similar to: Changing Asterisk install location...

Has anyone tried to (recently) install asterisk in a location not relative to /, as a non-root user? Ie editting the PREFIX directive in Makefile. Why? Several quite obvious reasons: a). Allows an asterisk user to be created, and operators to log into the box as asterisk user, without having root access. b). Much easier backups, because everything is beneath the same directory structure.

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hi, I'm using mysql to store my cdr data. I compiled asterisk-addon module without problems and I see nothing unusual in my cdr_mysql.conf but when I do a reload I get this messages (never seen before): Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open Asterisk database Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database unavailable But If I try to connect from

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file? It seems to only read it on startup. Thanks.

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

Hello, I m using asterisk version 1.6.2.0 beta. I m trying to test load on it, for which i m using WINSIP installed at two computers and facing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below.

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hello, I'm using filehash on the windows XP and it has been working fine with the newest R version 2.6.2. However, on the windows vista, when I ran the same code, I got the following error: > dbCreate("simdb") #create simdb database [1] TRUE > db<-dbInit("simdb") #initiate an object of database Error in sprintf(gettext(fmt, domain = domain), ...) : object

Hi, Is the following behaviour a bug or a feature ? Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces : [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:457

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving - although I could imagine it might create problems in a race condition case. p From:

I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if

Egads. Getting queues to work is like pulling teeth. extensions.conf: exten => q_main,1,Queue(oneeighty_main||||1) exten => 80014055,1,Dial(SIP/80014018,15,tr) exten => 80014057,1,Dial(SIP/80014018,15,tr) exten => 80014052,1,Dial(SIP/80014018,15,tr) queues.conf: [oneeighty_main] musiconhold = default joinempty = strict leavewhenempty = strict strategy = rrmemory retry = 0 member

I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.