[Asterisk-Users] VoiceMailMain dumps user back into my incoming context after leaving a message
Nik Martin
nmartin at radiancetech.com
Tue May 18 08:20:51 MST 2004
Do you mean after the Voicemail (vs. after VoiceMailMain?) in each
extension?
I do have:
exten => .,3,Hangup
As step three at the bottom of my extensions context. Do I have to add it
as step 3 for every extension in the dial plan?
>From my extensions.conf:
[extensions]
exten => 0,1,Dial(SIP/jsantacapita,20,Tt)
exten => 0,2,Voicemail(u100)
exten => 0,102,Voicemail(b100)
exten => 105,1,Dial(SIP/nmartin,20,Tt)
exten => 105,2,Voicemail(u105)
exten => 105,102,Voicemail(b105)
exten => 101,1,Dial(SIP/mthomas,20,Tt)
exten => 101,2,Voicemail(u101)
exten => 101,102,Voicemail(b101)
exten => 102,1,Dial(SIP/dli,20,Tt)
exten => 102,2,Voicemail(u102)
exten => 102,102,Voicemail(b102)
exten => 100,1,Dial(SIP/jsantacapita,20,Tt)
exten => 100,2,Voicemail(u100)
exten => 100,102,Voicemail(b100)
exten => .,3,Hangup
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of brian
> Sent: Tuesday, May 18, 2004 9:45 AM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] VoiceMailMain dumps user back
> into my incoming context after leaving a message
>
>
> You need to add a hangup after the VoiceMailMain I also think
> exten => o will work in that case too ... not sure from
> VoiceMailMain but you could try it.
>
> bkw
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> > admin at lists.digium.com] On Behalf Of Nik Martin
> > Sent: Tuesday, May 18, 2004 9:19 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] VoiceMailMain dumps user back into my
> > incoming context after leaving a message
> >
> > I have a dial plan that includes a company phone directory
> as a main
> > menu option. If they just sit at the main menu, after 20 seconds,
> > they are transferred to the operator. If the user picks an
> extension from the
> > directory, they are transferred to the proper extension.
> If the called
> > number is not available, they are transferred into VoiceMailMain.
> > They leave a message, and hang up. The hang up doesn't seem to be
> > detected in VoiceMailMain, and they are sent back into the main
> > incoming context of my incoming dial plan (radiance), which
> after 20
> > seconds transfers them to an operator. The operator answers and is
> > greeted with the very LOUD and annoying "phone is off hook"
> tone. If
> > the operator hangs up, all is well, and all the affected
> channels are
> > cleared. Any tips to this? Busydetect is NO in
> zapata.conf for other
> > reasons (calls being inadvertently dropped by asterisk).
> >
> >
> > My Dialplan:
> >
> > pbxMobile*CLI> show dialplan
> >
> > [ Context 'default' created by 'pbx_config' ]
> > Include => 'radiance'
> > [pbx_config]
> > Ignore pattern => '9'
> >
> > [ Context 'radiance' created by 'pbx_config' ]
> > '9' => 1. Background(radiancedirectory)
> > [pbx_config]
> > 2. DigitTimeout(3)
> > [pbx_config]
> > 3. ResponseTimeout(10)
> > [pbx_config]
> > 'i' => 1. Background(pbx-invalid)
> > [pbx_config]
> > 2. Goto(radiance|s|4)
> > [pbx_config]
> > 's' => 1. Wait(3)
> > [pbx_config]
> > 2. Answer()
> > [pbx_config]
> > 3. NOOP(${CALLERID})
> > [pbx_config]
> > 4. Wait(1)
> > [pbx_config]
> > 5. Background(radiancewelcome) [pbx_config]
> > 't' => 1. Playback(transferring)
> > [pbx_config]
> > 2. Dial(SIP/jsantacapita|20|tT)
> > [pbx_config]
> >
> > Include => 'extensions'
> > [pbx_config]
> >
> >
> >
> >
> > [ Context 'extensions' created by 'pbx_config' ]
> > '.' => 3. Hangup()
> > [pbx_config]
> > '0' => 1. Dial(SIP/jsantacapita|20|Tt)
> > [pbx_config]
> > 2. Voicemail(u100)
> > [pbx_config]
> > 102. Voicemail(b100)
> > [pbx_config]
> > '100' => 1. Dial(SIP/jsantacapita|20|Tt)
> > [pbx_config]
> > 2. Voicemail(u100)
> > [pbx_config]
> > 102. Voicemail(b100)
> > [pbx_config]
> > '101' => 1. Dial(SIP/mthomas|20|Tt)
> > [pbx_config]
> > 2. Voicemail(u101)
> > [pbx_config]
> > 102. Voicemail(b101)
> > [pbx_config]
> > '102' => 1. Dial(SIP/dli|20|Tt)
> > [pbx_config]
> > 2. Voicemail(u102)
> > [pbx_config]
> > 102. Voicemail(b102)
> > [pbx_config]
> > '105' => 1. Dial(SIP/nmartin|20|Tt)
> > [pbx_config]
> > 2. Voicemail(u105)
> > [pbx_config]
> > 102. Voicemail(b105)
> > [pbx_config]
> > '600' => 1. VoiceMailMain()
> > [pbx_config]
> > '601' => 1. MeetMe()
> > [pbx_config]
> > '800' => 1. Dial(Zap/25)
> > [pbx_config]
> > 2. Congestion()
> > [pbx_config]
> > '801' => 1. Dial(Zap/26)
> > [pbx_config]
> > 2. Congestion()
> > [pbx_config]
> > 'h' => 1. Hangup()
> > [pbx_config]
> > 'i' => 1. Hangup()
> > [pbx_config]
> > 't' => 1. Hangup()
> > [pbx_config]
> >
> >
> >
> > [ Context 'parkedcalls' created by 'res_parking' ]
> > '701' => 1. ParkedCall(701)
> > [res_parking]
> > '702' => 1. ParkedCall(702)
> > [res_parking]
> > '703' => 1. ParkedCall(703)
> > [res_parking]
> > '704' => 1. ParkedCall(704)
> > [res_parking]
> > '705' => 1. ParkedCall(705)
> > [res_parking]
> > '706' => 1. ParkedCall(706)
> > [res_parking]
> > '707' => 1. ParkedCall(707)
> > [res_parking]
> > '708' => 1. ParkedCall(708)
> > [res_parking]
> > '709' => 1. ParkedCall(709)
> > [res_parking]
> > '710' => 1. ParkedCall(710)
> > [res_parking]
> > '711' => 1. ParkedCall(711)
> > [res_parking]
> > '712' => 1. ParkedCall(712)
> > [res_parking]
> > '713' => 1. ParkedCall(713)
> > [res_parking]
> > '714' => 1. ParkedCall(714)
> > [res_parking]
> > '715' => 1. ParkedCall(715)
> > [res_parking]
> > '716' => 1. ParkedCall(716)
> > [res_parking]
> > '717' => 1. ParkedCall(717)
> > [res_parking]
> > '718' => 1. ParkedCall(718)
> > [res_parking]
> > '719' => 1. ParkedCall(719)
> > [res_parking]
> > '720' => 1. ParkedCall(720)
> > [res_parking]
> >
> >
> > Nik Martin
> > Lead Software Engineer
> > Radiance Technologies
> > nmartin at radiancetech.com
> > W 251.445.0045 x105
> > C 251.455.4665
> > F 251.445.0046
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
>
>
More information about the asterisk-users
mailing list