Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made?
best regards On 6/11/05, Carlos Alberto Lara de Hoyos <[EMAIL PROTECTED]> wrote: > Greetings to the list: > > this is my problen when I make a call from my asterisk towards a nortel > PBX , the call is made but in my telephone sip I do not listen the dial tone > or the busy tone but the call it is completed normally. > > > > sip-phone-g729-------------asterisk--------h323-g729--------------nortel-pbx > > thi is may configuration: > > RedHat 8 2.4.18-14 > Asterisk 1.0.7 > The NuFone Network's Open H.323 Channel Driver > G.729/PCM16 Codec Translator > Raw G729 data > > It is a problem of codecs compatiblility or wath? > > Thanks to all. > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users