Re: [asterisk-users] User unable to use DTMFs?
Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. --- On Tue, 7/1/08, Vincent [EMAIL PROTECTED] wrote: From: Vincent [EMAIL PROTECTED] Subject: [asterisk-users] User unable to use DTMFs? To: asterisk-users@lists.digium.com Date: Tuesday, July 1, 2008, 11:09 AM Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User unable to use DTMFs?
On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob [EMAIL PROTECTED] wrote: Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. Users call into our Asterisk voice server through a Zaptel PCI interface from regular phones, usually from a PBX (virtually all of them ISDN-based). The only files I modified are zaptel.conf, zapata.conf, and extensions.conf, which don't have anything DTMF-related, so Asterisk uses the default options. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User unable to use DTMFs?
On Tue, Jul 01, 2008 at 02:49:17PM +0200, Vincent wrote: On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob [EMAIL PROTECTED] wrote: Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. Users call into our Asterisk voice server through a Zaptel PCI interface from regular phones, usually from a PBX (virtually all of them ISDN-based). So those phones are analog or BRI? The only files I modified are zaptel.conf, zapata.conf, and extensions.conf, which don't have anything DTMF-related, so Asterisk uses the default options. Hmmm any chance we could have a llok at them? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
