> Hi all, > > in an effort to create a SIP <-> H.323 translator we've found and fixed > several problems in H.323 channel. These inlcude: > > for SIP->H.323 calls > > - no ringback tone > - ringback not related to H.323 events > - one-way audio with Cisco CallManager > - incorrect Caller ID > > for H.323->SIP calls > > - not able to establish call with Cisco IOS 12.3(4)T > - ringback not related to SIP events > - no support for 183 Call Progress > - incorrect Caller ID > > > Please find the patches against aterisk 0.7.2 release below. > > > M. >
Did you put these files to bugs.digium.com ? _______________________________________________ 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
