similar to: new version of asteriskguru queue statistics released

As promised, we just released the first version of the asteriskguru Queue Statistics. Screenshots and download at: http://www.asteriskguru.com/tools/queue_stats.php --- Small description: The Asteriskguru queue statistics, is a PHP based program, which gives anyone who uses queueing in Asterisk a deep insight in the quality of the service which is delivered to their customers. It is fully

I'm proud to announce the first version (early alpha) of the asteriskguru operator panel, (finally!) Its available for download on : http://www.asteriskguru.com/tools/ No documentation is available yet. (working on that, it will be available later on the same url). Features: ---------- - Support for multiple servers . - Optimized for speed (and big installations with a large number of

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP

browsing the web i don't find any opensource (and free of charge ) software for the web statistic about queues... i've tries queue_stats made from asteriskguru, it is a good tool, and it is free of charge, but it's not open-source :-( i'm considering to develop myself a web application, before that i would ask you if you are interested of this, i would like to activate a

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

What about IAX - SIP or IAX - IAX? ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, January 28, 2006 5:43 AM Subject: Asterisk-Users Digest, Vol 18, Issue 185 > Send Asterisk-Users mailing list submissions to >

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

We just released a new version of the idefisk iax2 softphone, version 1.21 beta, available for download at http://www.asteriskguru.com/tools/idefisk_beta.php Some bugs were fixed, some new bugs might have been introduced :) - The problem with delays is finally gone!!! (one of the bugs was a memory leak, everybody using an older version is encouraged to upgrade.) Privacy Warning: Version 1.21 of

Anybody with photo's (for this astricon or any asterisk related event), please upload them at: http://www.asteriskguru.com/gallery/main.php It's possible to upload as a guest without registering, if somebody sees kiddie porn etc, please warn me so that i can disable this. I will be adding some myself later today. Zoa. P.S. Free beer for everybody who makes pictures with Matt

Hi, Did someone ever managed to make Astertest (http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed all the instructions of this tutorial and corrected the mistakes pointed by the users but it still doesn't work. I can compile it and load app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I have this error : WARNING[31477] : loader.c: 325

How to add SipAddHeader in outgoing call file. I am implementing a Callback scenario, in which a user makes a call to Local Access Number. The system have to callback to the user. During callback a call file is generated. All I want, is to add SipAddHeader("pchargingvector","val") in outgoing Invite. How can I achieve this? Please help me, where can I add SipAddHeader() in

Hi All, Asterisk 1.4.12 on CentOS 5 I'm trying to increment an AstDB key with the length of the last outgoing call. Here's what I've got for "01" UK geographical numbers: exten => _01.,1,Dial(${UKGeographical}/${EXTEN},,g) exten => _01.,n,Log(NOTICE,Call to ${EXTEN} lasted ${DIALEDTIME}) exten => _01.,n,Set(CALLTIME=${DIALEDTIME}) exten =>

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com