Re: [GnomeMeeting-devel-list] GM 1.3.0
- From: daniel huhardeaux <devel tootai net>
- To: GnomeMeeting development mailing list <gnomemeeting-devel-list gnome org>
- Subject: Re: [GnomeMeeting-devel-list] GM 1.3.0
- Date: Mon, 14 Mar 2005 12:03:22 +0100
Damien Sandras a écrit :
Hi,
Salut Damien
Le lundi 14 mars 2005 à 00:33 +0100, daniel huhardeaux a écrit :
[...]
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?
I send you the remarks I have. If it should be reimplement you have to
decide. I always say that, what was integrated and is not creating
problems on new versions, why remove it ;-) And I use an old GNU GK :-)
Anyway, it's not important.
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.
I just have to open GM: it register to asterisk and then unregister.
Here are logs:
10:56:56 GnomeMeeting 1.3.0 démarré pour l'utilisateur dh
10:56:58 Registered to gk.tootai.com
10:56:58 Registered to voip.tootai.com
10:57:13 Registration to voip.tootai.com failed: Timeout
For asterisk, GM is still registered. Question, why port is 5064?
Sometimes I saw GM registered with 5068. REALM is set to voip.tootai.com
(in * too) canreinvite=no and nat=yes. Tried with nat=no and to register
with private ip from asterisk, same result. As I told, for asterisk GM
is registered.
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 ;)
Starting program: /usr/bin/gnomemeeting-snapshot
[Thread debugging using libthread_db enabled]
[New Thread -1246568320 (LWP 7676)]
[New Thread -1250972752 (LWP 7679)]
[New Thread -1251234896 (LWP 7682)]
[assert] error: 111 (Connexion refusée)
[assert] where: "socket.c", "sw_socket_tcp_connect", line: 720
[New Thread -1253254224 (LWP 7687)]
[New Thread -1261642832 (LWP 7691)]
[New Thread -1251898448 (LWP 7693)]
[New Thread -1270092880 (LWP 7694)]
[assert] error: 111 (Connexion refusée)
[assert] where: "socket.c", "sw_socket_tcp_connect", line: 720
[New Thread -1270355024 (LWP 7697)]
[New Thread -1270617168 (LWP 7698)]
[New Thread -1270989904 (LWP 7699)]
[New Thread -1271252048 (LWP 7700)]
[Thread -1270355024 (LWP 7697) exited]
[New Thread -1270355024 (LWP 7701)]
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?
Found the problem: when you browse and select your file, you have
file:///home/user/blabla in the path. By clicking play, it's not
working. Now I remove the leading file:// (as it is in 1.2) and it work ;-)
--
Daniel Huhardeaux ______ _____ _____ ______ ______ __
enum +48 32 285 5276 /_ _// _ // _ //_ _// __ // /
IAX FWD +1 7009 422493 / / / // // // / / / / /_/ // /
sip:101 h323:121 @voip./_/ /____//____/ /_/ /_/ /_//_/.com
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