[asterisk-users] evaluate SIP response codes in dialplan

[John covici]

That is very nice, but where are the HANGUPCAUSE values documented?

Thanks.

on Thursday 01/15/2009 Johansson Olle E(oej at edvina.net) wrote
 > 
 > 14 jan 2009 kl. 14.02 skrev Klaus Darilion:
 > 
 > > Hi!
 > >
 > > Is it somehow possible to evaluate the SIP response code inside the
 > > dialplan?
 > >
 > > I have an Asterisk server which forwards requests to various PSTN
 > > gateways with SIP. If the Dial() attempt is not successful I want to
 > > differ at least these 3 options:
 > > - called destination is busy (486): e.g. activate auto-redial
 > > - called destination does not exist, unassigned number (404)
 > > - gateway is broken, error, circuit busy (e.g. 503)
 > >
 > > 486 is mapped to DIALSTATUS=BUSY
 > > but both 503 and 404 is mapped to DIALSTATUS=CONGESTION
 > >
 > > As when Asterisk forwards the response with SIP to the caller the same
 > > response code is used, I suspect this information must be stored
 > > somewhere inside the channel variable. So, are there any means to  
 > > access it?
 > 
 > Check the HANGUPCAUSE, it's much more detailed than DIALSTATUS.
 > 
 > We do map the SIP (and all other protocol errors in various channel  
 > drivers) codes to ISDN hangup causes, which gives you much more  
 > information about
 > why a call failed.
 > 
 > The conversion we're using follows the RFC, and where that doesn't  
 > cover it, Cisco's documentation.
 > 
 > /Olle
 > 
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         covici at ccs.covici.com