HI Mike, You wouldn't be trying to connect to Comindico in Australia by any chance?
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Mike O'Connor > Sent: Monday, 18 October 2004 02:05 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation > > > Hi all > > I spent a few hours trying to information on asterisk, h323 > and sip support for codecs with 20ms packetisation, and have > not been able to find anything relivatant. > > Our supplier of call termination requires h323 the following: > > * The signalling port is 1720 > * H.323 version 2 with fast start and H.245 Tunneling. > * The call should be initialised as Gateway-Gateway not using RAS. > * The codecs supported are G.729, G.711alaw and G.711ulaw all > at 20 millisecond packetisation. Your equipment must support > all three and be able to dynamically negotiate these during > call setup. > * We use RFC 2833 for out-of-band DTMF. Your equipment must > support this. The NTE RTP Payload type supported is 99. > > I was able after reading the source code in chan_h323.c to > work out how to enable fast start and h.245 tunneling. > > But the 20ms packetisation has me beat. > > I have made a test call to the provider which did not work > becase I was sending 30ms voice packets. > > SO my question does any one know now to force the correct > voice packet size ? > > Thanks > > Mike > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
