David S. Madole
2006-Jan-25 12:31 UTC
[Asterisk-Users] SIP re-invites ignored by other end
Many of my dialplan scenarios involve transferring incoming calls back out to other numbers. For reasons of call quality and bandwidth, I would like for the calls to be reinvite'd to bypass my server with the audio channel. What I am seeing is that my server does indeed send the reinvites, and I get OK responses, but the audio never stops passing through my server. I've been fooling with this for a few days and am somewhat stumped. I would blame the provider on the other end (and still might) except that I see this with multiple providers (in fact none work), so I thought I'd try passing it by here first. Because the file is a little long, I have posted a sample SIP trace at http://madole.net/asterisk/sip-reinvite-prob.txt of a session that involves an incoming call through TelIAX calling back out again through TelIAX. If a SIP expert could take a look and pass along any suggestions, I'd appreciate it. Thanks, David