[asterisk-users] Failover SIP logic

[Jeremy Mann]

Asterisk 1.4.11

Sorry, meant to include that

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrea Spadaccini
Sent: Monday, September 10, 2007 10:59 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Failover SIP logic

Ciao Jeremy,

> I need some extensions logic assistance, I'm trying to dial out one of
> multiple SIP trunks, in sequence.  I need to detect a busy SIP trunk(I only
> allow 1 call per trunk) and roll over to a second or third depending on that
> busy status
>
> Here's what I've got for a macro thusfar, but it's not working(fails if the
> 1st trunk is busy) extensions.conf:
>
> [globals]
> trunk_1 => SIP/trunk1
> trunk_2 => SIP/trunk2
> trunk_3 => SIP/trunk3
>
> [macro-trunkdial]
> exten => s,1,Dial(${trunk_1}/${ARG1})
> exten => s,2,Hangup()
> exten => s,102,Dial(${trunk_2}/${ARG1})
> exten => s,103,Hangup()
> exten => s,203,Dial(${trunk_3}/${ARG1})
> exten => s,204,Hangup()


Which asterisk version are you using?
IIRC, priority jumping (ie. going to n+101) was disabled by default in some
1.2.x version. You should rely on DIALSTATUS. See Dial() page in voip-info.org.

HTH,


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