19 jan 2009 kl. 11.10 skrev Philipp Kempgen: > Johansson Olle E schrieb: >>> >>> >>> I still think we need a SIP_CAUSE channel variable. :-) >>> >> Then we need to start working on aggregation rules, like what if one >> IAX channel answers and one SIP channel is busy? >> >> For SIP-only calls, we need to add a lot of code from proxy rules for >> call forking and response aggregation. It's not an >> easy task. > > I know it's not an easy task if you'd want it to be done properly. > But then again Asterisk is not a SIP softswitch but a PBX. :-) > I've never seen people who are asking for SIP_CAUSE expect it > to work under all circumstances. All the use cases are pretty > simple: Well, but if we implement a half-done implementation, we will get a ton of bug reports within days... We can't do it like that, Philipp.
(Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) ) /O > > > Dial(SIP/buddy); // single argument > > When dialling to more than 1 SIP peer > > Dial(SIP/busy&SIP/answers_the_call); > > the best thing to do would be to store the last cause code that > we receive i.e. the one of the peer who answered. > > In a multi-protocol situation > > Dial(SIP/busy&IAX/answers_the_call); > > I don't expect SIP_CAUSE to be anything meaningful. It could be > set to "000" or somesuch. > > > Philipp Kempgen > > -- > AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > -- > > _______________________________________________ > -- 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 --- * Olle E Johansson - [email protected] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden _______________________________________________ -- 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
