[cisco-voip] h323 trunk between cisco and asterisk

[s m]

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 <bmeade90 at vt.edu> 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 <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
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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