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@mail 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. Anything wrong with contexts or something? Thx Guys
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@mail > > 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 <critch@basesys.com>
-----Original Message----->From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Steven >Critchfield >Sent: Friday, March 04, 2005 10:26 PM >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@mail >> >> 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 SIPSteven, I think he is referring to MWI notification to the ATA. Anton - I believe you need to identify the voicemail context in the sip.conf if you are using something other than [default] like: mailbox=201@home Karl Putz>device. >-- >Steven Critchfield <critch@basesys.com> >
True. I remember it was working on time but cant remember what config it had. Anybody using Granstreams handytone 286 atas? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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@mail > > 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 <critch@basesys.com> _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Worth a try Karl... Thx! Ill let you know how it went in a few minutes. You were right! Sip.conf needs to have the voicemail context on the Mailbox line on each sip phone config. Thx Guys! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Karl H. Putz Sent: Viernes, 04 de Marzo de 2005 09:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Stutter Tone -----Original Message----->From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Steven >Critchfield >Sent: Friday, March 04, 2005 10:26 PM >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@mail >> >> 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 SIPSteven, I think he is referring to MWI notification to the ATA. Anton - I believe you need to identify the voicemail context in the sip.conf if you are using something other than [default] like: mailbox=201@home Karl Putz>device. >-- >Steven Critchfield <critch@basesys.com> >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Sat, 2005-03-05 at 14:10, 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@mail > > 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. > > Anything wrong with contexts or something?Stutter tome works with Zap but does it work with SIP phones unless they have their own stutter tome which activates when they get a "NEW MESSAGE" header. My HOP 1002 SIP phones certainly don't have that.> > Thx Guys > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft." ------------------------------------------ "Flatter government, not fatter government; Get rid of the Australian states."
The grandstream 101's have stutter tone. Works great with asterisk@home. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Howard Lowndes Sent: Friday, March 04, 2005 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Stutter Tone On Sat, 2005-03-05 at 14:10, 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@mail > > Voicemail delivery and all works great but when I check sip extensionext1> (analog phone using a Granstream ATA 286), the stutter tone signaling > message waiting does not work. > > Anything wrong with contexts or something?Stutter tome works with Zap but does it work with SIP phones unless they have their own stutter tome which activates when they get a "NEW MESSAGE" header. My HOP 1002 SIP phones certainly don't have that.> > Thx Guys > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft." ------------------------------------------ "Flatter government, not fatter government; Get rid of the Australian states." _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users