[asterisk-users] Debugging SIP call hangup reasons

[James Lamanna]

Hi,
Is there any information that can be gathered from the logs about why
a SIP call was dropped/terminated without either side hanging up?
I've run asterisk pretty verbose and I guess I haven't seen anything
that pops out at me yet.
I'm trying to diagnose why some clients are getting dropped calls
every so often.

Thanks.

-- James