On Fri, Aug 12, 2011 at 8:01 AM, cyril constantin < cyril.constan...@gmail.com> wrote:
> Call from sip peers on Asterisk to PSTN Avaya works without issue. > Call from Sipxecs sip peers through Asterisk then using H323 connection to > Avaya extension works > Call from Sipxecs sip peers through Asterisk then using H323 connection to > Avaya PSTN doesn't works, I can ring my mobile phone but when I answer the > call I don't have the voice in both way. > > Codec used is G.711 Alaw. > - check that's the final codec selected in all paths by analyzing the SDP in SIP. check working and not working paths to ensure they are the same - check specified RTP ip addresses and port numbers in the SDP of the SIP messages.
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