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
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