similar to: 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

I apologize, but the fix I was thinking of wasn't directly related to this. It was in app_voicemail.c, but related to using the channel's context for the Directory application. The fix for your issue may be indirectly related, though. I would open a bug. Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com

As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I do seem to remember that in 1.2, it wasn't possible to send a message to ones-self. This bug fix apparently corrects that situation. Well, I guess it would, if only it

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

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e9dceeb4/attachment.htm -------------- next part -------------- Hello One of the option in VoicemailMain() is "Adavance Options". Could anyone explain what are these ?. Because whenever I select Advance Options, it repeats the same process of asking "Change Folders,Advance

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

I (sometimes) get this error message: WARNING[17191]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for 'irstname.lastname' I can see the value of the argument is "firstname.lastname" when this line executes in the std-exten macro: exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable... But the error message drops the first character. It

I have a extension 981 setup for entering VoiceMailMain: exten => 981,1,VoiceMailMain,([mailbox]@usvm) 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",

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

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten

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

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =

Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten => 201,1,VoicemailMain(@default) exten => 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see it in the debug output of the Asterisk CLI. The call only stops if before hung up, I

Hi, I have been trying to do numerical differentiation using R. I found some old S code using Richardson Extrapolation which I managed to get to work. I am posting it here in case anyone needs it. ######################################################################## richardson.grad <- function(func, x, d=0.01, eps=1e-4, r=6, show=F){ # This function calculates a numerical approximation

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Dear Asterisk Users, I'm a Japanese and now configuring Voicemail. Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences. What I want to do are... 1) Add words used in message retrieving guidance. I need to add different suffixes to numeric words due to Japanese way of mentioning time. (e.g. in English, you can say "Five

Hi in asterisk 1.4, I need to cancel the password check and allow users enter in the mailbox without entering password. I tried this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,n,VoicemailMain(${Mailbox}@default,s) exten => 911119,n,Hangup and this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,2,VoicemailMain(s) exten => 911119,n,Hangup But it does not work,