[Asterisk-Users] SIP re-invites ignored by other end

[David S. Madole]

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