[Asterisk-Users] Asterisk crashing after Voicemail box creation

[BK [address only for mailing lists]]

Hi

I have just been struggling for four days to get SIP working and now as 
I created a voicemail box, Asterisk has become very unstable and it 
can't bridge SIP phone to SIP provider calls anymore.

Calling internally from one SIP phone to another works fine.

Calling internally from a SIP phone to an analog phone on a Zap channel 
and vice versa works fine.

Incoming PSTN calls delivered to a SIP phone also works fine.

Dialing out from an analog phone on a Zap channel using a SIP provider 
works fine as well.

HOWEVER,

when dialing out using a SIP provider (both Nikotel and iConnect) 
Asterisk cannot bridge the two legs of the call and all I get is silence.

here is what the console shows:

     -- Executing Dial("SIP/Sip1-1862", "SIP/442071231234 at nikotel|60|r") 
in new stack
     -- Called 442071231234 at nikotel
     -- SIP/nikotel-4815 is ringing
     -- SIP/nikotel-4815 answered SIP/Sip1-1862
     -- Attempting native bridge of SIP/Sip1-1862 and SIP/nikotel-4815
   == Spawn extension (internal, 00442071231234, 1) exited non-zero on 
'SIP/Sip1-1862'

NB: PSTN number edited

I have stop/started Asterisk and even rebooted the machine, but no 
change.

This problem has popped up after I created a voicemail box. When I 
tested the voicemail, at the moment when I tried to listen to the 
recording (VoicemailMain) Asterisk crashed. After restarting, I can now 
get into VoicemailMain without a crash, but there is now a problem with 
SIP and Asterisk crashes once in a while.

Any ideas?

thanks in advance
rgds
bk