Critical issues are when SIP error come after 18X provisional response. - if bypass_media is false then dialplan stops and leg a is explicitly hang up (switch_ivr_bridge.c, line 513). - if bypass_media is true, then dialplan continue, but there is 10 seconds delay before next bridge application sends INVITE to gateway ( http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.h... ) I didn't track down yet why this happens looking at FS sources. Why I didn't open a bug on jira? Because FS behaves according to the design and specs :-) But not according to real world requirements... On Wednesday 13 January 2010, Brian West wrote:
Can you elaborate on these "Critical" issues you seem to be having? Why aren't you opening a jira for them if they are that critical to your needs?
/b
On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:
If you wish FreeSWITCH to be well adopted by the community, then FS should follow the real world "rules" but not specs...
Don't you agree that 99% of SIP servers are set up to interconnect with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
Some background - I did run asterisk for years, but switched to FS recently because of critical asterisk problems with SIP handling when asterisk is not in the media path. I spend a lot of time porting the billing system to FS. And I did it. What I got? Critical problems with FS early media handling. Hopefully I can switch back to asterisk if FS problems with early media will begin to draw customers away.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org