[Asterisk-Users] codec error connecting to cisco gateway

[Zafer Khodr]

I hope someone can help me with this.

 

I have a cisco gateway trying to dial my box via H.323.

The call comes through ok and gets routed properly.. only thing is " NO
AUDIO "

I am confident that I have narrowed down the problem to a codec issue.

I have the relevant G729 licences which were purchased from diguim.

The calling party was calling using G729.

I asked them which version of the codec and they told me G729r8 .  what ever
that means?????

 

I get about 150 of these lines in the log file per second while the call is
active.

 

Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.

 

Another party with a quantum device tried to call me and it was the same
error.. they even tried a SIP call and the result was the same.

 

I had one party call me through a softswitch and both sip and H323 worked
fine for that person.

 

In my mind it seem that there is some kind of incompatibility with the audio
codec the quintum and cisco are sending and the one asterisk is using.

Could this be the case and if so is there a work around?

Has anyone else had this issue before or does anyone know possibly if I am
wrong what the problem may be?

 

Thanks in advance

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