Re: [GnomeMeeting-devel-list] GM 1.3.0



Hi,

Le lundi 14 mars 2005 à 00:33 +0100, daniel huhardeaux a écrit :
> Hi everybody,
> 
> thanks to Kilian, I just installed the GM-opal version, Debian SID 
> kernel 2.6.10. For information, I didn't remove the gnomemeeting-cvs 
> packages so I had information concerning conflicts about the 2 versions 
> (png files). Installation finished ok.
> 
> 1) I reinstall through the druid and had one problem with the window 
> from devices test, this one when you speak in mic and GM repeat: I 
> could'nt close it with mouse, I had to press enter twice (the first 
> press closed the window under the test one, the second close it)
> 

Sounds weird, it didn't change from 1.2.1 and I can not reproduce it. I
will investigate.

> 2) In H323 parameters: not able to register to GnuGK/radius with the 
> username from account, the first and last name are sended (option use 
> this alias to register from 1.2 is missing) BTW, is the registration 
> status to GK each x minutes still  enabled?
> 

No, not yet reimplemented. The option to register to login as first
alias is not reimplemented yet, but it is not needed anymore with recent
GNU GK installations. So, should I really reimplement it?

> 3) I create SIP account to connect to my asterisk box (CVS HEAD 
> 02/27/05): I have the registration info in GM, I see in my logs that GM 
> is connected and then  GM  after ~ 1mn tell registration failed.  SIP 
> show peers from asterisk inform that GM is connected. I uncheck/recheck 
> the account, it's ok in GM status (asterisk show me unregistred / 
> registred so seems that everything was fine on asterisk side).
> 

I don't understand. What should I do to reproduce that problem and what
is the problem?

Actually, Asterisk should work very well. 

But don't be too picky with GM for now, I have concentrated most of my
efforts in OPAL, not in GM.

> 4) I call my asterisk box through H323/OH323 channel: perfect, like it 
> was in 1.2.
> 
> Then a call SIP://100 finish with an abnormal end of call from GM. In 
> asterisk logs, call is just passing fine! I think that perhaps I have to 
> add domain. So I call SIP://100 sipdomain Again call is going just well 
> in asterisk logs but GM is still ringing. I hangup and get a message 
> from GM that application crash and I have to inform developpers ... 
> Funny now, I clic where I want on the screen, GM restart the call and 
> crash definitely few seconds after. The first time I had this behaviour, 
> GM did'nt crash on the second call and I could'nt stop it (or with kill)
> 

Backtrace please ;)

> 5) I wanted to put my favorites ring files but they are not working :-( 
> After I choose them, I ask GM to play them and have ... silence ;-) The 
> original files are played ok. The files I want to use are the same that 
> from 1.2
> 

The code didn't change at all, and it is pwlib... Are you using the same
pwlib version?

> Voilà, results of my first play with 1.3
> 
> Great job from all the team, you're on the right way. Many thanks.
> 
-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
//\     FOSDEM 2005 : http://www.fosdem.org
v_/_    H.323 phone : callto:ils.seconix.com/dsandras seconix com




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