routing based on the DID dialed and am hoping<br>> someone<br>
> on the list can assist me.<br>><br><snip><br><br>Unless I'm misunderstanding you, how about trying this:<br><br>1. In your sip.conf:<br><br>[general]<br>useragent=Asterisk<br>port=5060<br>context=default<br>
tos=lowdelay<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br>rtptimeout=300<br>rtpholdtimeout=600<br><br>2. In your extensions.conf:<br><br>[default]<br>exten => s,1,Goto(${CALLERIDNUM},s,1)<br><br>[123456789]
<br>exten => s,1,Answer()<br>exten => s,2,Playback(beep)<br>exten => s,3,GoTo(queue-test,s,1)<br><br><br>So if you get an incoming SIP call from 123456789, it enters the "default"<br>context and is then routed to the "123456789" context.
<br><br>Flynn<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
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