<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<META NAME="Generator" CONTENT="MS Exchange Server version 6.5.7654.12">
<TITLE>RE: [asterisk-users] Using SIP to dial extension that will give anoutside line</TITLE>
</HEAD>
<BODY>
<!-- Converted from text/plain format -->
<BR>
<BR>
<BR>
<P><FONT SIZE=2>-----Original Message-----<BR>
From: asterisk-users-bounces@lists.digium.com on behalf of Carlos Chavez<BR>
Sent: Tue 8/3/2010 2:17 PM<BR>
To: Asterisk Users Mailing List - Non-Commercial Discussion<BR>
Subject: Re: [asterisk-users] Using SIP to dial extension that will give anoutside line<BR>
<BR>
On Tue, 2010-08-03 at 16:04 -0500, Danny Nicholas wrote:<BR>
> From: asterisk-users-bounces@lists.digium.com<BR>
> [<A HREF="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>] On Behalf Of<BR>
> Jeremy.Hellstrom@synovate.com<BR>
> Subject: [asterisk-users] Using SIP to dial extension that will give<BR>
> anoutside line<BR>
><BR>
><BR>
> <BR>
><BR>
> You could try this:<BR>
><BR>
> <BR>
><BR>
> ; use lwatsu line<BR>
><BR>
> Exten => 1234,1,dial(SIP/3001ww5551212)<BR>
><BR>
> <BR>
><BR>
> If dialing extension SIP/3001 from asterisk connects to the lwatsu<BR>
> with an open line, the ww5551212 will wait 1 second, the dial on using<BR>
> the lwatsu.<BR>
><BR>
> Actually, you nee to dial like this:<BR>
><BR>
>exten => 1234,1,Dial(SIP/lwatsu_sip/${NUMBER})<BR>
><BR>
>lwatsu_sip must be a defined peer in your sip.conf and ${NUMBER} would<BR>
>be the number you wish to dial through that peer. If you need to send<BR>
>the DTMF after the call is connected you can use the D option in the<BR>
>dial command. It is up to the PBX to interpret the number you sent<BR>
>using its internal dialplan.<BR>
><BR>
.<BR>
>--<BR>
>Telecomunicaciones Abiertas de México S.A. de C.V.<BR>
>Carlos Chávez Prats<BR>
>Director de Tecnología<BR>
>+52-55-91169161 ext 2001<BR>
<BR>
<BR>
Thanks all,<BR>
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.<BR>
<BR>
Mostly what I am seeing is ----<BR>
<BR>
*CLI> == Using SIP RTP CoS mark 5<BR>
-- Executing [96046642400@phones:1] Dial("SIP/testphone1-00000053", "SIP/6046642400") in new stack<BR>
== Using SIP RTP CoS mark 5<BR>
[Aug 3 14:41:02] WARNING[1948]: chan_sip.c:5340 create_addr: No such host: 6046642400<BR>
[Aug 3 14:41:02] WARNING[1948]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<BR>
== Everyone is busy/congested at this time (1:0/0/1)<BR>
-- Executing [96046642400@phones:2] Congestion("SIP/testphone1-00000053", "") in new stack<BR>
== Spawn extension (phones, 96046642400, 2) exited non-zero on 'SIP/testphone1-00000053'<BR>
<BR>
or<BR>
<BR>
*CLI> == Using SIP RTP CoS mark 5<BR>
-- Executing [96046642400@phones:1] Dial("SIP/testphone1-00000057", "SIP/Iwatsu/6046642400") in new stack<BR>
== Using SIP RTP CoS mark 5<BR>
-- Called Iwatsu/6046642400<BR>
[Aug 3 14:47:36] WARNING[3239]: chan_sip.c:17865 handle_response_invite: Received response: "Forbidden" from '"TestPhone1" <sip:testphone1@10.30.20.156>;tag=as60718fca'<BR>
-- SIP/Iwatsu-00000058 is circuit-busy<BR>
== Everyone is busy/congested at this time (1:0/1/0)<BR>
-- Executing [96046642400@phones:2] Congestion("SIP/testphone1-00000057", "") in new stack<BR>
== Spawn extension (phones, 96046642400, 2) exited non-zero on 'SIP/testphone1-00000057'<BR>
<BR>
<BR>
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<BR>
</FONT>
</P>
</BODY>
</HTML>