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
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
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
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
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
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:
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?
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 =
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:
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
-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
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
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.
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
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
[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
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
@lists.digium.com
Subject: Re: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:39:35 -0400
Cristian,
The only other line in extensions.conf that references VoicemailMain is
this:
exten = a,1,VoicemailMain(${ARG1})
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241
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
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
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
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
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
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
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
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
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
, 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
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
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
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
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
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 |
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:
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
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
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
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
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
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
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
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
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;
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
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
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
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,
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${CALLERIDNUM})
exten =
what does your voicemail.conf and sip.conf look like?
Mark
On 7/25/05, Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
Hi everybody,
Hi Mauro!
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain
Hello
I think I've found a new bug, but first I'm asking for experts...
I have the following simple configuration :
in extensions.conf :
exten = 0660,1,VoicemailMain(${CALLERIDNUM})
So the caller is directly connected to his mailbox, it works great with
other users (like xlite, 0467161616,
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
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
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
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
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
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
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.
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:
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
: [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]
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
, 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
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
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
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
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
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
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
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
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)
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 =
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
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
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
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
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