-----Original Message----- From: asterisk-users-boun...@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-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > jeremy.hellst...@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 [96046642...@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 [96046642...@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 [96046642...@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:testpho...@10.30.20.156>;tag=as60718fca' -- SIP/Iwatsu-00000058 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [96046642...@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
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