Asterisk-1.6.2_00h323 - Avaya(Nortel CS1000 Release 7.5)

Hi support,

I have installed an asterisk:

Asterisk-1.6.2.22
Asterisk-addons-1.6.2.2

Avaya(Nortel CS1000 Release 7.5)

      I have configured ooh323 (ooh323.conf). from nortel to asterisk calls are successful (signs - voice), but the calls originated from asterisk to nortel marked properly (setup - > ringing - > anwerd) in this last point when the call is answered by the enpoint this is finished unexpectedly, were catches of the traffic, and the nortel ingeniro indicates that in the messaging that reaches towards the nortel does not the origin of the signallig server ip address, or the source of the media server ip. attached you diagrams and screenshots.

I thank in advance for his reply.

I have the information in the link

diagram : imageshack.us/photo/my-images/18/diagramf.png/

Capure from the asterisk : imageshack.us/photo/my-images/80 … na001.png/

Capture from the nortel

.16:03:28 20/04/2012
VTN 100 1 04 27
KEY 0 SCR MARP ACTIVE VTN 100 1 04 27

ORIG VTN 096 0 01 00 VTRK IPTI RMBR 60 1 INCOMING VOIP GW CALL
FAR-END H.323 SIGNALLING IP: 0.0.0.0 ; no IP signalling
FAR-END MEDIA ENDPOINT IP: 0.0.0.0 PORT: 0 ; no IP media
FAR-END VendorID: objsys;v0.8.3
TERM VTN 100 1 04 27 KEY 0 SCR MARP CUST 0 DN 2735 TYPE 2050PC
SIGNALLING ENCRYPTION: INSEC
FAR-END H.323 SIGNALLING IP: 10.20.10.11
FAR-END MEDIA ENDPOINT IP: 10.20.20.242 PORT: 5200
FAR-END VendorID: Not available
MEDIA PROFILE: CODEC G.729A NO-LAW PAYLOAD 30 ms VAD OFF
DIAL DN 2735
MAIN_PM ESTD
TALKSLOT ORIG 17 TERM 116
EES_DATA:
NONE
QUEU NONE
CALL ID 0 19140

ooh323.conf

[general]
port=1720
bindaddr=0.0.0.0
gatekeeper=DISABLE
context=from-h323
dtmfmode=rfc2833
progress_setup=8
progress_alert=8
faststart=yes
h245tunneling=yes
mediawaitforconnect=yes
;rtptimeout=999
;tos=lowdelay
;amaflags = default
dtmfmode=h245alphanumeric
;accountcode=h3230101
disallow=all
allow=g729

[trunk-dst]
type=friend
host=10.20.20.10
context=salida
disallow=all
allow=g729

--------------------------------- o ---------------------------------

extensions.conf

[from-gw1]
exten => _X.,1,Dial(OOH323/${EXTEN}@10.30.20.10,999,tT)
exten => _X.,2,Hangup

End of Life for Asterisk 1.6.2.x was this Saturday. As you appear to have just installed it, you should consider using a supported version, instead.

Also, from a peer support point of view, very few people use H.323, as against SIP.

I would assume, though, that Asterisk provides similar debugging information to that which it provides for SIP.