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senario
2003 Aug 03
0
RTP / SIP routing issues
...Is there someway to better debug the 'native
bridge'?
Going the other way (from SIP to the PSTN) I can hear the audio from the SIP
client over the PSTN, but I can't hear the PSTN audio comming back to the
SIP client.
Is anyone running a private SIP gateway behind asterisk like in this
seniario? What needs to be done to get audio going both ways? Any hints?
Thanks in advance,
M. Gamble