asterisk users - Jan 2005 - g729 problem

Hi.

I've searched the wiki and the newsgroup to no avail.

 

We have a couple of grandstream phones and purchased seceral copies of
the G729 codec for use with them.

 

We registered the codec (first time) on a testing machine and it worked.

We registered it on the production machine (second time) and it worked.

 

Then the  production system crashed.

 

After reinstallation we had to call Digium to re-register (third time).

We have, and it is registered, and yet asterisk keeps giving us these
error messages when we try to use them

 

    -- Executing VoiceMailMain("SIP/203-7cc0", "") in new
stack

Jan 19 07:46:51 NOTICE[8809]: channel.c:1696 ast_set_write_format:
Unable to find a path from gsm to g729

Jan 19 07:46:51 WARNING[8809]: file.c:779 ast_streamfile: Unable to open
vm-login (format g729): No such file or directory

Jan 19 07:46:51 WARNING[8809]: app_voicemail.c:3310 vm_execmain:
Couldn't stream login file

  == Spawn extension (internal, #, 1) exited non-zero on 'SIP/203-7cc0'

 

Any suggestions?

 

Thanks.

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Hoping someone can point me in the right direction. I
have the following setup:

Trixbox latest (asterisk 1.2.11)
DID thru IAX trunk (EXGN/Vitelity)
Termination thru IAX trunk (EXGN/Vitelity)
3 GXP-2000 phones with firmware (1.1.0.16)
6 Digium g729 licenses

I have no codecs specified per trunk or extension. In
iax.conf and sip.conf, with disallows=all and
allow=gsm, everything works fine. When I change it to
allow=g729, I have problems with incoming calls, and
calls from the extensions into the system (voicemail,
etc). Outgoing calls work just fine (no transcoding).
Incoming calls are automatically answered by the
system attendant with a quick greeting, then dumped
into a call queue. With g729 enabled, calls get
answered, but no audio is heard. Same with phones
trying to check voicemail. After hanging up, it
doesn't release the codecs. 'show g729' gives:

6/0 encoders/decoders of 6 licensed channels are
currently in use

And it will stay like that until asterisk is
restarted. I'm assuming its a transcoding issue with
the system attendant/prompts, but I can't see how it
would need more than one license per incoming call,
much less six, and why the licenses aren't released
after the call ends. And I haven't found a console
command that actually lists the per license usage. Any
help resolving this would be appreciated. I already
purchased 3 more licenses than I theoretically need,
just to throw $30 at the problem and save some hair,
but that didn't work. Thanks in advance.

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