similar to: Paging MEETME_RECORDINGFILE Variable

I am looking to see if anyone has seen this problem before. I am setting the MEETME_RECORDINGFILE variable in a macro, then using the r option with the Page application to record the page. But the page is only recorded to the file specified in MEETME_RECORDINGFILE sometimes... Sometimes it works and sometimes it doesn't. When it doesn't work it places the recorded file in

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park button while in a call but this does not actually cause the call to be parked. Doing a SIP debug, I don't see that anything is transmitted as a result of pressing the call

Hi i am experiencing this problem: MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE exten => 9999,1,MeetMe(666|1Arxq) exten => 9998,1,MeetMe(666|1Axq) exten => 9997,1,MeetMe(666|1xq) I make a conference between 3 person dialing A dials 9999 B dials 9998 C dials 9997 all works fine but the datetime won't be updated, it still remain for example 13:40 until i do a

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to define CONFNO as part of the file name, which is handy when sorting for users to review. I call meetme

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Exten => h,1,DeadAGI(vol_rest.agi) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andy Swing Sent: Monday, May 03, 2010 3:19 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Run a script after Page application I am trying to run a script before and after the Page application in

Full_Name: Adolfo Jiménez Pérez Version: 1.8.0 (2003-10-08) OS: Window'98 2ed Submission from: (NULL) (81.203.134.156) I've a crash when reading a specific file with the command: > read.table("file.dat") This file can be readed with the 1.1.1 version using the same command and R reads it correctly. The content file is this: ---->file beginning<---------

Am 21.02.22 um 16:24 schrieb Fabian Arrotin: > On 21/02/2022 15:49, Leon Fauster via CentOS wrote: >> Hey all, back from vacation and seeing ansible 2.12 in the repos now. >> Anything to be aware of when upgrading from 2.9 to 2.12 in CS8? >> > > You'd be lucky if it works directly , as there were some semantic > changes in ansible, so you'll probably have to

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hi List, I recently upgraded to Dovecot 1.1.3 on FreeBSD and got into trouble over the quotas. We implemented per-user quotas using plugin { quota = maildir } and protocol lda { mail_plugins = quota } We use a passwd-file userdb with extra fields, which used to be quota=maildir:storage:<bytes> which I changed to quota_rule=*:bytes:<bytes> What

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

On 18/07/2009 8:15 AM, murdoch at stats.uwo.ca wrote: > On 17/07/2009 7:57 PM, Marilyn & Rich Short wrote: >> Hello, >> >> I'm having a problem in R with repeated use of the "load", "save", and >> "close" commands. I'm getting an error message that reads, "Too many >> open files". I'm opening files and

I have a price attribute in my model which I need to validate as a positive number if the attribute price_terms.requires_value is equal to ''Y'', otherwise price can be blank or zero. Price_terms is a select list which my model table belongs_to. How do I do this? - I tried doing this: validates_numericality_of :price, :if => self.price_term.requires_value.upcase ==

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

The Wine development release 1.1.11 is now available. What's new in this release (see below for details): - Numerous fixes for IE7 support. - Support for 64-bit cross-compile using Mingw64. - User interface support for crypto certificates. - Better support for MSI installation patches. - Various Direct3D optimizations. - Various bug fixes. The source is available from the