[Asterisk-Users] Dial plan help

[John Fraizer]

exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXXXXXX,2,Dial(IAX2/username at iax-conf-entry/${EXTEN})
exten => _NXXNXXXXXX,3,Congestion

The above will attempt to dial out your Zap interface first.  If that 
fails, it will dial out using "username" for the username and the 
password, IP address info for the IAX2 peer will be grabbed out of the 
iax2.conf entry that matches [iax-conf-entry].

Works for us.

John


usedcanon wrote:

> Checkout http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
> and http://www.voip-info.org/wiki-Asterisk+t+extension
> 
> You could use extention t, which is reached after dial times out.
> 
> Umar.
> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Simon Brown
> Sent: 07 June 2004 07:18
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Dial plan help
> 
> 
> I wish to have outgoing calls try to use a SIP/IAX provider and if this
> fails, then fall back to PSTN and I am not sure how the dial plan should
> look.
> 
> Can someone please post a sample of how it should look.
> 
> Thanks in advance,
> 
> Simon Brown
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users