BK [address only for mailing lists]
2003-Jul-07 13:31 UTC
[Asterisk-Users] Asterisk crashing after Voicemail box creation
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@nikotel|60|r") in new stack -- Called 442071231234@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