[asterisk-users] Voicemailmain not changing password?
hi all, i am using voicemailmain application in ast 1.4.2. Its not changing my password in the change password menu. i have no idea why. my voicemail configuration is: 25= 52,sipura i always have to enter 52 for password even if i have changed it previously. can anyone tell me why its not changing the password. is it a bug in this apllication or is there something which i have to do to make it work? thanx in advance... -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMailMain plays oldest message first
When I listening to messages, VoiceMailMain always goes from the oldest message to the newest message. For new messages, this order is ok. But for old/archived messages, I would like to hear the reverse order. What can I do? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemailmain
When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten = 2500,1,Wait(2) exten = 2500,2,VoicemailMain(s100) exten = 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail traffic. Thanks --john ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
What do you get in the CLI ? - Original Message - From: John Hill [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 30, 2006 11:24 PM Subject: [asterisk-users] voicemailmain When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten = 2500,1,Wait(2) exten = 2500,2,VoicemailMain(s100) exten = 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail traffic. Thanks --john ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemailmain menu
Hi, Is there way a way to restrict access to certain menus, such as the following: 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Record your temporary message (new in Asterisk v1.2) Thanks in advance, Jack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicemailMain()
Hi! Is this possible to make asterisk follow the dial plan after executing VoicemailMain? Thanks, Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicemailMain()
Didnt quite get ur question. But, if you mean, you want to, for e.g. play a file, dial out another number, sing a song, dance around, after execution of VoicemailMain, yes, its very much possible. Just add your enhanced dialplan at the next priority of VoicemailMain. cheerz - Ben Michel Zenone wrote: Hi! Is this possible to make asterisk follow the dial plan after executing VoicemailMain? Thanks, Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicemailMain()
Michel Zenone wrote:
Hi!
Is this possible to make asterisk follow the dial plan after executing
VoicemailMain?
Happens by default, unless the caller hangs up of course.
; Give voicemail at extension 3509
exten = 3509,1,SetVar(LOOP=1)
exten = 3509,2,Answer
exten = 3509,3,Wait(.5)
exten = 3509,4,GotoIf($[X${RDNIS} = X]?5:10)
exten = 3509,5,VoicemailMain
exten = 3509,6,Wait(.5)
exten = 3509,7,GotoIf($[${LOOP} = 3]?11:8)
exten = 3509,8,SetVar(LOOP=$[${LOOP} + 1])
exten = 3509,9,Goto(5)
exten = 3509,10,VoiceMail(u${RDNIS})
exten = 3509,11,Hangup
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[asterisk-users] voicemailmain errors on CLI
Hello ppl, I am getting the following errors when accessing voicemails Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Sep 13 16:43:59 ERROR[19020]: app.c:1196 ast_unlock_path: Could not unlock path '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Sep 13 16:43:59 ERROR[19020]: app.c:1196 ast_unlock_path: Could not unlock path '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Tho this duznt affect the fetching of voicemails, but do get these errors on the CLI. Wot are they?n any unwanted effects?? cheerz Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain errors on CLI
Benjamin Jacob wrote: Hello ppl, I am getting the following errors when accessing voicemails Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Just as the error states, the directory Old doesn't exist. Check to see if it does. If it is there, check it's permissions, if not then create it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voicemailmain errors on CLI
You have to leave a message in the voicemail, then listen it and the error will not apear again. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Doug Lytle Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] voicemailmain errors on CLI Benjamin Jacob wrote: Hello ppl, I am getting the following errors when accessing voicemails Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Just as the error states, the directory Old doesn't exist. Check to see if it does. If it is there, check it's permissions, if not then create it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
On Thu, Aug 24, 2006 at 04:08:01PM -0400, existx wrote:
Howdy,
I have a Debian box using Debian's Asterisk package.
Just to be clear about the version: I assume that the version is:
http://packages.debian.org/stable/comm/asterisk
(1:1.0.7.dfsg.1-2sarge3 or 1:1.0.7.dfsg.1-2)
If you don't lack disk space on that system, than install the package
asterisk-doc . It will install a huge pile of unnecessary API docs. But
also /usr/share/doc/asterisk-doc/examples with the sample configs.
People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()
My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.
This is normally an issue with detecting the DTMFs in the call. What
phones are the users using? How are they connected to Asterisk?
If those are SIP phones, then both sterisk and the phones need to agree
on the DTMF encoding method. See the dtmfmode option in sip.conf.
(Note that 1.0 does not have dtmfmode=auto)
Also: VoicemailMain can take a argument for a username. Usually the
caller's caller ID will also match its mailbox number (at least for
internal calls). In such a case you can use the following hack:
exten = _299[89],1,Answer
exten = _299[89],2,Wait,2 ; try waiting just 1?
exten = _2998,3,Voicemailmain(s${CALLERIDNUM})
exten = _2999,3,Voicemailmain()
(Note that this is asterisk 1.0 syntax. In Asterisk 1.2 use
Voicemailmain(${CALLERID(num)},s)
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] voicemailmain
Aaron Daniel wrote: Not sure about that Doug. It should read: exten = a,1,VoicemailMan([EMAIL PROTECTED]) You are correct. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemailmain
Howdy, I have a Debian box using Debian's Asterisk package. People can leave voicemail for the extensions that are setup in the configuration, and asterisk e-mail's the user a .wav file (voicemail.conf). This works perfect. However, I want to have VoicemailMain sit on an extension so people can call in, change their greeting, listen too voicemail, etc. extensions.conf: exten = 2999,1,Answer exten = 2999,2,Wait,2 exten = 2999,3,Voicemailmain() My understand is, that this should allow any user to call up. Enter in their mailbox number (currently the same as their extension) and password. However, I cannot dial this extension after reloading asterisk. I'm thinking I should add something in another configuration file, or perhaps my syntax is wrong. Any help would be much apperciated! Thanks in advance. Regards, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
On Friday 25 August 2006 08:39, existx wrote: The error from the CLI is: Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not exist It looks like you have created 2699 in a different context than your phones. You will need to include = the-context to be able to dial the extension. -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
Cristian,
The only other line in extensions.conf that references VoicemailMain is this:
exten = a,1,VoicemailMain(${ARG1})
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist
Regards,
Jason
On 8/24/06, kritikus Araklidas [EMAIL PROTECTED] wrote:
Hi:
First it at all check if you have a different extension for voicemailmain.?
Then use VoiceMailMain syntax.
And send me the CLI log when you try to connect to VoiceMailMain.
regards.
Cristian.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400
Howdy,
I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()
My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.
I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!
Thanks in advance.
Regards,
Jason
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Re: [asterisk-users] voicemailmain
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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Re: [asterisk-users] voicemailmain
Ok you have two optionsthe iax extension is created under default
context???
The VoceMilMain could be configured with the options of wich context use
like this:
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain(@test)
Where test is the context where the iax client belong.
Let me know.
Chers.
Cris.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:39:35 -0400
Cristian,
The only other line in extensions.conf that references VoicemailMain is
this:
exten = a,1,VoicemailMain(${ARG1})
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist
Regards,
Jason
On 8/24/06, kritikus Araklidas [EMAIL PROTECTED] wrote:
Hi:
First it at all check if you have a different extension for
voicemailmain.?
Then use VoiceMailMain syntax.
And send me the CLI log when you try to connect to VoiceMailMain.
regards.
Cristian.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400
Howdy,
I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()
My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.
I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!
Thanks in advance.
Regards,
Jason
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RE: [asterisk-users] voicemailmain
Hi: First it at all check if you have a different extension for voicemailmain.? Then use VoiceMailMain syntax. And send me the CLI log when you try to connect to VoiceMailMain. regards. Cristian. From: existx [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemailmain Date: Thu, 24 Aug 2006 16:08:01 -0400 Howdy, I have a Debian box using Debian's Asterisk package. People can leave voicemail for the extensions that are setup in the configuration, and asterisk e-mail's the user a .wav file (voicemail.conf). This works perfect. However, I want to have VoicemailMain sit on an extension so people can call in, change their greeting, listen too voicemail, etc. extensions.conf: exten = 2999,1,Answer exten = 2999,2,Wait,2 exten = 2999,3,Voicemailmain() My understand is, that this should allow any user to call up. Enter in their mailbox number (currently the same as their extension) and password. However, I cannot dial this extension after reloading asterisk. I'm thinking I should add something in another configuration file, or perhaps my syntax is wrong. Any help would be much apperciated! Thanks in advance. Regards, Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Search from any web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemailmain
Not sure about that Doug. It should read:
exten = a,1,VoicemailMan([EMAIL PROTECTED])
If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.
On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [asterisk-users] voicemailmain
Howdy guys,
Thanks for your help, it works fine without editing the default line of:
exten = a,1,VoicemailMain(${ARG1})
The issue was that I had specified VoicemailMain by the default line,
which was way above the rest of my extensions (out of context).
Hopefully this will help someone in the future.
Regards,
Jason
On 8/24/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Not sure about that Doug. It should read:
exten = a,1,VoicemailMan([EMAIL PROTECTED])
If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.
On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] voicemailmain()
Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemailmain()
Hi! in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? A simple solution is to just edit/remove some of the voice prompts that announce the unwanted options, so the user will not be informed about their existence. Also I want to know if there is a option that erase all message in a user box. You can create that yourself outside of the voicemail application with the appropriate voice prompt and e.g. a simple shell script. Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemailmain()
Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMailMain(@context) Problem with Option 5 (Advanced)
Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I press 3 for advanced options then press 5 to leave a message I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox Im trying to leave a message for is in the same [context] Im currently in. Error at the CLI: Mar 21 14:11:18 WARNING[21294]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '1002' Is there a way to remedy this? Voicemail.conf [default] [context] 1001 = 1001,line1 1002 = 1002,line2 extension.conf exten = *155,1,Voicemailmain(@context) Thanks. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5(Advanced)
I had the same problem yesterday. I thought it might have been a realtime problem. Guess not. Bloody annoying too. -Original Message- From: JR Richardson [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 21, 2006 2:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5(Advanced) Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I press 3 for advanced options then press 5 to leave a message I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I'm trying to leave a message for is in the same [context] I'm currently in. Error at the CLI: Mar 21 14:11:18 WARNING[21294]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '1002' Is there a way to remedy this? Voicemail.conf [default] [context] 1001 = 1001,line1 1002 = 1002,line2 extension.conf exten = *155,1,Voicemailmain(@context) Thanks. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5 (Advanced)
What version of Asterisk are you running? The reason I ask is that I think I remember a fix for this on the svn-commits list awhile back. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Tuesday, March 21, 2006 4:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoiceMailMain(@context) Problem with Option 5 (Advanced) Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I press 3 for advanced options then press 5 to leave a message I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I'm trying to leave a message for is in the same [context] I'm currently in. Error at the CLI: Mar 21 14:11:18 WARNING[21294]: app_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '1002' Is there a way to remedy this? Voicemail.conf [default] [context] 1001 = 1001,line1 1002 = 1002,line2 extension.conf exten = *155,1,Voicemailmain(@context) Thanks. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemailmain() refusing connection problem
Please help for this. I really got stuck at this. After a few tries , asterisk refuses connection anymore until the previous connection timeout. Let me know if you require more info. Regards,Sam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam LeeSent: Thursday, February 09, 2006 3:44 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Voicemailmain() refusing connection problem I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until the 6th time when it will give up and say " Feb 9 14:18:15 WARNING[22945]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 731b65f6-7dec21 " I don't understand, is it waiting for some reply from OPENSER which never came ? or what ? I don't know whether its the same problem, but ifi call 400 a couple of times to access the VoicemailMain() without actually going in (once i've hear the password prompt, i hangup , simulating a DoS attack) , asterisk refuses to take anymore call at extension 400 for VoicemailMain() . Please let me know if you don't understand what i mean. Please help! Regards,Sam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemailmain() refusing connection problem
I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until the 6th time when it will give up and say " Feb 9 14:18:15 WARNING[22945]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 731b65f6-7dec21 " I don't understand, is it waiting for some reply from OPENSER which never came ? or what ? I don't know whether its the same problem, but ifi call 400 a couple of times to access the VoicemailMain() without actually going in (once i've hear the password prompt, i hangup , simulating a DoS attack) , asterisk refuses to take anymore call at extension 400 for VoicemailMain() . Please let me know if you don't understand what i mean. Please help! Regards,Sam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMailMain Pass Mailbox
I have a extension 981 setup for entering VoiceMailMain: exten = 981,1,VoiceMailMain,([EMAIL PROTECTED]) exten = 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready.*CLI -- Executing VoiceMailMain(SIP/2504-ba66, [EMAIL PROTECTED]) in new stack -- Playing 'vm-login' (language 'en') Any ideas? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMailMain Pass Mailbox
use ${CALLERIDNUM} instead of [mailbox]
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and mailbox
numbers are the same).
This dosen't seem to work. I get this in the console:
Asterisk Ready.
*CLI -- Executing VoiceMailMain(SIP/2504-ba66, [EMAIL PROTECTED]) in
new stack
-- Playing 'vm-login' (language 'en')
Any ideas?
Thanks!!
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RE: [Asterisk-Users] VoiceMailMain Pass Mailbox
[mailbox] does not exist
use
exten = 981,1,VoiceMailMain,(${CALLERID(num)}@usvm)
this
is provided that your callerid settings in your sip, iax, and zap configs are
correct and relect the extension calling.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
BeckSent: Wednesday, January 04, 2006 11:43 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
VoiceMailMain Pass Mailbox
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and
mailbox numbers are the same).
This dosen't seem to work. I get this in the console:
Asterisk Ready.*CLI -- Executing
VoiceMailMain("SIP/2504-ba66", "[EMAIL PROTECTED]") in new
stack -- Playing 'vm-login' (language
'en')
Any ideas?
Thanks!!
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Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
On Sun, Oct 30, 2005 at 10:02:24AM -0500, David Bandel wrote: Perhaps a good enhancement would be a syntax checker for the various .conf files. There is a vim syntax file floating around. Also an emacs mode. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
And, I couple of times now I have offered to post a BBEdit language module to the wiki, but have no idea where to put it. Last chance for anyone who's interested... Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 5-Nov-05, at 12:03 AM, Tzafrir Cohen wrote: On Sun, Oct 30, 2005 at 10:02:24AM -0500, David Bandel wrote: Perhaps a good enhancement would be a syntax checker for the various .conf files. There is a vim syntax file floating around. Also an emacs mode. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
I'm interested. Thanks, Waldo On Nov 5, 2005, at 2:54 PM, Anthony Rodgers wrote: And, I couple of times now I have offered to post a BBEdit language module to the wiki, but have no idea where to put it. Last chance for anyone who's interested... Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 5-Nov-05, at 12:03 AM, Tzafrir Cohen wrote: On Sun, Oct 30, 2005 at 10:02:24AM -0500, David Bandel wrote: Perhaps a good enhancement would be a syntax checker for the various .conf files. There is a vim syntax file floating around. Also an emacs mode. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
Here you go - place it in ~/Library/Application Support/BBEdit/ Language Modules. It's not complete, but I add new keywords to it as I go along. It is also case-sensitive (my preference - you can turn this off). AsteriskCodelessLanguageModule.plist Description: Binary data I'd like to put this on the wiki, but have no idea where it should go. Do I just create a brand new page? Any thoughts? Anyone? On 5-Nov-05, at 11:59 AM, Waldo Rubinstein wrote: I'm interested. Thanks, Waldo On Nov 5, 2005, at 2:54 PM, Anthony Rodgers wrote: And, I couple of times now I have offered to post a BBEdit language module to the wiki, but have no idea where to put it. Last chance for anyone who's interested... Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 5-Nov-05, at 12:03 AM, Tzafrir Cohen wrote: On Sun, Oct 30, 2005 at 10:02:24AM -0500, David Bandel wrote: Perhaps a good enhancement would be a syntax checker for the various .conf files. There is a vim syntax file floating around. Also an emacs mode. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] VoiceMailMain() in 1.2-beta
Folks, * newbie trying out 1.2-beta. Want to make sure I haven't missed some dialplan invocations (or perhaps waving of chicken feet, etc.). calling voicemailmain() works for me to the point I get to hear the message left by someone. However, the * docs I've read don't seem to say much, so I _ass-u-me_ it is complete (i.e., prompts to delete, save, etc.). However, that doesn't seem to be the case. After hearing the message, I'm disconnected (hangup()). Pointers to the correct FM to RTFM appreciated. Need to incorporate the usual press 3 to delete, 7 to save, 9 to skip to the next message prompts. (Odd no examples in the extension.conf.samples for this.) What did I miss? TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
On Sun, 2005-10-30 at 09:03 -0500, David Bandel wrote: Pointers to the correct FM to RTFM appreciated. Need to incorporate the usual press 3 to delete, 7 to save, 9 to skip to the next message prompts. (Odd no examples in the extension.conf.samples for this.) Well www.asteriskdocs.org has the orielly asterisk book that was just published. If you wana read docs off that site or the orielly book there is a good start. www.voip-info.org is also another good resource to do some reading. If you google: asterisk cmd voicemailmain you will see a link as the first item or very near it to voip-info.org that has the correct page there for you. As for why its not working, have you tried using the CLI and seeing if there are any messages displayed when you dial in? My guess is that there is either a permission problem or missing sounds or something, but I dont have enough information to say. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
Solved. On 10/30/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Sun, 2005-10-30 at 09:03 -0500, David Bandel wrote: Pointers to the correct FM to RTFM appreciated. Need to incorporate the usual press 3 to delete, 7 to save, 9 to skip to the next message prompts. (Odd no examples in the extension.conf.samples for this.) Well www.asteriskdocs.org has the orielly asterisk book that was just published. If you wana read docs off that site or the orielly book there is a good start. Have the OReilley book. Also the new 1.2 book from asteriskdocs.org. www.voip-info.org is also another good resource to do some reading. If you google: asterisk cmd voicemailmain you will see a link as the first item or very near it to voip-info.org that has the correct page there for you. thanx for the link. Good info here. As for why its not working, have you tried using the CLI and seeing if there are any messages displayed when you dial in? My guess is that there is either a permission problem or missing sounds or something, but I dont have enough information to say. using the CLI in - mode showed the problem. Apparently, I can't spell (or I can, but when I was typing, I transposed two letters and made it vm-recieved vice vm-received). Perhaps a good enhancement would be a syntax checker for the various .conf files. Ciao, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
using the CLI in - mode showed the problem. Apparently, I can't spell (or I can, but when I was typing, I transposed two letters and made it vm-recieved vice vm-received). Perhaps a good enhancement would be a syntax checker for the various .conf files. Been there... sure wish telnet/putty had a spell checker. ;) ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
On 10/30/05, David Bandel [EMAIL PROTECTED] wrote: Have the OReilley book. Also the new 1.2 book from asteriskdocs.org. Pt... they're the same book :) -- Leif Madsen - http://www.leifmadsen.com http://www.asteriskdocs.org -- Co-Founder http://www.oreilly.com/catalog/asterisk -- Co-Author ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] VoiceMailMain() in 1.2-beta
On 10/30/05, Leif Madsen [EMAIL PROTECTED] wrote: On 10/30/05, David Bandel [EMAIL PROTECTED] wrote: Have the OReilley book. Also the new 1.2 book from asteriskdocs.org. Pt... they're the same book :) OK, well, I have two and they are definitely different books. For one thing, one has color drawings, the other has only words, they're different length (the one with pictures is shorter). Sooo not sure what I have but they are different. Ciao, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.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
Re: [Asterisk-Users] Voicemailmain automatic extension detection?
On Wed, Oct 05, 2005 at 09:14:09PM +0100, Kevin Walsh wrote:
Do you mean something like VoiceMailMain(${CALLERIDNUM})?
Yes, that works nicely. Thank you!
--
Mason Loring Bliss [EMAIL PROTECTED] Cthulhu fhtagn!
http://blisses.org/awake ? sleep : random() 2 ? dream : sleep;
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[Asterisk-Users] Voicemailmain automatic extension detection?
Is there a way I can have voice mail check calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for clues! -- Mason Loring Bliss [EMAIL PROTECTED] http://blisses.org/ Anything can be impossible, given sufficient bureaucracy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemailmain automatic extension detection?
On 15:46, Wed 05 Oct 05, Mason Loring Bliss wrote:
Is there a way I can have voice mail check calls coming from my internal
users automatically get to the right extension, without having the user
enter their extension?
I'm thinking that I could have the local SPA boxes translate, or have
each user live in a context where the extension in question exists
uniquely per user, but both of these seem kludgey.
Give the users a voicemailbox with the same number as their
callerid number.
Then add something like this in your extensions.conf (taken
from my own private setup)
exten = 8500,1,VoicemailMain(${CALLERIDNUM})
On another system I implemented it so users call their own
extension number to reach voicemail:
exten = 2001/2001,1,VoicemailMain(2001)
The first method is easier in a larger setup, since it
matches all users/voicemailboxes with only one line in
extensions.conf
Good luck
--
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http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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RE: [Asterisk-Users] Voicemailmain automatic extension detection?
Mason Loring Bliss [EMAIL PROTECTED] wrote:
Is there a way I can have voice mail check calls coming from my internal
users automatically get to the right extension, without having the user
enter their extension?
I'm thinking that I could have the local SPA boxes translate, or have
each user live in a context where the extension in question exists
uniquely per user, but both of these seem kludgey.
Thanks in advance for clues!
Do you mean something like VoiceMailMain(${CALLERIDNUM})?
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_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
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Re: [Asterisk-Users] Voicemailmain automatic extension detection?
On Wed, 2005-10-05 at 15:46 -0400, Mason Loring Bliss wrote:
Is there a way I can have voice mail check calls coming from my internal
users automatically get to the right extension, without having the user
enter their extension?
I'm thinking that I could have the local SPA boxes translate, or have
each user live in a context where the extension in question exists
uniquely per user, but both of these seem kludgey.
Thanks in advance for clues!
I use this in extensions.conf:
exten = 999,1,Answer(); Voicemail call number
exten = 999,2,Wait(1);
exten = 999,3,VoicemailMain(${CALLERIDNUM}); This requires username of SIPs to
be their VM box #
Users are still asked for password, but an added 's' above (I forget
exactly where) will make that go away too.
--
Jesse Keating
GameHouse -- Systems Engineer
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[Asterisk-Users] VoiceMailMain issue..
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
Why do I get Forbidden 403 and one console display :
Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No
application 'VoiceMailMain ' for extension (home, 22999, 1)
Anybody knows why?
Ciao and thank you!
Mauro Zanin
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Re: [Asterisk-Users] VoiceMailMain issue..
what does your voicemail.conf and sip.conf look like?
Mark
On 7/25/05, Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
Why do I get Forbidden 403 and one console display :
Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No
application 'VoiceMailMain ' for extension (home, 22999, 1)
Anybody knows why?
Ciao and thank you!
Mauro Zanin
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--
regards,
Mark P. Edwards
FWD: 667917
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Re: [Asterisk-Users] VoiceMailMain issue..
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
Hi everybody,
Hi Mauro!
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
your dialplan looks good
Why do I get Forbidden 403 and one console display :
Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No
application 'VoiceMailMain ' for extension (home, 22999, 1)
Anybody knows why?
Have you checked /usr/lib/asterisk/modules/ and made sure that
app_voicemail.so is there?
Christoph
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Re: [Asterisk-Users] VoiceMailMain issue..
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
Don't put extra spaces in extensions.conf
exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
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[Asterisk-Users] VoiceMailMain login problem... new BUG ?
Hello
I think I've found a new bug, but first I'm asking for experts...
I have the following simple configuration :
in extensions.conf :
exten = 0660,1,VoicemailMain(${CALLERIDNUM})
So the caller is directly connected to his mailbox, it works great with
other users (like xlite, 0467161616, nfovdt...) but with the user pnunes :
When I tring to connect with the user pnunes I cannot enter into the
mailbox... it seems there is a mistake somewere (see the folder who is
nunespnunes instead of pnunes). Any idea ?
*CLI -- Executing VoiceMailMain(SIP/petitvillage-0813b1e0, pnunes)
in new stack
-- Playing 'vm-login' (language 'en')
-- No username but # key pressed. Using CID 'pnunes'
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-helpexit' (language 'en')
-- Playing 'vm-options' (language 'en')
-- Recording the message
-- Playing 'vm-rec-busy' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing: voicemail/default/nunespnunes/busy format: wav49,
0x80f0130
-- x=1, open writing: voicemail/default/nunespnunes/busy format: gsm,
0x80ed3b8
-- x=2, open writing: voicemail/default/nunespnunes/busy format: wav,
0x814ac70
PS: I'm using MYSQL Voicemail and the database seems correctly invoked :
SELECT password,fullname,email,pager,options FROM users WHERE
context='default' AND mailbox='pnunes'
Thanks for advice
Nicolas
http://www.call.fr
PS: We're planning making a small page on our VideoVoicemail test, it works
perfectly at this moment...
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Re: [Asterisk-Users] voicemailmain hotkey
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote: I'm having a similar problem. Do you have operator=yes in your voicemail.conf under [general]? Argh, thats it, solved! Thanks a lot :) ...cut -- Tho/\/\as ___ 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
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten = _8500, 1, Wait(2)
exten = _8500, 2, VoicemailMain(${CALLERIDNUM})
exten = _8500, 3, Hangup
You don't mention the type of phone you're using, but on our setup with
SIP phones, we add a sipdtmfmode(inband) to what you have above. You
might try fiddling with that.
-russ
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RE: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
It BT100. it works.
thanks!
steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Russ
Beaupre, P.E.
Sent: Sunday, December 19, 2004 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoicemailMain can't read from phone
keyboard!
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called,
it
prompts for mailbox and password. It seems not able to read from the
phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten = _8500, 1, Wait(2)
exten = _8500, 2, VoicemailMain(${CALLERIDNUM})
exten = _8500, 3, Hangup
You don't mention the type of phone you're using, but on our setup with
SIP phones, we add a sipdtmfmode(inband) to what you have above. You
might try fiddling with that.
-russ
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Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Steven Wang wrote: Hello I try to set up voicemails for extension. When VoicemailMain gets called, it prompts for mailbox and password. It seems not able to read from the phone. So the authentication always fails. This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ 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
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. Yep, and the BT will only work right with certain codecs. I think it's iLBC that suddenly won't recognize DTMF while it works with the same setting in ULAW, for example. I keep forgetting why I don't use iLBC on the BT, set it up, and then find DTMF b0rken with dtmfmode=info ___ 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
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Wilson Pickett wrote: This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. Yep, and the BT will only work right with certain codecs. I think it's iLBC that suddenly won't recognize DTMF while it works with the same setting in ULAW, for example. I keep forgetting why I don't use iLBC on the BT, set it up, and then find DTMF b0rken with dtmfmode=info As most people know inband DTMF only works with the ulaw and alaw codecs. This is a codec issue, not an Asterisk issue. I thought GS fixed the need for INFO mode DTMF. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ 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] voicemailmain hotkey
Hi Folks Since updated to 1.0.1/2 I got a prob with the hotkey to access voicemailmain. According to the wiki 0 jumps to extension oand* to a 0 isn't working, I get vm-sorry followed by HangUp :( * is working and I get access. So I changed the dialplan to get my voicemail managed. Tested on zaphfc and capi Is there something new/changed? Any hints? Thanks -- Tho/\/\as ___ 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] VoiceMailMain(sexten@context) doesn't work in CVS 11/03
Can anyone else verify this or is it just me? -- MBM ___ 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
RE: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
Message: 1 Date: Fri, 5 Nov 2004 09:31:27 -0500 From: Matthew Marlowe [EMAIL PROTECTED] Subject: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't work in CVS 11/03 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Can anyone else verify this or is it just me? Yes, I can't get it to work either. I get no audio out. It says it is playing, but nothing comes out. I've tried the various formats, but none seem to work. ___ 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
Re: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
exten = 55,1,Voicemailmain([EMAIL PROTECTED]) works fine for me with latest CVS. Matthew - Original Message - From: Noah Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, November 05, 2004 10:05 AM Subject: RE: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't Message: 1 Date: Fri, 5 Nov 2004 09:31:27 -0500 From: Matthew Marlowe [EMAIL PROTECTED] Subject: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't work in CVS 11/03 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Can anyone else verify this or is it just me? Yes, I can't get it to work either. I get no audio out. It says it is playing, but nothing comes out. I've tried the various formats, but none seem to work. ___ 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
Re: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
It works for me but it asks for the password. No audio problems. On Fri, 5 Nov 2004 11:25:29 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: exten = 55,1,Voicemailmain([EMAIL PROTECTED]) works fine for me with latest CVS. Matthew - Original Message - From: Noah Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, November 05, 2004 10:05 AM Subject: RE: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't Message: 1 Date: Fri, 5 Nov 2004 09:31:27 -0500 From: Matthew Marlowe [EMAIL PROTECTED] Subject: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't work in CVS 11/03 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Can anyone else verify this or is it just me? Yes, I can't get it to work either. I get no audio out. It says it is playing, but nothing comes out. I've tried the various formats, but none seem to work. ___ 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 -- MBM ___ 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
Re: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
This seem to be fixed in CVS 11/05 - Altho ALERT_INFO is still broken in CVS 11/05 On Fri, 5 Nov 2004 12:52:45 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: It works for me but it asks for the password. No audio problems. On Fri, 5 Nov 2004 11:25:29 -0600, Matthew Boehm [EMAIL PROTECTED] wrote: exten = 55,1,Voicemailmain([EMAIL PROTECTED]) works fine for me with latest CVS. Matthew - Original Message - From: Noah Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, November 05, 2004 10:05 AM Subject: RE: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't Message: 1 Date: Fri, 5 Nov 2004 09:31:27 -0500 From: Matthew Marlowe [EMAIL PROTECTED] Subject: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't work in CVS 11/03 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Can anyone else verify this or is it just me? Yes, I can't get it to work either. I get no audio out. It says it is playing, but nothing comes out. I've tried the various formats, but none seem to work. ___ 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 -- MBM -- MBM ___ 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
Re: [Asterisk-Users] VoiceMailMain(sexten@context) doesn't
On Fri, 5 Nov 2004, Matthew Marlowe wrote: This seem to be fixed in CVS 11/05 - Altho ALERT_INFO is still broken in CVS 11/05 Isn't this an effect of the new automatic variable inheritance? Since ALERT_INFO is used in the called channel you would have to set _ALERT_INFO instead of ALERT_INFO? As Michalis Manousos wrote in an email to asterisk-dev earlier today: The new dial app does not copy to the new channel created by it just some special variables (like the ALERT_INFO). It copies channel variables based on their name. If the first character of the variable's name is '_' then the variable is copied to the channel and the initial underscore is removed (so, a second dial won't pass the variable). If the variable's name start with '__' (two underscores) then the variable is copied to the new channel without removing the underscores (so, additional dial()s will always copy this variable. If the variable's name doesn't start with underscore, the variable is not copied. For your case, set an _ALERT_INFO variable and it will work. Peter ___ 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] VoicemailMain Issues
I have a very bizarre issue for ya'll. Asterisk seems to crash after I hang up on VoicemailMain, but only if the user logs in. I am completely dumbfounded with this. We have been running our production system on asterisk HEAD 7/14/2004 for a few weeks now, and this error only happened when I updated to 8/4/2004. I am calling my voicemail extension via X-Lite, and the error message received on the console when asterisk crashes is simply Killed. Has anyone else seen this issue before? I am just trying to figure out if it is something in my config or if there my be a problem with CVS 8/4/2004. Thanks for your help, Robert Jackson ___ 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
Re: [Asterisk-Users] VoicemailMain Issues
On Fri, 2004-08-06 at 11:56, Robert Jackson wrote: I have a very bizarre issue for ya'll. Asterisk seems to crash after I hang up on VoicemailMain, but only if the user logs in. I am completely dumbfounded with this. We have been running our production system on asterisk HEAD 7/14/2004 for a few weeks now, and this error only happened when I updated to 8/4/2004. I am calling my voicemail extension via X-Lite, and the error message received on the console when asterisk crashes is simply Killed. Has anyone else seen this issue before? I am just trying to figure out if it is something in my config or if there my be a problem with CVS 8/4/2004. I don't have an answer to your particular problem but in general it sounds like a SEGFAULT or some other similar bug. Try this... 1. Start Asterisk with safe_asterisk 2. Cause asterisk to crash the way you describe. Asterisk will dump a core file into /tmp 3. Enter gdb asterisk /tmp/core. (you need to have gdb installed of course) 4. Enter bt while in gdb (or do a bt full) to see the back trace. You will probably see an Address out of range or similar error in the last function call on the stack. Take note of the function where the error occurred an the parameter that had the out-of-bounds memory address (if applicable). If your C skills are up to snuff to can try and debug it yourself. If not open up a bug on the bug tracker with all of the info you have collected. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] VoicemailMain Issues
I am not sure what is going on, but * is restarting itself every time a user hangs up after calling to check their voicemail. I am running CVS-HEAD-07/26/04-22:14:48, and this problem just started happening after I updated last night. I am rolling back to CVS-7/14/2004 so that we can keep working, but we need to address the voicemail issue. I will open a bug if this is not just something on my end. Anybody else having issues? Robert Jackson ___ 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] 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
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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
[Asterisk-Users] VoiceMailMain skipping extension and password prompting
OK, Here is a down and dirty which will work in limited situations (like
when there are not to many extensions to re-define - which is one of the
things I want to avoid)... The channel is the first parameter passed to
[globals]
Zap/5-=s6147
Zap/16=s6158
exten = 6199,1,GoToIf(${${CHANNEL:0:6}}?6199|2:6199|4)
exten = 6199,2,VoicemailMain2(${${CHANNEL:0:6}})
exten = 6199,3,Hangup
exten = 6199,4,VoicemailMain2
exten = 6199,5,Hangup
John
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[Asterisk-Users] VoiceMailMain skipping extension and password prompting
I would like to access VoiceMailMain2 skipping extension and password
prompting if calling from a resource that has a mailbox defined. What
variables can I use to retrieve the calling channel calling extension (if
it exists)?
Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox}
is not a real way to retrieve this info)...
exten = 7799,1,gotoif(${CallingResourse.MailBox}?7799|2:7799|4)
exten = 7799,2,VoicemailMain2(s${CallingResourse.MailBox})
exten = 7799,4,VoicemailMain2
... currently extensions.conf is ...
exten = 7799,1,VoicemailMain2
... and from zapata.conf ...
callerid=TCC hcaar 321-222-2553
mailbox=7731
channel=19
Any suggestions?
John Harragin
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Re: [Asterisk-Users] VoiceMailMain skipping extension and password prompting
Hi,
At 12:13 25-9-2003 -0400, you wrote:
I would like to access VoiceMailMain2 skipping extension and password
prompting if calling from a resource that has a mailbox defined. What
variables can I use to retrieve the calling channel calling extension (if
it exists)?
Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox}
is not a real way to retrieve this info)...
exten = 7799,1,gotoif(${CallingResourse.MailBox}?7799|2:7799|4)
exten = 7799,2,VoicemailMain2(s${CallingResourse.MailBox})
exten = 7799,4,VoicemailMain2
... currently extensions.conf is ...
exten = 7799,1,VoicemailMain2
... and from zapata.conf ...
callerid=TCC hcaar 321-222-2553
mailbox=7731
channel=19
Any suggestions?
Actually, I've experienced that its not always -just- that local extension
that you want to give this kind of access to, so I've written some AGI code
that helps. Here is a snippet that gives you the general idea...
=
// parse agi headers into array $agi[callerid]
while ($env=read()) {
errlog($env);
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
// Main run
$clid = $agi[callerid];
switch($clid) {
// enter the mailbox number for each valid callerid
// prepend 's' if you wish to trust the callerid and skip the password check
case 3001: $parms = s1; break;
case 06123456:$parms = s1; break;
default: $parms = 0; break;
}
if($parms != ) $parms = $parms;
echo EXEC VoiceMailMain2$parms\n;
=
I find this approach more flexible. Hope this gets you somewhere.
Best regards,
Florian
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Re: [Asterisk-Users] VoicemailMain
You have to modify the sourcer code yourself. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 15, 2003 6:20 PM Subject: [Asterisk-Users] VoicemailMain Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying mailbox, I would like it to say something like please enter your mailbox number now. Is there a way for me to do this? I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu over again. Would this be a VoicemailMain problem or could it be a problem since the call is being passed over IAX? Again, thanks in advance guys. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicemailMain
On Sunday 15 June 2003 20:20, [EMAIL PROTECTED] wrote: Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying mailbox, I would like it to say something like please enter your mailbox number now. Is there a way for me to do this? All of the sounds are in the /var/lib/asterisk/sounds directory. You're certainly welcome to re-record them as you see fit. Note, however, that every time you do a 'make install', the sounds will be overwritten. I recommend that you store your sounds elsewhere and symlink them into place. I have the following patch for my Makefile to do this: Index: Makefile === RCS file: /usr/cvsroot/asterisk/Makefile,v retrieving revision 1.14 diff -u -r1.14 Makefile --- Makefile13 May 2003 20:37:08 - 1.14 +++ Makefile16 Jun 2003 03:14:55 - @@ -202,6 +202,9 @@ exit 1; \ fi; \ done + for x in /var/lib/asterisk/sounds/tilghman/*.*; do \ + ln -sf $$x $(ASTVARLIBDIR)/sounds ; \ + done mkdir -p $(ASTVARLIBDIR)/mohmp3 mkdir -p $(ASTVARLIBDIR)/images for x in images/*.jpg; do \ I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu over again. Would this be a VoicemailMain problem or could it be a problem since the call is being passed over IAX? Again, thanks in advance guys. This would probably be functionality which hasn't yet been added. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
