It seems like it is a v1.8 only function at present (unless a backport is released).
>From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause ----- Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)} Asterisk 1.8 also comes with a 'use_q850_reason' configuration option for generating and parsing, if available: ----- That will give you what you want if you consider upgrading to v1.8. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 01 March 2011 16:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing Try this - it says it is for 1.8 but might work in 1.6 http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan Sent: Tuesday, March 01, 2011 10:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing SIP_HEADER() gives you only access to headers of the initial INVITE request (and not, for example, the final BYE message) How will I check sip response with this like 404 or 503? -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 01 March 2011 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing -----Original Message----- From: Bob Beers [mailto:bob.be...@gmail.com] Sent: 01 March 2011 13:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Deepika Nijhawan Subject: Re: [asterisk-users] Failover Routing On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan <deepika.nijha...@oxygen8.com> wrote: > Ya, below is my routing: > Exten => 1234,1,Dial(SIP/abc) > Exten => 1234,n,Dial(SIP/xyz) > > If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE} > variable. For this I don't want it to try SIP/xyz. But overall, if we > get SIP 4xx reason then call should hangup like it sends > back 404 not found for this case and if we get SIP 5xx response then should > try SIP/xyz. > Is there any way to check sip responses and do failover routing based > on that? > Have you looked at SIP_HEADER() dialplan function? <https://wiki.asterisk.org/wiki/display/AST/Function_SIP_HEADER> Maybe you can parse Reason header in 4xx or 5xx response? HTH, -Bob -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan Sent: Tuesday, March 01, 2011 9:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Failover Routing It says it for asterisk1.8. I am using asterisk1.6, can we use this function in this version. Is it possible for you to give example on how to use? I just went into my 1.4.37 console and find that SIP_HEADER is there in "Core show functions" so it should be useable in 1.6. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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