similar to: VoiceMail - no user pre-registration

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/bd36cdaa/attachment.htm

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, 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

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

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

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

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

I wonder if I'm the only one who finds Allison reading the timestamp on my voicemails in GMT, after months of having it done with the local time I have set in voicemail.conf. The timestamps on the message files, and the emails that are sent, are correct. Only the spoken dates appear to be affected. Thanks. B.

It seems voicemail recordings have broken sound. It cuts out randomly throughout the recording. Has anyone had any similar experiences? I've included some snips of my voicemail.conf Cheers, Ben ----------SNIP------- [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav ; Who the e-mail notification should appear to come from

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 =>

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

> Hi all, Hi Marco, > I was thinking about integrating an old PBX with Asterisk and I was wondering > some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all