Hi,Enable 'sip debug' at the CLI and send some detailed log file.
I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating.
Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards.
There is nothing in the logs and nothing on the console, the call just seems to 'go away'!
/O
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