[clue-tech] Asterisk

[Mike Staver]

Yeah, that sounds about right. I gave up on the AMP gui, it doesn't do 
near enough.  Since I started playing with the conf files, I did get 
internal calls to come in now. The problem is, they all go to one 
extension, completely ignoring my DID inbound routes. They automagically 
go to the ext set for the receptionist that is supposed to answer after 
hours. Even though I have tried in vain to disable this, they still all 
go there. This seems to be a common issue with A at H in this release since 
A at H was written mostly for home users with a single DID, so I'll try to 
find the solution here.

Nate Duehr wrote:
> 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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