Now it works, now it doesn't - identical install - VM prob?

I installed Asterisk@Home 1.5 packaged with Centos 3.5 last year on a thinkpad and used some .conf files provided by a friend since I could neither understand the documentation nor get the config helper tools to work. I am using it only as a test system for PC to PC calls on a LAN and it is very stable.

Last week I decided to install the same image to a VMware Virtual Machine. Installation went fine, and then I copied my three modified *.conf files (extensions, sip, and voicemail) to /etc/asterisk, set up the network and rebooted. Result? The phones can all call each other and set up conversations. However, all the system messages (the person at extension xxxx is not available, etc.) do not play.

How should I go about troubleshooting this? Thanks in advance!
/sawtode/

step 1 would be looking at the log files. /var/log/asterisk/full

step 1 would be looking at the log files. /var/log/asterisk/full[/quote]

Here is the log file. I started up the system and made one call from ext. 2005 to ext. 2000. 2000 was not present on the network, but as soon as I called it I got a “connected” status on the phone, but no announcement for voicemail. Here is the log. I would be grateful for assistance in reading it.

Asterisk server is 192.168.0.2, extension 2005 is 192.168.0.150.

Jun 22 14:10:53 DEBUG[1361]: Setting NAT on RTP to 0
Jun 22 14:10:53 DEBUG[1361]: Stopping retransmission on ‘0782473a2258cb6954b74d5760db9038@chicago’ of Request 102: Found
Jun 22 14:11:05 DEBUG[1361]: Auto destroying call ‘5F0E7E5FF5E14EBDA610BD0C18DD7D56@asterisk’
Jun 22 14:11:18 DEBUG[1361]: Setting NAT on RTP to 0
Jun 22 14:11:18 DEBUG[1361]: Stopping retransmission on ‘83341AF5-8831-438B-BA5E-30511C41B30B@192.168.0.150’ of Response 20106: Found
Jun 22 14:11:18 DEBUG[1361]: Setting NAT on RTP to 0
Jun 22 14:11:18 DEBUG[1361]: Check for res for 2005
Jun 22 14:11:18 DEBUG[1361]: Call from user ‘2005’ is 1 out of 0
Jun 22 14:11:18 DEBUG[1361]: build_route: Contact hop: sip:2005@192.168.0.150:5060
Jun 22 14:11:18 VERBOSE[1361]: – Executing Dial(“SIP/2005-86ab”, “SIP/2000|20”) in new stack
Jun 22 14:11:18 DEBUG[1361]: Setting NAT on RTP to 0
Jun 22 14:11:18 NOTICE[1361]: Unable to create channel of type ‘SIP’
Jun 22 14:11:18 VERBOSE[1361]: == Everyone is busy/congested at this time
Jun 22 14:11:18 DEBUG[1361]: Exiting with DIALSTATUS=CHANUNAVAIL.
Jun 22 14:11:18 VERBOSE[1361]: – Executing VoiceMail(“SIP/2005-86ab”, “b2000”) in new stack
Jun 22 14:11:18 DEBUG[1361]: Stopping retransmission on ‘83341AF5-8831-438B-BA5E-30511C41B30B@192.168.0.150’ of Response 20107: Found
Jun 22 14:11:18 DEBUG[1361]: voicemail/sip/2000/busy doesn’t exist, doing what we can
Jun 22 14:11:18 DEBUG[1361]: Ooh, format changed from unknown to ulaw
Jun 22 14:11:18 DEBUG[1361]: Scheduling timer at 160 sample intervals
Jun 22 14:11:18 VERBOSE[1361]: – Playing ‘vm-theperson’ (language ‘en’)
Jun 22 14:11:18 DEBUG[1361]: Oooh, format changed to 2
Jun 22 14:11:21 DEBUG[1361]: Hang up during prefile playback
Jun 22 14:11:21 VERBOSE[1361]: == Spawn extension (any, 2000, 102) exited non-zero on ‘SIP/2005-86ab’
Jun 22 14:11:21 DEBUG[1361]: Scheduling timer at 0 sample intervals
Jun 22 14:11:21 DEBUG[1361]: cdr_mysql: inserting a CDR record.
Jun 22 14:11:21 DEBUG[1361]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (‘2006-06-22 14:11:18’,’“2005” <2005>’,‘2005’,‘2000’,‘any’, ‘SIP/2005-86ab’,’’,‘VoiceMail’,‘b2000’,3,3,‘ANSWERED’,3,’’)
Jun 22 14:11:21 DEBUG[1361]: update_user_counter(2005) - decrement inUse counter

Thanks,
Sawtode

you need to go back and look at the logs generated when asterisk is loading, the “unable to create a channel of type SIP” would indicate something much more fundamental is wrong. did you compile asterisk ? any errors ?

I am happy to do that, but I don’t know which log file(s) you refer to. I have the Asterisk Manual V2 here in front of me which I downloaded, and there are no searchable instances of logfile, “log file”, diagnostic, or troubleshooting. A few hits on logs, log and fail are not revealing. So I have no place to start. I looked at other files in /var/log/asterisk and found nothing that seemed relevant. I also looked at /var/log/asterisk/cdr-csv/Master.csv which lists calls to the test extensions but the Asterisk manual makes no mention or explanation of this file. A search on Master.csv and on csv alone comes up empty.

This image I am using is an ISO I downloaded from the Asterisk pages last October. It was already compiled and installs automatically.

Keep in mind that it is possible to make a normal call between two online extensions. These problems appear when I call an offline extension and should expect to hear voicemail.

Thanks,
Sawtode.

what codec are the phones using ?