similar to: how many voicemail box asterisk can support

hi: after change voicemail.conf. instead of "restart now", is there any other way to let it take effect? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/d9701dca/attachment.htm

Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel not found". Thanks. Yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040722/4608e1e5/attachment.htm

Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

Hi all Looking for a solution to create a flexible voicemail solution in Asterisk without the need to preregister the voicemail users (via databases etc etc). Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. I am looking for * to do the following: * Call comes in *

Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi _________________________________________________________________ Convierte las fotos que m?s te gustan en tu nuevo fondo de escritorio para el ordenador. Es f?cil y adem?s gratis

Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and ring... any ideas? They are setup with a voicemailbox, and it is set to transfer after 15 seconds of

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

syslinux-owner at zytor.com: how can i run isolinux on my cf disk? Yuri++?yhuang at i-security.com.cn 2005-12-13

On Mon, Jun 22, 2020 at 4:02 PM John Hubbard <jhubbard at nvidia.com> wrote: > > On 2020-06-22 15:33, Yang Shi wrote: > > On Mon, Jun 22, 2020 at 3:30 PM Yang Shi <shy828301 at gmail.com> wrote: > >> On Mon, Jun 22, 2020 at 2:53 PM Zi Yan <ziy at nvidia.com> wrote: > >>> On 22 Jun 2020, at 17:31, Ralph Campbell wrote: > >>>> On

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's

I've got a voicemailbox with one message store. When I try to read it, I get the followiing error: ast_openstream_full: File digits/1F does not exist in any format Obviously, I can just clear out that mailbox, but is this a bug that I should be reporting? /Per Jessen, Z?rich

On Mon, Jun 22, 2020 at 3:30 PM Yang Shi <shy828301 at gmail.com> wrote: > > On Mon, Jun 22, 2020 at 2:53 PM Zi Yan <ziy at nvidia.com> wrote: > > > > On 22 Jun 2020, at 17:31, Ralph Campbell wrote: > > > > > On 6/22/20 1:10 PM, Zi Yan wrote: > > >> On 22 Jun 2020, at 15:36, Ralph Campbell wrote: > > >> > > >>> On

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

On 22 Jun 2020, at 17:31, Ralph Campbell wrote: > On 6/22/20 1:10 PM, Zi Yan wrote: >> On 22 Jun 2020, at 15:36, Ralph Campbell wrote: >> >>> On 6/21/20 4:20 PM, Zi Yan wrote: >>>> On 19 Jun 2020, at 17:56, Ralph Campbell wrote: >>>> >>>>> Support transparent huge page migration to ZONE_DEVICE private memory. >>>>> A

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>