[Asterisk-Users] oh323 <-> sip communication problem

[Paweł Gołaszewski]

I've got problem with connections h323 -> sip and sip -> h323.

I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As 
gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5


When I call from Cisco (SIP) to h323 node by alias registered on 
gatekeeper and h323 node will answer the phone... I have on my Cisco still 
Ringing. Call termination, no matter from which side works fine.

koga*CLI> oh323 show info
koga*CLI> 
^^^^^^^^^^^^^^^^^^^^^^^ why is here empty line? :)
Information about active OpenH323 channel(s)
--------------------------------------------
 Num. Token                          State   Init      RX/TX   Format       
Remote RTP Addr.      Local RTP Addr.          0 ip$localhost/21538             
RING    Local      0/160  NULL         0.0.0.0:0             0.0.0.0:0            
 
koga*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data           
     H323:21538  (voip-h323  s            1   ) Ringing AppDial       (Outgoing Line)
  SIP/blue-ebfa  (default    marosin      2   )    Ring Dial          OH323/marosin  


I can't call from h323 phone to sip. I get that user is not registered on 
gatekeeper...

My configuration:
oh323.conf:

[register]
alias=asterisk
alias=123
alias=blue
alias=blues
;alias=marosin


extensions.conf:

[voip-h323]
exten => marosin,1,Ringing
exten => marosin,2,Dial,OH323/marosin
exten => marosin,3,Hangup


marosin is h323 phone
blues and blue are sip phones.

-- 
pozdr.  Paweł Gołaszewski 
---------------------------------
worth to see: http://www.againsttcpa.com/
CPU not found - software emulation...