similar to: Re: Asterisk-Users Digest, Vol 18, Issue 158

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. c).

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 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

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, 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'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

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, 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.

Hi, I have been using filehash for a while. It has performed very well. However, recently I found filehash gives an error when I need to do something like db[c("a", "b")] when the db is in RDS format. Does any one know a way to get around that? The code below reproduces the error thanks Jeff filehashOption(defaultType = "DB1") dbCreate("mydb3", type =

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and

unix-odbc with Asterisk Realtime is one good way to use a different backend DB than MySQL. I haven't heard of "bit rot" problems running it over long times, but I'd like to if there are any. I'm particularly interested in seeing reports of Asterisk Realtime backed by Postgres. The problem with pointing dialplan DB functions like Set(DB) at unix-odbc (or any relational

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

All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx "database show" > output.txt and work with that and then set a new value such as asterisk -rx "database put $key $value". The whole process can take over 1 second for EACH ENTRY which adds up for more than a

Hi, After only 4 days I have 107472352 bytes in 46007 allocations in file 'pbx_spool.c' asterisk*CLI> show memory summary 180 bytes in 2 allocations in file 'netsock.c' 12 bytes in 1 allocations in file 'devicestate.c' 2268 bytes in 1 allocations in file 'jitterbuf.c' 8160 bytes in 1 allocations in file

Hi, On Lenny, I typed "apt-get install db4.5-util " then (as root) : # db4.5_dump /var/lib/asterisk/astdb db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument # file /var/lib/asterisk/astdb /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order) Is db4.5_dump appropriate to dump an