similar to: Logging Asterisk console

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]:

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -

Hello, I am trying to get a repeatable build setup for asterisk. Part of doing so involves using the -C option to specify the master config file. The problem is that asterisk reads the config file location that i specify, however it still tries to read two other config files, namely: * /etc/asterisk/extconfig.conf, and * /etc/asterisk/logger.conf I have specified in my config file that the

Hello On a uClinux-based appliance, "ps aux" shows multiple Asterisk processes: 380 root 11990 S asterisk -f 381 root 11990 S asterisk -f 383 root 11990 S asterisk -f 384 root 11990 S asterisk -f 385 root 11990 S asterisk -f 386 root 11990 S asterisk -f 387 root 11990 S asterisk -f 388 root 11990 S asterisk -f

The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer in sip.conf and enable TCP However, I'm not sure that they should be installed by default by packages. Most package managers provide a way to diff

hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection.... ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

machine O is outside firewall, machine I is inside (machine names changed to protect the innocent :-) firewall allows ssh connections if inititiated from I to O, but not if the other way. both machines have an /etc/rsyncd.conf of: [rt] path = /tmp/rsync_test comment = Test area O runs rsync daemon, I initiates a rsync cammnad like rsync -rvvv --delete --rsh=ssh O::rt /tmp/rsync_test

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing [301 at from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack -- Called SIP/301 -- SIP/301-00000046 is ringing

On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you do not have to (re-) make your whole Asterisk. The Makefile generates: codecs/codec_opus_open_source.so formats/format_ogg_opus_open_source.so formats/format_vp8.so res/res_format_attr_opus.so Without any of

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body bgcolor="#ffffff" text="#000000"> <font face="sans-serif">Hello all,<br> <br> I've googled

Hi all, I have to make for a client an asterisk system for process up to 250 calls between conference and normal call. At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client require a failover system. Anyone have experience for this type of solution? Is better ultramonkey, dundi or SER proxy in front of * server? Thanks Enrico P.S. Now during all this year I have to work

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second

We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear,

Hi, I am using asterisk 11.1.0. How to display the caller number (from asterisk -rvvv terminal) in the first step of the extension (before doing any action) ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130417/8e6cbb1a/attachment.htm>

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the