Jeremy.Hellstrom at synovate.com
2010-Aug-03 20:57 UTC
[asterisk-users] Using SIP to dial extension that will give an outside line
I am trying to add an Asterisk box to an Iwatsu ECS (Software Version 7.0 R.01) hopefully without using a physical T1/E1 card. Internally the SIP works fine, it is dialling an outside line that is giving me difficulties. One way that I think it might be possible is for an outbound call to connect to extenstion 3001, which is one of 36 PRI trunks available to the Iwatsu system. Dialling ext 3001 on an Iwatsu immediately gives an Iwatsu phone an open outbound line. I cannot figure out a way to define that extension in the dial plan when I enter the Iwatsu as a channel. Am I totally barking up the wrong tree with this method? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/0803fc15/attachment.htm
Danny Nicholas
2010-Aug-03 21:04 UTC
[asterisk-users] Using SIP to dial extension that will give anoutside line
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/bb1c1943/attachment.htm
Carlos Chavez
2010-Aug-03 21:17 UTC
[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 M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +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