similar to: Destroying a SIP extension doesn't destroy voicemail box?is this a bug?

Whey you 'destroy' a Sip extension you are only removing the entrys that allow you to make and receive the auth needed to do so. Your voicemail files are not tied to an extension but are independent and are only 'married' when you specify it in your sip.conf or other channel configs. Removing a confi from a channel will not touch voicemail. You need to go into the voicemail.conf

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi guys, I'm trying to reuse Mitel 5215 from proprietary system now into Asterisk :) ! I've them already with SIP and handling calls sucessfully! I've followed instructions from: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Mitel+5220 Additional Servers * Outbound Server: Off * Outbound Server URL: blank * Outbound Server Port: blank * Voice Mail

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

Should be mailbox = 7752365815@vpbx-wpti Best Regards, Ben Bawkon --------- Original Message --------- From: Bruce Komito To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] voicemail notify problem on sip extension Sent: 5/19/2004 4:27:51 PM I'm having a problem with the voicemail notify feature. Although I have the voicemail box configured for the sip extension, the

Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to

Hi, My case is a little bit complicated. I would like to use my Asterisk Box for 2 different services/providers : - Voicemail server for one - SIP Registrar and Proxy for some other extensions The problem is that Voicemail service is for another provider which has defined Extension like ABC ... We are connected to them through a SIP Trunk. Everything works fine .... Except IF ABC is also defined

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

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