On Wed, Apr 9, 2008 at 7:36 AM, Brian West <brian@freeswitch.org> wrote:
On Apr 9, 2008, at 1:17 AM, Nicolas Brenner wrote:
Hello everyone,
I'm having some trouble with FS :( apparently with mod_shout. I want to play an mp3 file after answering a call so I compiled mod_shout following the wiki, then configured an extension to answer a call and play an mp3 file I uploaded to the server. The thing is, FS supposedly plays the file, but I can't hear it on the softphone with which I'm calling, and also, after playing the file, FS seems to freeze and then I get a lot of:
sofia_event_callback() event [nua_r_bye] status [408][Request Timeout] session: n/a sofia_event_callback() event [nua_i_state] status [408][to BYE] session: n/a
lines on the console/log.
Well I can't see enough of the log with the two lines you posted to many any kinda of educated guess.
Ok, attached is a console log with sip traces as well. What I did was: - start freeswitch - register softphone (xlite) with extension 1000 - call 9998: FS answered the call and played the tetris sound, although I was only able to hear it for 3 secs - hanged up: FS received the hang_up and terminated the call - call 9998 again: FS answered the call and supposedly played the tetris sound, but I didn't hear anything on my side - hanged up: FS did not recieve the hang_up signal, and kept the call 'open' - tried calling 9998 again: FS didn't show any sip packets being received or anything, the call could not be made - shutdown FS
Also, after I try to call once, the softphone does not work anymore, and I have to make it register again with FS. After all this, when I shutdown FS, it takes some time, and while it's trying to shutdown it prints a lot of the same lines as above.
I can only guess you're using x-lite or eyebeam.
Yes, I'm using x-lite
I thought it was mod_shout, so I commented it from autoload_configs/modules.conf.xml so it does not load, and replaced the mp3 file with a wav file (FS has permission to read both of them), but I'm getting the same behaviour. Now I'm even getting that behaviour when dialing 9999 or 9998 (default dialplan pre-configured extensions).
Is this a real server or virtual server? ie vmware or xen?
This is a virtual server I think, made with virtuozzo. Is a dedicated virtual server hosted by mediatemple (www.mediatemple.net).
Now I can't get FS to work right, everytime I try to make a call, either initiating it from the console, the softphone or js, I get the same weird behaviour. Any clues? I'm sorry if this is trivial or a known issue, but I haven't been able to figure it out, thanks a lot for your time and help.
Your best bet at this point is to join the IRC channel and ask for help in realtime... Have you tried "make current" to ensure you don't have any code skew?
I tried make current several times. For the tests above (for which the attachment log file is), I removed /usr/local/freeswwitch, then checked out FS source from svn and compiled with default options (did not add mod_shout), and used default config files with no modifications (also installed audio and music files). Additionally, the server has a public IP address, and I'm connected directly to Internet (not nat involved whatsoever).
--
Nicolás Brenner
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Brian West sip:brian@freeswitch.org
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The files attached are: - consolelog.txt: What I saw on the FS console while doing the test - siptrace.log: Output of sip trace to a file using TPORT_DUMP env var - freeswitch.log: log/freeswitch.log on FS folder for this test My email was rejected by the list moderator, the files are here (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log Thanks for your help! -- Nicolás Brenner