[Asterisk-Users] SIP-H.323 dial tone and busy tone problem.

Moises Silva moises.silva at gmail.com
Mon Jun 13 08:48:12 MST 2005


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 at 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 at 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"



More information about the asterisk-users mailing list