similar to: Voicemail Caller ID

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

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

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

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 *

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:

Hi - I just deployed a system using IMAP Voicemail. During my testing, voicemail worked fine. I could check vm from the phone, and the messages would get marked as read, or I could read the messages in a mail client, and the phone's mwi light would turn off. Very neat. I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now

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'm trying to config a couple of things on Asterisk CVS-03/22/04-16:41:51. The number shows up, but I can't get the "words" to show on a local bell line. The text always comes up as "unavailable". In sip.conf for each extension, I've tried: callerid="VERTEX" <2142618000> callerid=VERTEX <2142618000> Neither one works. Suggestions? On the

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

I'm having a little problem with voicemail. Actually I'm not getting the ability to leave a voicemail-message. This is part of the dialplan : > exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1}) > exten => s,n,NoOp(${ARG1}@boxes) > exten => s,n,Voicemail(${ARG1}@boxes) > exten => s,n,Hangup() > exten => s,n,MacroExit This is the

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

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've noticed a few differences in my recent upgrade from 1.4 to 1.6.2 that have me baffled. I thought I'd write to the list and see if anyone has any ideas. - In 1.4, app Directory matched users based on the name listed in their voicemail.conf entry. Now it appears that 1.6 matches only on the full name in the users.conf file and ignores the definition in voicemail.conf - In

I am trying to create a system as follows: incoming call --> ivr --> sent to dummy extension 1000 --> redirects user to ring group 2000 --> (ring group consists of extension 3000 and 4000) --> no answer --> sends to voicemailbox 1000 I want to do this to be able to have one extension number (and mailbox) for a person but allow them to have multiple extensions transparently

Hi, I am having a problem with inbound calls (from a SIP VIOP provider). When caller ID information is not available, the calls go straight to voicemail. We are using a mix of either Sipura 841 phones or SPAs. When the call is passed to the phone/SPA, Asterisk reports "Got SIP Response 406 "Not Acceptable" back from..." I have searched a while now and can't seem to

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

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

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

Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and only in voicemail) after about 20 seconds with the error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt).". I

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000