True. I remember it was working on time but cant remember what config it had.
Anybody using Granstreams handytone 286 atas? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Viernes, 04 de Marzo de 2005 09:26 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stutter Tone On Fri, 2005-03-04 at 21:10 -0600, Anton Krall wrote: > I think I have something misconfigured regarding voicemails. They work > great, I have this setup: > > Sip.conf > > [ext1] > Context=phones > Mailbox=201 > > Voicemail.conf > > [home] > > 201,password,name,[EMAIL PROTECTED] > > Voicemail delivery and all works great but when I check sip extension > ext1 (analog phone using a Granstream ATA 286), the stutter tone > signaling message waiting does not work. SIP dialtones come from the SIP device. Look up the config on your SIP device. -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ 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
