[Asterisk-Users] Vegastream assistance?

[Neil Bullock]

I wonder if anyone can help me?

Am trying to terminate to H323 Vegastream. I'm using OH323 with little
success.

I can dial out and answer but voip end just keepings ringing and ringing.

Thanks for any help.

Neil

Config file:
[general]
listenAddress=ALL
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=no
h245Tunnelling=no
h245inSetup=no
jitterMax=100
outboundMax=100
inboundMax=100
simultaneousMax=200
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/tmp/oh323_debug.log
gatekeeper=DISABLE
gatekeeperTTL=300
userInputMode=TONE
amaFlags=default
accountCode=H323
context=sip

[register]
alias=ASTERIX

[codecs]
;   G711U       -   G.711 u-Law
;   G711A       -   G.711 A-Law
;   G7231       -   G.723.1(6.3k)
;   G72316K3    -   G.723.1(6.3k)
;   G72315K3    -   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G728        -   G.728
;   G729        -   G.729
;   G729A       -   G.729A
;   G729B       -   G.729B
;   G729AB      -   G.729AB
;   GSM0610     -   GSM 0610
;   MSGSM       -   Microsoft GSM Audio Capability
;   LPC10       -   LPC-10
codec=G729
codec=G7231
codec=G711A
codec=G711U