All,
I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file, and my dialplan syntaxes don't appear to be liked by the asterisk-gui program (not a big deal, was just something shiny to look at for me, to try to figure out a way to get this going).
What my problem is with Asterisk is my SPA-3201 is my primary voice gateway, as I do not own any Digium hardware, and currently do not have a SIP provider outside of my PBX at home. When I restart Asterisk, everything works perfectly. I let Asterisk sit for an hour or so, and it stops allowing calls to be routed into the assigned extension. I do see stuff from the communications, at the time the call lands on the Asterisk server:
== Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6The logic is that the SPA is registered as an extension on my system, and incoming calls are routed into the system VIA that extension. The dialplan that the SPA connects to is:
[gw8028]
exten => 8028,1,Answer
exten => 8028,n,Set(CallerNum=${CALLERID(num)})
exten => 8028,n,Set(CallerName=${CALLERID(name)})
exten => 8028,n,Set(CDR(accountcode)="8203")
exten => 8028,n,Set(CDR(UserField)="POTS")
exten => 8028,n,Goto(from-internal,111,1)
exten => 8028,n,Hangup
the 'from-internal' is my current call filtering/processing subsystem.
The outbound side of this works just fine though, as well as my ATA's
and Cisco 7960's are able to make and receive calls when this is
happening. I can include any additional details if requested, as I don't
know exactly what would be helpful to others with this. The SPA itself
hasn't been changed in seven months, and is stable with Callweaver.
Thanks in Advance, Seann Clark
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