Luki wrote: > Hi all, > > this is probably a asterisk-devel question but I'll try it here first. > > Is there a way to delay a ReInvite on SIP? I have an issue where my > provider's server is only ~1 ms RTT away and for about 1/3 of the > incoming calls I get a "482 Loop Detected" error because the ReInvite > is processed by the calling server before the ACK packet on Answer(). > Asterisk ReInvites right after answering and both packets leave my end > virtually simultaneously (within 0.1 ms based on time stamp in > tcpdump). I looked at the code in CVS and the 482 Loop Detected > message is sent back when an Invite comes in with a call ID of a > pending outgoing invite that has not yet been answered (at least this > is how I understand it) -- and in this case it would be a loop. > > It's not a major issue because the calling end re-tries the Invite in > a second and then it usually works. This is quite reproducible on my > setup and I can provide tcpdump captures if anyone has some ideas. > Which version of Asterisk are you using? I have a vague memory of fixing this in CVS head, but I might be wrong. Can't really check here, sorry. Test with CVS head, and if you still have problems please open a bug report in the bug tracker at bugs.digium.com with the call trace.
THank you. /Olle _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users