[cisco-voip] h323 trunk between cisco and asterisk
Brian Meade
bmeade90 at vt.edu
Wed Apr 29 11:03:47 EDT 2015
"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 <sam.gh1986 at gmail.com> 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 <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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