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
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