[clue-tech] Asterisk

[Mike Staver]

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?

Jim Ockers wrote:
> Hi Chris,
> 
> Chris Hirsch wrote:
>> If you want to do fax receiving to real fax machines you want to look
>> into the FXS cards (or modules) that Digium sells. Then you can plug fax
>> machines into those and Asterisk will detect fax tones and route faxes
>> there. Asterisk is also capable of receiving faxes itself and routing
>> them to email addresses. A couple of years ago I had this working with
>> about half of the incoming faxes (older faxes seemed to work better than
>> newer) but it wasn't reliable. I've heard that major advances have
>> happened here and things should be better now. I'm actually looking into
>> this and should hopefully have a better answer here in the next few weeks.
> 
> Could you please elaborate on how you had this working?  What did you
> do to receive faxes into Asterisk?
> 
> I knew Asterisk could recognize a fax tone on an incoming call and
> transfer it to another station like a fax machine (or faxmodem) plugged 
> into an FXS card.  Is this what you did?
> 
> I've looked at t38modem but it seemed difficult to get working; were
> you successful in doing this?  If so how did you get it to work?  For
> the unfamiliar I think t38modem could run as a "channel" in Asterisk
> and is a software implementation of a fax dsp.  Therefore Asterisk
> could 
> 
> I receive faxes on the same box that I run Asterisk but I use 
> mgetty+sendfax to run the external faxmodem, which is not plugged 
> into anything that has anything to do with Asterisk.  mgetty+sendfax 
> e-mails me the faxes, not Asterisk. Of course Asterisk e-mails me 
> voicemails.  I'm interested to know how you combined the faxing with 
> Asterisk.
> 
> Thanks,
> Jim
> 
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