Hi! > > Look at qualify= for sip.conf, and consider to extend your > > diaplan for a > > better routing decision with a snippet like this > > Actually, I noticed that setting qualify= alone solves my issue. I > apparently don't require extra dialplan logic because if the peer is > "unreachable" (according to "qualify state") then I guess that Asterisk's > Dial() immediately fails.
That's right, but a) you might want to make a routing decision already before starting Dial() for a smoother handling of calls, and b) the extra code helps to differentiate the case when qualify says all is fine, yet the peer still cannot be reached for whatever reason. Asterisk will translate the SIP error code into a HANGUPCAUSE, and with that translation you loose a lot of information. Philipp -- _____________________________________________________________________ -- 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
