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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...