Re: [Asterisk-Users] H323 calls will not stay connected
Daren J. Howell DTCommunication wrote: 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? Hi Daren, I believe the endpoints negotiate the master slave thing, so I'm not sure this is the issue here. I had the exact same problem when I set up and it was caused by a codec mismatch, but I'm sure there are other factors that will give the same result. Sorry I can't offer any more. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 calls will not stay connected
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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 calls will not stay connected
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=allallow=codec of choiceat the asterisk end and whatever you need to do at the legacy end.Regards,Richard Daren J. Howell [EMAIL PROTECTED] www.dtcommunication.com PH 678.388.9163 FX 678.921.2133 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 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 '0002010507d6080b21380ef4016a'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 [EMAIL PROTECTED] ~]# DJ. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users