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
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@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Simon Brown Sent: 07 June 2004 07:18 To: asterisk-users@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@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
exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXXXXXX,2,Dial(IAX2/username@iax-conf-entry/${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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Simon Brown
> Sent: 07 June 2004 07:18
> To: asterisk-users@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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
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I would use:
exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXXXXXX,2,Dial(IAX2/username@iax-conf-entry/${EXTEN})
exten => _NXXNXXXXXX,3,Congestion
exten => _NXXNXXXXXX,102,1,Busy
exten => _NXXNXXXXXX,103,1,Busy
That way if number you dial is busy it will not immediately try dialing
the same number via IAX. If you want REAL information about what
happened with the call you need to use ${CAUSECODE}.
On Mon, 2004-06-07 at 14:23, John Fraizer wrote:> exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
> exten => _NXXNXXXXXX,2,Dial(IAX2/username@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].
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
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