[asterisk-users] Using SIP to dial extension that will give anoutside line

[Jeremy.Hellstrom at synovate.com]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com on behalf of Carlos Chavez
Sent: Tue 8/3/2010 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Using SIP to dial extension that will give anoutside line
 
On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Jeremy.Hellstrom at synovate.com
> Subject: [asterisk-users] Using SIP to dial extension that will give
> anoutside line
> 
> 
>  
> 
> You could try this:
> 
>  
> 
> ; use lwatsu line
> 
> Exten => 1234,1,dial(SIP/3001ww5551212)
> 
>  
> 
> If dialing extension SIP/3001 from asterisk connects to the lwatsu
> with an open line, the ww5551212 will wait 1 second, the dial on using
> the lwatsu.
> 
>	Actually, you nee to dial like this:
>
>exten => 1234,1,Dial(SIP/lwatsu_sip/${NUMBER})
>
>lwatsu_sip must be a defined peer in your sip.conf and ${NUMBER} would
>be the number you wish to dial through that peer.  If you need to send
>the DTMF after the call is connected you can use the D option in the
>dial command.  It is up to the PBX to interpret the number you sent
>using its internal dialplan.
>
.
>-- 
>Telecomunicaciones Abiertas de México S.A. de C.V.
>Carlos Chávez Prats
>Director de Tecnología
>+52-55-91169161 ext 2001


Thanks all, 
Unfortunately it is only the Iwatsu IP phones that grab the open line @ 3001 currently, the softphones do not.  I might try programming the extension and see if I can get a response that way.

Mostly what I am seeing is ----

*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [96046642400 at phones:1] Dial("SIP/testphone1-00000053", "SIP/6046642400") in new stack
  == Using SIP RTP CoS mark 5
[Aug  3 14:41:02] WARNING[1948]: chan_sip.c:5340 create_addr: No such host: 6046642400
[Aug  3 14:41:02] WARNING[1948]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [96046642400 at phones:2] Congestion("SIP/testphone1-00000053", "") in new stack
  == Spawn extension (phones, 96046642400, 2) exited non-zero on 'SIP/testphone1-00000053'

or

*CLI>   == Using SIP RTP CoS mark 5
    -- Executing [96046642400 at phones:1] Dial("SIP/testphone1-00000057", "SIP/Iwatsu/6046642400") in new stack
  == Using SIP RTP CoS mark 5
    -- Called Iwatsu/6046642400
[Aug  3 14:47:36] WARNING[3239]: chan_sip.c:17865 handle_response_invite: Received response: "Forbidden" from '"TestPhone1" <sip:testphone1 at 10.30.20.156>;tag=as60718fca'
    -- SIP/Iwatsu-00000058 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [96046642400 at phones:2] Congestion("SIP/testphone1-00000057", "") in new stack
  == Spawn extension (phones, 96046642400, 2) exited non-zero on 'SIP/testphone1-00000057'


Dependent on defining Iwatsu as a friend in the latter or as a variable in the former.  By the way Exten => 1234,1,dial(SIP/3001ww5551212) had asterisk return No such host: 3001ww5551212
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/a95df432/attachment.htm