Hi
everyone
I have successfully
compiled and installed OH323 support (finally) into my
Asterisk.
I want to connect
the Asterisk server to our Alcatel OmniPCX Office (OXO) PABX, which has an
internal H.323 gateway.
I have created the
correct dialplans in Asterisk and same in OXO.
The OXO only
supports G711a G711u G729 and G723.1 codecs.
When I call from a
SIP phone to OXO using my Grandstream 100 handset with PCMA as the first
priority codec, I get perfect speech.
When I call from the
Alcatel to the Grandstream, I get one way speeche, i.e. I can hear the person on
the Alcatel handset, but they can't hear me.
Is there any way to
debug the connections so as to see what codecs are used in asterisk? I can see
all the call setup info with debug on, but I can't see the codec info. It seems
strange that the call will only work in one direction as the Alcatel can
obviously find a compatible codec when the call is initiated
inbound.
Any ideas greatly
appreciated.
Regards
Mark
Dutton.
_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users