similar to: Hiding extensions from app_directory

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known

Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission

Hi Shakeel, I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission. I

Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile

Guys, I need you help on this one. Don't know when this broke but we have a custom gui that runs on top of Asterisk running a real-time static for configurations. Nothing has changed with the database other than upgrades of Asterisk 10. Customer complained that there password was not changing when they called into voicemail and changed it. Database is running standard ast_config with the

Hello, Can anyone recommend the 'best' kernel and zaptel versions to use with asterisk? we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can

Is the list going to be the same after sangoma take over digium? On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp <jcolp at digium.com> wrote: > On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote: > > I see a lot of tag lines on posts for the Asterisk Community Forum. Is > > that forum supposed to supersede this mailing list ? > > Both remain available but the community

Hi, I was wondering if anyone has had a problem adding the 'delete' field to the voicemail_users table. I have no problems adding other fields e.g. alter table voicemail_users add column hidefromdir varchar(3) NOT NULL default 'no'; However when I do alter table voicemail_users add column delete varchar(3) NOT NULL default 'no'; I get a message telling me that I

Hi, Been playing with Call parking, and I can`t help but wonder if I am doing something incorrectly. The way I understand it (using default config in features.conf), is I would transfer a call to extension 700, which would park the call, tell me "701". I could then hang up, go fetch the fright person and tell him "call 701 you have a call waiting for you". The way I

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

I have internal mailboxes that I don't want visible to callers going through the directory. Is it possible (in * 1.4) to hide mailboxes fom the directory, without creating a new context? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090911/2e01ee5e/attachment.htm

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hello, I have separate contexts defined in voicemail.conf as follows: [abcdental] 100 => 1234,John Doe And call application directory using the following syntax: exten => 1,1,Directory(abcdental[,abcdental,f]) However since I migrated from 1.4 to 1.6, app_directory no longer parses the voicemail.conf to match the full name of the mailbox. App_directory only matches directory names based

I've checked the wiki, searched the mailing list, Mantis (bug tracker), and looked through the source code, but found not mention of this issue: If using ODBC Storage for voicemail messages, name-playback by app_directory breaks. app_directory has no ODBC Storage code and therefore only looks for the greet.wav file (recording of user's name) in the filesystem, whereas app_voicemail saves

Hi. I'm trying to use the Directory app. I was wondering how can I enter the first 3 letters from the phone keypad... I need a keypad with also the ABCD tones? or a standard one is ok? Doing some tests I never been able to select an entry till now... (using the demo voicemail.conf in asterisk) Thanks for any help. -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

Hi all, sorry if the subject is a bit confusing, but I just couldn?t think of a good way of better describing the situation? Basically, I travel a lot and have several SIM cards for my phone from local carriers. What I?d like to do now is to setup Asterisk, so that people who want to reach me just have to dial one number which forwards the call to all my cellphone numbers in turn. I?m still

Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call

Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 13:59, covici at ccs.covici.com wrote: > > Hi. I am having problems accessing subdirectories on a samba share. I > > am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba > > 4.2.7. I have two shares, one called audio and the other called > > myshare. I cannot access the subdirectories