[cisco-voip] h323 trunk between cisco and asterisk

[gentoo at ucpenguin.com]

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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