Carlos Alberto Lara de Hoyos
2005-Jun-11 15:05 UTC
[Asterisk-Users] SIP-H.323 dial tone and busy tone problem.
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.
Moises Silva
2005-Jun-13 08:48 UTC
[Asterisk-Users] SIP-H.323 dial tone and busy tone problem.
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 <clara@unfime.uadec.mx> 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"