I’ve installed Asterisk/CentOS using Vmware 5.0 on my winxp laptop to show it as a proof of concept. Even though calls between extensions are working well, I hear no voice coming when I try voicemail and the call drops when I dial e.g. *99 after 6-7 seconds. When I took a tcpdump of RTP/RTCP of voicemail I saw that there is no RTP/RTCP going out of the IP address Asterisk is using, I do see RTP coming in from my winxp IP where I have my X-lite client running so there is one way speech. I think the call drops due to codec negotiation failure (my guess, I don’t see it in the debug log). I searched for similar problems on the net but didn’t see any. I know this is a weird configuration but I think it should work, any help or suggestions are welcome to correct the problem.
Do you have the console logging cranked up to include ‘debug’ via logger.conf ? If so, it WILL provide a painfully detailed explanation of where the failure is occuring…
From logger.conf: console => debug,notice,warning,error
Starting Asterisk: asterisk -dvvvvvvvvvvc # just make sure to use a lot of v’s
This should reveal what is going on, the console logging statement does not include debug by default (as well it shouldn’t).
I agree that your configuration will work just fine via vmware so long as you don’t attempt to bring any PCI telephony hardware into the equation. SIP & no-analog-dialout capability should work just fine…I’d recommend bridged vm networking as well so that the asterisk vm host has an autonomous IP address.
Thanks for the tips. I had already done all of them (bridged networking, debug enabled since first boot), and being a former project/field support engineer I also went through the log file before posting for help to the forums. I don’t believe seeing an error message but I’ll check again. If I’m still stuck, is it allowed to attach part of the logging to the text message so that more experienced people might see what I don’t?
