Hello,
I am trying a setup that is the following:
SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN
Any calls from H.323 GW through GK goes to PSTN, no problem.
SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem.
SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN
phone picks up, no audio on both directions.
PSTN GW support both G.723.1 and G.729. Zultys suposedly supports G.729,
G711u and a.
I Have successfully compiled in Asterisk G.723.1 and G.729 following a mail
from the list, and codecs appears in 'SHOW TRANSLATION'. Also, both
codecs
are configured in H323.conf and sip.conf.
Is there a way to know what is happening on the audio or RTP stream by means
of the asterisk CLI ?
All I know (by protocol analyzer) is that SIP Phone sends stream to
Asterisk, but none goes to PSTN GW.
GK is not doing proxy.
Regards,
Jorge A.
