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

Carlos Chavez cursor at telecomabmex.com
Tue Aug 3 16:17:59 CDT 2010

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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/48738dde/attachment.pgp 

More information about the asterisk-users mailing list