VoIP Mailing List Archives

[sam.gh1986 at gmail.com]

hello every body,


i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with ooh323 module. i configured both side and have successful call from cisco to asterisk. but when call comes from asterisk to cisco, my phone rings but no audio is heard and call is disconnected after 5 second. i enable "debug voice rtp" in cisco and see the source address for receiving rtp packets is 0.0.0.0

 Apr 28 07:46:34.765: RTP(50493): ps rx s=0.0.0.0(0), d=192.168.0.139(17112), pt=8, ts=BF40, ssrc=2C1690C9

any body knows how should i fix it?


this is my ooh323.conf file:
[general]
port=1720
context=from-trunk
gatekeeper=DISABLE
bindaddr=192.X.X.X
disallow=all
allow=all
AcceptAnonymous=yes
directrtpsetup=yes
directmedia=yes
faststart=yes
h245tunneling=yes
mediawaitforconnect=yes
tos=lowdelay

[sam]
type=user
host=192.X.X.X
directmedia=yes

[sam-1]
type=peer
host=192.X.X.X
directmedia=yes


any comments or hints are really appreciated.

SAM