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 <[email protected]> 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 > > _______________________________________________ > cisco-voip mailing list > [email protected] > https://puck.nether.net/mailman/listinfo/cisco-voip >
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