similar to: stale voicemail messages / greeting

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

I'm going to be testing the new realtime stuff further in the next few days, and just wanted some clarification on a couple of things before I start on it. I believe I can store any config file in a external config such as mgcp.conf for example, by adding it to extconfig.conf with the below syntax. mgcp.conf => mysql,asterisk,mgcpchans Doing this will require a reload of asterisk to read

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

Hi all, I'm sure this is a stupid question, but is there a way to check your voicemail by calling your extension from the outside? When I call my own extension from outside and hit pound or star, it just stops my greeting and gives me the "beep". I'd like to call my extension and press a key and be prompted for my password. Otherwise the only way I can think to get around

Hi there, I am configuring a small voicemail server and I am facing the following problem. Executing this command: exten => 1234,1,VoiceMail(${NUMBER}@test) When a user does not have a customized temporary greeting vm-intro message is played asking for the message to the user but when the user has already a temporary greeting both the temporary greeting and vm-intro are played. Basically

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

I'm using Asterisk 1.4.13, the latest released version. The linux platform is FC7. I setup my Asterisk server to use IMAP storage. Dovecot is the IMAP server. Its storing messages perfectly--no problems. I should also mention that I'm using MySQL for real-time configuration. That must be working (at least partially), because I can authenticate v. the voicemail table. However, the

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello, I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and I can't delete message from VoiceMailMain using option 7 Default folder is /var/spool/asterisk/voicemail and it's owned by asterisk:asterisk with 777 permissions Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post :

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

Hi, I'm localizing the voicemail messages to Portuguese. To make it possible for another person to translate it, I've set up a couple of extensions that call the following macro for each message on the system. After recording, I can perfectly hear each message using Playback. When I try to play the new recorded message using VoiceMailMain, I can't hear the new message (line goes

I have voicemail configured to store messaging in an odbc database. Does anyone have any thoughts on how best to play back someones greeting from the db? Thanks, ~jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080212/09c64a88/attachment.htm

Hi anybody, I did setup fax/voice switch according http://www.voip-info.org/tiki-index.php?page=Asterisk%20fax somthing like: [macro-faxvoiceswitch] exten => s,1,Answer exten => s,2,Macro(oneman) exten => s,3,Hangup exten => fax,1,Macro(faxmail) where: [macro-oneman] exten => s,1,SetVar(klapka=${MACRO_EXTEN:-2:2}) exten => s,2,SetVar(voicemail=${PFX_ISDN1}${klapka}) exten

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel]

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?

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf