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datacomms
2005 Jul 26
2
sip+oh323 - no voice at sip side
Hello,
I have something like this:
SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN
After calling from SIP to PSTN (and from PSTN to SIP too)
I can't hear anything only in my SIPUSER. At the PSTN side everything is OK.
I have another network with another h323/sip (in the place of asterisk)
and there everything is OK.
In AUDIOCODES logs I see that everything goes