similar to: RE: VoiceMailMain(@context) Problem with Opt ion 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 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:jr.richardson@cox.net] > 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

Do you mean the peristence of connecting a specific phone to a specific server? If so, then it's relatively easy. The ldirectord has a persistence setting that does that. If I'm misunderstanding you, then could you explain further what you mean? Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and failover of the Asterisk boxes, and my dialplan is set up so that it need not be changed

This is mostly in a traditional pbx-like setup. That is, these are individual remote offices of a larger corporation each with their own cluster (or clusters, in the case of one site). So there is no NAT, and it is an Asterisk-only solution (at least insofar as telephony software is concerned). Regards, - Brad _____ From: asterisk-users-bounces@lists.digium.com on behalf of David Thomas

I understand what you're saying now. While I have absolutely no proof of this, I have to believe that it's something they've solved. I've got several production systems (since early December of last year) using the type of cluster that I'm talking about, and I've yet to hear of any issues that could be related to this. I also did extensive testing both in the lab and at

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 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

Ah... I searched for half an hour for this function... you know, the help function in R could really be a lot better... But wait a minute... looking at this, it appears you have to pass in an expression. What if it is an unknown function, where you only have a handle to the function, but you cannot see it's implementation ? Will this work then ? -----Original Message----- From: Berton Gunter

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

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

Hello list. The VoiceMailMain()-application has an advanced menu. Can I get a Voicemail-application that has less functionality ? I only want the user to listen to new voicemail-messages (and delete them), not the functionality with the folders and redirecting messages to other mailboxes... I've looked at the code in /usr/src/asterisk-1.4.30/apps/app_voicemail.c but it seems complicated

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

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > JR Richardson > Sent: Tuesday, December 05, 2006 3:49 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] RE: regcontext,NoOp extension > vanishes when extension reload > > > > > Let me guess: The

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

Hello guys, Is there a decent click-to-dial CTI which works well with Asterisk? We have vanilla asterisk implementation and I have tried a few (ADA, Outcall etc) but they have poor documentation and don't work very well. We are looking for an application which can allow us to dial a number from Outlook and IE/Firefox for outbound calls and get a pop-up for inbound calls with call history

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- 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

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

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",