[Freeswitch-users] Problem playing media

[Brian West]

Good to hear you have it working.  That is always an unknown when you  
don't own the server and makes trouble shooting a problem.

/b

On Apr 13, 2008, at 12:12 AM, Nicolas Brenner wrote:

> Well, I finally resolved my problem. How? I got a server on
> theplanet.com and installed FS there. I'm really disappointed, because
> at first FS worked great on mediatemple dv (the server I had first,
> with which I started having trouble), and also because it's been
> really hard dealing with the guys at the planet, ordering a server is
> a pain in the butt, and mediatemple's web interface is a breeze.
> Anyway, that's totally offtopic. Thanks if anybody read this, and just
> in case, consider the info above when you are considering dedicated
> servers' services.
>
> On 4/10/08, Nicolas Brenner <nicolas at medularis.com> wrote:
>> On Wed, Apr 9, 2008 at 7:36 AM, Brian West <brian at 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
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> Brian West
>>> sip:brian at 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
>>
>
>
> -- 
> Nicolás Brenner
>
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Brian West
sip:brian at freeswitch.org