similar to: Monitoring which users are online in realtime

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]:

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

hello all, My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 regards rene -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anybody know the sql type for the "call-limit" field under sip peers? Everything on voip-info is missing that entry.

Since 1.4.22 realtime status for sip peers seems to be broken. If I do a "sip show peers" from the CLI I get this: 2001/2001 192.168.2.234 D 5060 UNKNOWN Cached RT It is arbitrary which peers will say OK and which will say UNKNOWN and it changes over time. This is a problem with an application like the Asternic Flash panel because it uses the peer

Hello I need to add sip extensions from my UI so without going through sip.conf so i created table CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `username` varchar(40) default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `ipaddr` varchar(20) NOT NULL default '', `port`

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hi All; Which is better (to have more stable or release versions) of zaptel, libpri and asterisk: to use cvs or svn? In case of using cvs, why I need to type: export CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot In other words: what is the use of pserver, anoncvs, ... with cvs checkout? Note: How can I know all the variables needed for cvs checkout so I might need to do

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro, and passes arguments like this as well, you lose the original values. 2. When the macro's 'return' some value, it has to set a channel

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk?

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen