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: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Tuesday, May 18, 2004 9:19 AM
To: [EMAIL PROTECTED]
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
[EMAIL PROTECTED]
W 251.445.0045 x105
C 251.455.4665
F 251.445.0046
RE: [Asterisk-Users] VoiceMailMain dumps user back into my incoming context after leaving a message
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Sent: Tuesday, May 18, 2004 9:45 AM
To: [EMAIL PROTECTED]
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: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Tuesday, May 18, 2004 9:19 AM
To: [EMAIL PROTECTED]
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
RE: [Asterisk-Users] VoiceMailMain dumps user back into my incoming context after leaving a message
Nik Martin [EMAIL PROTECTED] wrote: Do you mean after the Voicemail (vs. after VoiceMailMain?) in each extension? Add a call to Hangup at the point where you'd like the call to terminate. exten = 0,1,Dial(SIP/jsantacapita,20,Tt) exten = 0,2,Voicemail(u100) exten = 0,102,Voicemail(b100) Modify your extension definition to look like this: exten = 0,1,Dial(SIP/jsantacapita,20,Tt) exten = 0,2,Voicemail(u100) exten = 0,3,Hangup exten = 0,102,Voicemail(b100) exten = 0,103,Hangup By the way, I see you're using Tt as a Dial parameter. Do you really want your incoming callers to be able to transfer the call? I imagine that someone could have fun playing with that facility. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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
