search for: oinko

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I was wondering: is there any way to build a version of Bristuff for 1.2 beta 1? Bye for now, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it

Hello list, I put together a quick note about how to "see" oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca.

...ntact the Call Center and he is assisted by an agent (P2), (P2) transfers the call to his supervisor (P3) by pressing "flash" plus "extension # of his supervisor". The comunication between P1 and P3 doesn't get recorded. What can I do to get this recorded? [1] http://www.oinko.net/astrecipes/index.php?from=1&q=astrecipes/recording+queue+transfers+to+disk Maxi

...SS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. -- Home of QueueMetrics - http://queuemetrics.com

...he correct paramater set seems to be: mpg123 -s --rate 44100 --mono /src/mp3/fpm-sunshine.mp3 > fpm-sunshine.raw sox -r 44100 -w -s -c 1 fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav I have also prepared a small tutorial explaining how to setup the whole thing (you can find it at http://www.oinko.net/astrecipes/index.php?n=152 - feel free to add or modify it if you think it's necessary). Thanks l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about i...

Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for

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

Hi All: I am trying to compile asterisk with oh323 but I can't compile it. I am using instruction provided at http://www.oinko.net/astrecipes/index.php?from=1&q=astrecipes/compiling+asterisk+with+oh323. The compile error I am getting is as follows. Quite a few other people are getting exactly same error but no one has posted a fix for this error yet. Any help would greatly be appreciated. gcc -Wall -pipe -Wall -W...

Hello list, I have been researching a bit into the way the queue app works and how different timeouts play together, and have prepared a short tutorial on understanding queue timeouts - see http://www.oinko.net/astrecipes/index.php?n=118 - any suggestion, error found or correction is welcome. While I was at it, I came across a strange bug: imageine you have three callback agents defined, but the terminal for one of them is unavailable. In my example they are 101, 102 and 103, where 103 is unava...

Hi, What do I need to do to put an agent into two queues? The idea being that the agent will get the call no matter which queue it comes into? ~ Matt

...ons - what is nice is that it's pretty easy to implement and you can decide which extensions are monitored through the asterisck CLI without touching the extensions.conf files. http://astrecipes.net/index.php?n=293 Any comment is welcome. And feel free to vote them up on AstPligg http://oinko.net/astpligg :-) l. -- Home of QueueMetrics - http://queuemetrics.com

Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses

Hi I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files, scripts... along with a brief description of the architecture and working of the cluster. It's available here: http://www.bisente.com/blog/2007/08/26/asterisk-cluster-fonebridge2/? lan=english Hope somebody finds it useful. :) Regards -- Vicente Aguilar <bisente at bisente.com> |

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

hello all i am having a problem in compiling openh323. [root@kamran openh323]# ./configure checking for g++... g++ checking for C++ compiler default output... a.out checking whether the C++ compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C++ compiler... yes

...To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <op.txgmiefd3wzjep at pc-lenz> Content-Type: text/plain; format=flowed; delsp=yes; charset=iso-8859-15 Well done! It's top-news on AstPligg right now. http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_I t Thanks l. On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson <jmr.richardson at gmail.com> wrote: > Here you go folks: > > ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf > > If someone would be so kind as to upload t...