The current astcc Makefile puts the sound files into the wrong directory. It uses /usr/share/asterisk/sounds but it should be /var/lib/asterisk/sounds.
Karl Putz > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa > Sent: Saturday, January 29, 2005 12:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] How to use ASTCC with SIP ?? > > > > I got this error when i try to dial: > > -- Executing Answer("SIP/8000104-71a3", "") in new stack > -- Executing DeadAGI("SIP/8000104-71a3", "astcc.agi") in new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi > Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File > astcc-tone does not exist in any format > Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File > astcc-accountnum does not exist in any format > Jan 29 18:11:37 WARNING[3412]: file.c:779 ast_streamfile: Unable to open > astcc-accountnum (format alaw): No such file or directory > == Spawn extension (prepaid, 77, 2) exited non-zero on > 'SIP/8000104-71a3' > > Can somebody tell me why and how to solve it ?? > > Regards. > Daniel. > > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Darren > Wiebe > Sent: samedi 29 janvier 2005 18:12 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] How to use ASTCC with SIP ?? > > I would recommend using the local trunk and then you just need a context > > that will dial out in your extensions.conf. Just put the context name > into the "Peer/Trunk" field on the trunks page. Currently there is not > support in astcc for oh-323. It would be trivial to add but.... > > Darren Wiebe > [EMAIL PROTECTED] > > Daniel Eboa wrote: > > > Hello List, > > > > I've set up asterisk and install astcc application, everything was > > well installed, but i'm having problem using astcc with SIP. I don't > > have any Trunk card or any other analogic VoIP card connected to my > > asterisk box. I'm using SIP and asterisk-oh323 to connect to my VoIP > > provider. Does anyone knows how I can use astcc to work with my config > ? > > > > Thanks. > > > >----------------------------------------------------------------------- > - > >_______________________________________________ > >Asterisk-Users mailing list > >[email protected] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
