SIP Voicemail recording not working

I’m curently using Asterisk 1.2, but i had the same issue with 1.0.7

Whenever a call gets redirected to voicemail, the recording fails…
here’s a log sample: (phone number replaced wix x’s)
– Playing ‘beep’ (language ‘fr’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/xxxxxxxxxx/INBOX/msg0005 format: wav, 0x8150468
Nov 24 09:33:38 WARNING[23025]: app.c:653 ast_play_and_record: No audio available on SIP/xxxxxxxxxx-2257??
– User hung up
== Spawn extension (vms, xxxxxxxxxx, 6) exited non-zero on ‘SIP/xxxxxxxxxx-2257’

From what i see there it could be a codec problem, but i don’t get any SIP error concerning codec negotiation…

And before getting to voicemail, codec negotiation goes well:
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

the peer is a Tekelec T7000 switch, and currently the only codec i set on it is ulaw.
I’d just like to know if recording voicemail is compatible with using only ulaw as codec…
Or if something else could create the problem: Nov 24 09:33:38 WARNING[23025]: app.c:653 ast_play_and_record: No audio available on SIP/xxxxxxxxxx-2257??

Thanks in advance

Is there any NATing going on between Asterisk and the Tekelec T7000 switch?

Ok, we fixed our problem…

The problem was the t7000 was doing load balancing over 2 FastEthernet port, and one of them could not ‘talk’ the to asterisk box…

Sorry for posting a stupid question ;-p