[Asterisk-Dev] Record app on Sip channel doesn't send RTP

[Tony Mountifield]

I'm doing some SIP testing, using a CVS-HEAD box to initiate calls
via SIP, and then to put each call into the dialplan to execute
Record to capture what it hears:

.call file:

Channel: SIP/remotebox/12345
Extension: s
Context: record
Priority: 1

extensions.conf:

[record]
exten => s,1,Answer
exten => s,2,Record(/tmp/test-%d:wav|0|0)

exten => h,1,NoOp(RECORDED_FILE=${RECORDED_FILE})

I have found, using Ethereal, that while Record is executing, no outgoing
RTP frames are generated. Is this correct behaviour? I would have expected
an RTP stream containing silence, since my understanding is that Asterisk
does not yet support silence suppression.

I've posted this to -dev, since I assume it will be an issue with the code.

Interestingly, when using OH323 instead of SIP, the outgoing stream
continues, but I think OH323 uses its own RTP stack.

I'd be grateful for any comments or suggestions.

One thing I do intend to try is applying the async patch in bug #5374.
Should I expect this to make a difference?

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org