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74dd
2003 Aug 03
0
RTP / SIP routing issues
...SIP working through Asterisk without any problems
(using various sip clients).
When I call from the PSTN to the CSG, here is what I see in the asterisk
console:
-- Executing Dial("SIP/10.10.100.40:5060", "SIP/mgamble") in new stack
-- Called mgamble
-- SIP/mgamble-7fdd is ringing
-- SIP/mgamble-7fdd answered SIP/10.10.100.40:5060
-- Attempting native bridge of SIP/10.10.100.40:5060 and
SIP/mgamble-7fdd
The SIP/mgamble extention rings, however, when I pick up the phone I get no
audio in ether direction. Is there someway to better debug the 'native
br...