| Summary: | ASTERISK-02256: rtptimeout and canreinvite=yes | ||
| Reporter: | adomjan (adomjan) | Labels: | |
| Date Opened: | 2004-08-23 02:15:03 | Date Closed: | 2008-01-15 15:05:31.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | When I enable canreinvite and I make call with 2 sip phone asterisk will disconnect after rtptimeout expire. It's ok but unusable rtptimeout+reinvite. I would happy if I can disable it for sip-sip connections with reinvite (it's tollfree). | ||
| Comments: | By: Mark Spencer (markster) 2004-08-23 10:29:33 Fixed in CVS. Thanks. By: Digium Subversion (svnbot) 2008-01-15 15:05:31.000-0600 Repository: asterisk Revision: 3635 U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r3635 | markster | 2008-01-15 15:05:30 -0600 (Tue, 15 Jan 2008) | 2 lines Qualify rtptimeout with a reinvite having taken place (bug ASTERISK-2256) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=3635 | ||