Johansson Olle E schrieb: > 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. I guess you're right. Give them an inch and they will request a mile. SIP_CAUSE_HALFBAKED could do the trick. ;-) > (Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) ) ;-) 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