Callback and bridge Problem, revived


I’ve seen others post about this, so I’m guessing I’m not the only one with this problem. The fact that I’m using MS Speech server may be a red herring.

I’ve heard there is a bug with sending a REFER from MS Speech Server to Asterisk. Here’s what happens:

[ul]Inbound call connects to MSS.
MSS sends a refer.
Asterisk dials the new destination.
The destination answers. There is roughly 1 second of dialog.
Asterisk disconnects the new destination.[/ul]

In the Asterisk log, there is this entry (the cause?)

[quote]chan_sip.c Native bridging SIP/XXXX and SIP/YYYY
logger.c Re-invite to non-existing call leg on other UA. SIP dialog ‘[some guid]’. Giving up.
rtp.c Got a FRAME_CONTROL (8) frame on channel SIP/XXXX
logger.c Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on ‘SIP/XXXX’ in macro ‘dialout-trunk’[/quote]

I’ve seen other posts about this, but no resolution.

Any ideas?