[cisco-voip] h323 trunk between cisco and asterisk

[s m]

hello guys,

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 cisco config:

voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
dial-peer voice 1 voip
 destination-pattern 2.+
 voice-class codec 1
 session protocol sipv2
 session target ipv4:192.168.0.240
!
dial-peer voice 2 voip
 destination-pattern 1.+
 voice-class codec 1
 session target ipv4:192.168.0.71:1720

any comments or hints are really appreciated.
SAM
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