[clue-tech] Asterisk
Nate Duehr
nate at natetech.com
Wed Feb 22 15:14:30 MST 2006
Mike Staver wrote:
> I have Asterisk at Home up and running sort-of successfully. I can get
> outbound calls to work great, that's not a problem at all. The inbound
> calls are a problem however. My setup is going to include around 30
> different DID's each being routed to a separate extension. For a simple
> test, I have one trunk setup and I have one incoming route defined. I
> want the number 3033960044 to go to a specific extension. It sounds
> simple, for some reason I'm getting this error message each time:
>
> Feb 22 13:53:33 DEBUG[2473] chan_sip.c: Setting NAT on RTP to 0
> Feb 22 13:53:33 DEBUG[2473] chan_sip.c: Checking SIP call limits for
> device 3033960044
> Feb 22 13:53:33 DEBUG[2473] chan_sip.c: build_route: Contact hop:
> <sip:720236XXXX at 66.185.96.28>
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Executing
> AbsoluteTimeout("SIP/3033960044-7fb9", "15") in new stack
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Set Absolute Timeout to 15
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Executing
> Congestion("SIP/3033960044-7fb9", "") in new stack
> Feb 22 13:53:33 VERBOSE[5061] logger.c: == Spawn extension
> (from-sip-external, s, 2) exited non-zero on 'SIP/3033960044-7fb9'
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Executing
> AbsoluteTimeout("SIP/3033960044-7fb9", "15") in new stack
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Set Absolute Timeout to 15
> Feb 22 13:53:33 VERBOSE[5061] logger.c: -- Executing
> Congestion("SIP/3033960044-7fb9", "") in new stack
> Feb 22 13:53:33 VERBOSE[5061] logger.c: == Spawn extension
> (from-sip-external, h, 2) exited non-zero on 'SIP/3033960044-7fb9'
> Feb 22 13:53:33 DEBUG[5061] cdr_addon_mysql.c: cdr_mysql: inserting a
> CDR record.
> Feb 22 13:53:33 DEBUG[5061] cdr_addon_mysql.c: cdr_mysql: SQL command as
> follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,c$
> Feb 22 13:53:33 DEBUG[5061] chan_sip.c: update_call_counter(3033960044)
> - decrement call limit counter
>
> It's the "-- Executing Congestion("SIP/3033960044-7fb9", "") in new
> stack" part that's got me concerned. When I make a call to the number,
> I ofcourse get the dreaded busy signal.
>
> Does anybody know how to get multiple DID's routing to specific
> extension with A at H?
I don't, 'cause I haven't even played with it yet, but a friend of mine
seems to understand it.
He says:
"Yes and no. You have to create a different context for each channel on
the digium card, and then create a different incoming route for each. I
don't think you can do it with the current AMP GUI on A at H. Have to dive
into the .conf files. There is a web based editor in the maintenance
tab. It will probably break the current A at H setup."
Nate
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