similar to: RT voicemail greetings not played

Hi all. I've got an odd situation with my RT asterisk server. I've got a number of users who are reporting that their voicemail greeting isn't being played anymore. This used to work before a recent asterisk restart. The dialplan is in AGI, so it wasn't changed. I'm storing voicemail in a mysql database and that is working properly. It's just the greeting message

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

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

Hi all, I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints

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 everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I'm have a bit of a problem with temporary greetings. I'm using 1.6.0.1 with IMAP storage. If I go into comedian mail and record a temporary greeting, I get a Message [0] on my IMAP server. In addition, I get /var/spool/asterisk/voicemail/default/134/temp.[gsm|wav|WAV] files. I have imapgreetings=no in the voicemail configuration file, as

Hello everyone, Is it possible to ask the VoiceMail command to play the user greeting before recording a new voice message? I've looked at the documentation and I don't see how to do that... Any help would be really appreciated! Best, Leo

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

Hi all, I had a scary thing happen to me when I did my database backup last night. Within seconds of starting the backup script, ALL of my real time sip peers disappeared! All of my voicemailboxes, voicemail messages, and sip peers are in a Mysql database. I was able to do a database backup 3 days ago without any incedent. Now suddenly, the whole thing comes crashing down. I also notice that

I've read several of the Wiki sections on Voicemail and "Asterisk variables" but could not find an obvious answer to this question. I would like to run a script that post-processes the voicemail after the Voicemail application returns (with AGI or DeadAGI), but I cannot figure out how to easily determine the name of the file written by the Voicemail application. Does anyone

Hi all, I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've discovered that my server crashes as soon as I leave a voicemail message. I'm using odbc voicemail storage as well as mysql dynamic configuration. I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 I suspect that the odbc drivers are the problem. Is ther an alternative drive that I should be using?

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi->exec("background $msg");

I'm using Asterisk to read voicemail users out of a SQL database. I am assigning users real phone numbers as their voicemail box. The problem is that if I re-assign a phone number (say, 972-245-0001), the new user is stuck with the old user's greeting and saved messages. What is the best way to resolve this? I don't want to use unique mailbox ids because my dialplan looks like this

Hi all, I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. I've tried to unload various modules; nothing resolved the issue. Any suggestions? -- Mike Diehl

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the