similar to: Voicemail2

Does anybody have any reason why I should *not* permenantly replace app_voicemail with app_voicemail2? If so, speak now or forever cvs update -D "8/18/2003". Mark

Hi, compiled 1.0 on macosx latest (10.3.5). compiled fine. when running, complains about voicemail2 module. Any hints? Marc. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

>> Wouldn't that break everybody's dialplans where they would have to >> replace all occurrences of Voicemail2 with Voicemail and all >> occurrences of Voicemailmain2 with Voicemailmain? > > No, we would register with both names. Is it necessary (with reasonably current cvs) to make any changes in the *.conf files to use Voicemail2, or is that happening

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing

I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press "#" and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin

Hi all - I'm making gradual progress implementing asterisk on my box! Now, when I type asterisk it dies at this point. Does anyone have any idea why this is happening! It have checked everything but running out of options! [app_voicemail2.so] => (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2'

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

I would like to call my own DID number from outside, get into voicemail, and then push '#' to exit into VoicemailMain. Is there a way to do this?

Hi, I have some troubles with voicemail sending message to my email address. Do I have to configure sendmail to use Voicemail ? What's about sendmail configuration ? In voicemail. i have lines like this : [default] 1000 => 1234, xxxx yyyy,beasse.lsc@wanadoo.fr I don't see what's password 1234 is ? (password for what ?) But i never received any mail with wav file attached in

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hello, I recently started playing with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives. 1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this? 2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

I'm having tons of trouble getting the fromstring to work in voicemail.conf. I've tried both voicemail and voicemail2 but the emails still seem to be coming from asterisk pbx. Has anyone had any luck with this? ================= Here's my voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav49|gsm|wav ;

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

in my setup i have several users with DID lines coming in from various sip/iax providers. within our old phone system, a user could call their own DID line, then hit the * key when they hear their voicemail greeting and be prompted for their password. is there any way this could be replicated within asterisk? i'm having trouble figuring it out since it steps through things sequentially,

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user