Any change if you allow h323 to h323 and disable toll fraud prevention?
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
What does "debug voip dialpeer" show when the call is coming from
Asterisk? Which dial-peer do you intend for incoming calls from
Asterisk to use?
On 2015-04-28 02:55, s m wrote:
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 [1]
any comments or hints are really appreciated.
SAM
Links:
------
[1] http://192.168.0.71:1720
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