exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN}) exten => _NXXNXXXXXX,2,Dial(IAX2/[EMAIL PROTECTED]/${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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Simon Brown Sent: 07 June 2004 07:18 To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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