asterisk users - May 2006 - H323 calls will not stay connected

Have Asterisk connected to a H323 compatible legacy PBX using QSIG protocol
and IP trunks.

I can call to Asterisk, and from Asterisk using X-Lite softphone but
whenever either end picks up, the calls disconnects.

No gatekeeper is installed. I have attached a copy of my h323 logfile for
debugging. 

What do you suggest what change needs to take place to keep calls connected?


 

11:33:19:864  Queued H245 messages 1. (incoming, ooh323c_7)

11:33:19:864  msgCtxt Reset? Done (incoming, ooh323c_7)

11:33:19:864  MasterSlaveDetermination done - Slave(incoming, ooh323c_7)

11:33:19:864  Not opening logical channels as Cap exchange remaining

11:33:19:864  Finished handling H245 message. (incoming, ooh323c_7)

11:33:19:864  Receiving H.2250 message (incoming, ooh323c_7)

11:33:19:864  Received Q.931 message: (incoming, ooh323c_7)

11:33:19:864  Received H.2250 Message = {

11:33:19:864     protocolDiscriminator = 8

11:33:19:864     callReference = 2

11:33:19:865     from = originator

11:33:19:865     messageType = 5a

11:33:19:865     Cause IE = {

11:33:19:865        Unsupported Cause Type

11:33:19:865     }

11:33:19:865     h323_uu_pdu = {

11:33:19:865        h323_message_body = {

11:33:19:865           releaseComplete = {

11:33:19:866              protocolIdentifier = {

11:33:19:866                 {

11:33:19:867  0 0 8 2250 0 2 }

11:33:19:867              }

11:33:19:868              callIdentifier = {

11:33:19:868                 guid = {

11:33:19:869                    '0002010507d6080b213800000ef4016a'H

11:33:19:870                 }

11:33:19:870              }

11:33:19:871           }

11:33:19:871        }

11:33:19:872     }

11:33:19:872  UUIE decode successful

11:33:19:872  Decoded Q931 message (incoming, ooh323c_7)

11:33:19:872  }

11:33:19:872  H.225 Release Complete message received (incoming, ooh323c_7)

11:33:19:872  Cause of Release Complete is 0. (incoming, ooh323c_7)

11:33:19:873  Closing H.245 connection (incoming, ooh323c_7)

11:33:19:873  Closed H245 connection. (incoming, ooh323c_7)

11:33:19:873  In ooEndCall call state is - OO_CALL_CLEARED (incoming,
ooh323c_7)

11:33:19:873  Cleaning Call (incoming, ooh323c_7)- reason:OO_REASON_UNKNOWN

11:33:19:873  Closing H.245 Listener (incoming, ooh323c_7)

11:33:19:873  Removed call (incoming, ooh323c_7) from list

[root@asterisk1 ~]#

DJ.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060509/e7ef4a65/attachment.htm

Daren J. Howell DTCommunication wrote:
> Have Asterisk connected to a H323 compatible legacy PBX using QSIG 
> protocol and IP trunks.
> 
> I can call to Asterisk, and from Asterisk using X-Lite softphone but 
> whenever either end picks up, the calls disconnects.
Try restricting both ends to one codec;

disallow=all
allow=codec of choice

at the asterisk end and whatever you need to do at the legacy end.

Regards,

Richard

I have restricted the asterisk server to G711 to match the choice on the
PBX, and still same result.

I have read that either endpoint have to be either a master or slave to
communicate to each other. I see in the logs that both are shown to be a
slave. The pbx side has to be set to slave. How can I lock the asterisk side
to be a master? Or is this something to worry about?

 

_____________________________________

Richard wrote:

Try restricting both ends to one codec;
 
disallow=all
allow=codec of choice
 
at the asterisk end and whatever you need to do at the legacy end.
 
Regards,
 
Richard

 

 

Daren J. Howell

dhowell@dtcommunication.com <mailto:dhowell@dtcommunication.com> 

www.dtcommunication.com

PH 678.388.9163

FX 678.921.2133

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/53e338f6/attachment.htm