[Asterisk-Users] H323 channel

[T. Chan]

Dear All,

There is a question about the H323 channels (H323 driver, not OH323), it is
not passing CallerID. If a call comes in on ZAP and out H323 to another
gateway, the other gateway does not see the ANI, and if Asterisk is used as
a passthrough, it receives callerID from the other gateway, but when sent
out to the terminating endpoint, the terminating endpoint is not seeing it.

Is there anywhere we should configure so that it is passing the callerID to
a terminating endpoint on the outbound H323 channels. Please enlighten me,
thanks

TC

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris A.
Icide
Sent: Tuesday, July 06, 2004 6:31 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] H323 channel


On 03:23 AM 7/6/2004, administrator tootai wrote:
 >Hello everybody,
 >
 >my * box is connected to gnugk with H323 channel. If I call from an H323
 >EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
 >start but noisy (scratch) , then became ok for callee (SIP EP) but still
 >scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
 >EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
 >
 >No need to say that H323<->H323 is working, as well as SIP<->SIP.
 >Running CVS version from yesterday. Used codecs are G711U & A, G723.1
 >and G729. If I just use G711 it's the same. SIP EP has to call first
 >when * is started to make it work. Any hint?
 >
 >Also, H323 is still broken and working without FastStart. Is there a
 >workaround existing?

Just to help troubleshooting, which h323 implementation for asterisk did
you use?


-Chris

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