Displaying 5 results from an estimated 5 matches for "konstantoulakis".
2004 Nov 30
1
realTime configuration help needed
...namic',
'default', '192.168.1.203');
Though I see asterisk connecting to the DB from mySQL-logs, I can't
seem to get anything else....
How can I get my configuration from the DB ?
Am I doing something worng ?
My traces are bellow..
Thanx in advance, for any help,
George Konstantoulakis.
------------------------------------------
When I start with vvvcd I get :
[res_config_mysql.so] => (MySQL RealTime Configuration Driver)
Nov 30 12:44:21 DEBUG[3088]: config.c:517 __ast_load: Parsing
/home/gkon/slash/etc/asterisk/res_mysql.conf
Nov 30 12:44:21 WARNING[3088]: res_config_mysq...
2005 Feb 28
2
Asterisk-OH323 no ringing
Hello,
I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5).
Everything is working fine, well, except that : when a call is made from
an h323 device (gnomemeeting for example), the caller does not hear any
ringing at all, he suddenly hears the person who answers the phone.
That can be quite disturbing for the users.
Any help would be very welcome. thank you.
Yves
2005 Aug 12
8
Incompatible destination (88) Error Message
I have connected asterisk 1.0.7 with Avaya Definity via E1 with a
TE100P Digium Card.
Inbound calls are working perfectly and I dont have any problem. But
when I try to make an outgoing call with my softphone (xlite) I am
getting the following messages.
Hungup 'Zap/13-1'
Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack
Called g1/3118
Channel 0/1, span 1 got
2005 Jan 12
3
Bristuff 0.20RC3 loses connectivity after short line interruption?
I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the phone jack for the ISDN line. Even
after plug it back in asterisk still reports that it could not create a
zap channel when I try to dial out and the line gives an engaged tone when
I try to dial.
Re-starting asterisk doesn't solve this, I have to stop asterisk, unload
the modules, reload
2005 Mar 18
15
Meetme2 compilation problem
Hi All,
I am trying to compile meetme2 in my asterisk box and getting the
following compilaton error. Please help me to sort it out.
cc -fPIC -c -o app_dial.o app_dial.c
In file included from app_dial.c:14:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared
(first use in this function)