[cisco-voip] h323 trunk between cisco and asterisk

[Anthony Holloway]

Can you get the call established, even for 5 seconds, and then run this
command?

show call active voice | in Peer|Remote|Dtmf|Coder|VAD
On Tue, Apr 28, 2015 at 2:56 AM s m <sam.gh1986 at gmail.com> 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
>
> any comments or hints are really appreciated.
> SAM
>
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>
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