"network 'C0A80047'H" is the IP address. It's just in hex. That would be 192.168.0.71.
Can you send the full H.245 exchange for a call? That should show us where it is failing. We'll want to make sure it gets all the way yo both sides sending OpenLogicalChannelAcks. On Wed, Apr 29, 2015 at 1:14 AM, s m <[email protected]> wrote: > thank you Brian, yes i have set bind address. when i enable h245 > debugging, all messages have no ip address like this: > value OpenLogicalChannel ::= > { > forwardLogicalChannelNumber 1001 > forwardLogicalChannelParameters > { > dataType nullData : NULL > multiplexParameters none : NULL > } > reverseLogicalChannelParameters > { > dataType audioData : g711Ulaw64k : 20 > multiplexParameters h2250LogicalChannelParameters : > { > sessionID 1 > mediaChannel unicastAddress : iPAddress : > { > network 'C0A80047'H > tsapIdentifier 17680 > } > mediaControlChannel unicastAddress : iPAddress : > { > network 'C0A80047'H > tsapIdentifier 17681 > } > } > } > } > > > > Apr 29 05:09:23.499: H245 FS OLC INCOMING ENCODE BUFFER::= > 0003E90C6013800A04000100C0A800474511 > Apr 29 05:09:23.499: > Apr 29 05:09:23.499: H245 FS OLC INCOMING PDU ::= > > value OpenLogicalChannel ::= > { > forwardLogicalChannelNumber 1002 > forwardLogicalChannelParameters > { > dataType audioData : g711Ulaw64k : 20 > multiplexParameters h2250LogicalChannelParameters : > { > sessionID 1 > mediaControlChannel unicastAddress : iPAddress : > { > network 'C0A80047'H > tsapIdentifier 17681 > } > } > } > } > > i think it is problem. cisco does not know where should send rtp packets. > am i right??? do you have any hint about it??? > > thank you for your attention. > SAM > > On Tue, Apr 28, 2015 at 8:04 PM, Brian Meade <[email protected]> wrote: > >> Do you have "h323-gateway voip bind srcaddr x.x.x.x" configured on an >> interface? >> >> You'll want to run "debug h245 asn1" to see if media negotiations as well. >> >> On Tue, Apr 28, 2015 at 3:55 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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