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.