I run FreeSWITCH Version 1.0.5pre10 (16012M), sorry but I don't feel comfortable running SVN trunk on production servers... Perhaps because of my experience with asterisk. Neither version after 1.4.18 did work OK to me. On Wednesday 13 January 2010, Brian West wrote:
This is exactly why you open a jira. But we have zero control over the far end sending a 18X then an error. That is NOT something we can fix. As for the others I suspect they have been fixed... what Rev are you running?
/b
On Jan 13, 2010, at 10:08 PM, Sergey Okhapkin wrote:
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/02435 4.html ) 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...