Hello
I compiled the cvs version of app_conference without problem, and it is loaded normally by asterisk (1.2.9.1)
When my first client connect on the conference command, Asterisk stop with "Illegal Instruction", after some "translate.c:116 ast_translator_build_path: No translator path from unknown to unknown" ...
Does someone know from where the problem comes ? I tried with astersk 1.2.7, and with older versions of app_conference, with the same results... It's running on an old FC1.
Thanks
222
333Monitor interface and speaker detection
by squale, Thursday 20 of April, 2006 [07:35:59 UTC]
It is said in this page that app_conference presents messages on the Monitor interface for determine which speakers are active. I can't manage finding how it works. Is CLI the Monitor interface ? Does anyone know how to see the active speaker ?
Thanks
Finally I found it. Asterisk sends events with the asterisk management protocol. So you can see events with telnet for exemple by connecting to the AMP.
222
333Error in CVS instructions
by rodney, Sunday 13 of November, 2005 [00:29:08 UTC]
CVS instructions should read:
CVSROOT=:pserver:anonymous@cvs.sourceforge.net:/cvsroot/iaxclient
... otherwise anonymous password will be rejected
222
333Downloads
by , Wednesday 02 of February, 2005 [20:24:56 UTC]
Better question... where's the tarball? I go to their SF site, no files released. I go to the SF CVS site, there seems to be no way to get a snapshot.... show do I GET This program?
222
333Master of Aster
by , Tuesday 26 of October, 2004 [19:22:42 UTC]
are there any documentations on app_conference available?
for example if A wants to initiate a conference with B and C, how does he proceed? How could one add a leg to a conference?
VoIP SIP SDK is a soft phone sdk solution to quickly build voip softphone to dial and receive phone calls or add voip chat features in your software application.
VoIP SDK provides a powerful and highly customizable solution (SDK includes such features: SIP activeX control, Dynamically loadable codecs, DTMF, STUN support, IM interface, Adaptive silence detection and many more) to quickly add SIP based dial and receive phone calls (to make a long story short - voip client) features in your software applications. It accelerates the development of SIP compliant softphone with a fully-customizable user interface and brand name.
...
Comments
333asterisk crash with app_conference
I compiled the cvs version of app_conference without problem, and it is loaded normally by asterisk (1.2.9.1)
When my first client connect on the conference command, Asterisk stop with "Illegal Instruction", after some "translate.c:116 ast_translator_build_path: No translator path from unknown to unknown" ...
Does someone know from where the problem comes ? I tried with astersk 1.2.7, and with older versions of app_conference, with the same results... It's running on an old FC1.
Thanks
333Monitor interface and speaker detection
Thanks
Finally I found it. Asterisk sends events with the asterisk management protocol. So you can see events with telnet for exemple by connecting to the AMP.
333Error in CVS instructions
CVSROOT=:pserver:anonymous@cvs.sourceforge.net:/cvsroot/iaxclient
... otherwise anonymous password will be rejected
333Downloads
333Master of Aster
for example if A wants to initiate a conference with B and C, how does he proceed? How could one add a leg to a conference?