Joshua Colp schrieb: > ----- "Klaus Darilion" <klaus.mailingli...@pernau.at> wrote: > >> Philipp Kempgen schrieb: >>> Klaus Darilion schrieb: >>>> Is it somehow possible to evaluate the SIP response code inside >>>> the dialplan? >>> No. Part of the reasoning is that Asterisk is meant to be a >>> multi- protocol PBX, not a SIP softswitch. >> This is IMO a stupid limitation. There are dozens of ISDN cause >> codes, >> >> dozens of SIP response codes and similar in other protocols, but >> Dial() only exports BUSY or CONGESTION ...... >> > > Right, app_dial condenses down the information it gets into some > basic string representations. You can also access a more specific > Q.931 representation by using the ${HANGUPCAUSE} dialplan variable. > While this is not the SIP response code this gives you more > information. You can also control the SIP response code by passing a
I see. I thought HANGUPCAUSE works only with zaptel. I will give it a try. thanks klaus > Q.931 value to the Hangup() application itself. Unfortunately the > mappings of SIP response code <-> Q.931 are hard coded in chan_sip > though so that is where you can find what maps to what. > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users