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August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would like to thank everyone involved in Cluecon for all their support! Thanks guys! Brian West Asterlink.com _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists...

Does Anyone have a breakdown on what each option means in manager.conf system,call,log,verbose,command,agent,user I want to make a user who does not get a ton of events in the socket and is just for sending a query and getting that 1 reply I dont want to keep restarting my pbx to figure it out. I'm sure some may be self-explanatory but I was wondering if anyone knows for sure which

I found this information on how to make XP have a dialpad in Windows Messenger which was awesome news HKEY_CURRENT_USER\Software\Microsoft\MessengerService\CorpPC2PHone (change it from 0 to 1 and a magic new choice to make phone calls appears) only to be crushed hours later when I realized It doesnt seem to do dtmf right. If i make an ext lead to AgentLogin for instance and press my

I just made my first 2 modules for asterisk (The 1st one is depriciated already). I was annoyed that i couldn't get GotoIf to take any expressions besides a boolean then i made a module to mimic gotoif and parse a few expressions like (${var} > 12) exten => 1,1,gotoif_expr,${var} > 12:1|4:1|5 Then I immediatly obseleted it with this new embedded perl module that lets you

...u, 02 Oct 2003 17:03:17 +0200 From: "Olle E. Johansson" <oej@edvina.net> Organization: Edvina AB To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] WINXP Messenger SIP Client (Good News, Bad News) Reply-To: asterisk-users@lists.digium.com WipeOut wrote: > Anthony Minessale wrote: > >> I found this information on how to make XP have a dialpad in Windows >> Messenger >> which was awesome news > Some more crushing news is if you upgrade MSN messenger past ver 4.x it > no longer uses SIP.. (so I have been told).. MSN messenger does not use...

Chanspy is now officially in CVS ! Many thanks to Anthony Minessale for the work he's put into this. Julian LS.

Question: is anyone planning on going to the Cluecon convention this year? (www.cluecon.com <http://www.cluecon.com/> ) I'm hoping to go this year and I'm hoping to meet other OSS telephony users and developers. BTW, Anthony Minessale said that there is a need for Asterisk speakers, so if you're an Asterisk user (or expert) and you're in the Chicago area in early August then perhaps you could check out the conference and possibly even be a guest speaker... I'm sure that the attendees would like to hear from develope...

Dear all, now we have the same problem of res_perl compilation with asterisk 1.4. It is the same problem that was present when asterisk was upgraded to version 1.2. I hope Anthony Minessale will be able to solve that problem as he did on that case. But if any of you know a hack to this problem please let us know. Here is the same compile problem again: gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_P...

...e peers. The realtime architecture is not documented very well yet, but there is a lot of testing and development still being done on this architecture. Documentation will follow! *** res_perl: A shining Perl within your Asterisk PBX! ------------------------------------------------------ Anthony Minessale, anthm, is hereby awarded the "Asterisk programmer of the month" award. He has written a large number of additions to Asterisk and continues to add great applications and functions to Asterisk. One of the most unknown is res_perl, code that was added to the asterisk_addons cvs repository...

Ever since late September I am unable to get any newer version of zaptel drivers on a machine I keep at home to play with. If I am not home and the answering machine gets the call the system hangs up on the caller during the greeting but if I'm there to answer it, the call will work fine. If I use an older copy of zaptel it doesnt happen at all so it has to be something introduced

A nice flag in app_dial ? would be f and F to indicate weather to send flash to the fxo or fxs device in a bridged call. so if you have a pots line on an x100p bridged to a tdm400p and the pots line has call waiting you hear the call waiting signal 'f' would behave as normal and flash on the tdm400p and 'F' would send the flash on the x100p The flashing the tdm400p

I was using the Asterisk::Manager perl module and had some troube with it so I decided to make my own. I have a pretty good prototype after a few hours (My main point about the protocol is at the bottom of this example.) http://asterisk.650dialup.com is where you can download it. my $man = init Asterisk::AstMan ({ -user => "asterisk",

I made a post a few days ago with a patch to the queue to allow you to insert a penalty value in dynamic queue adding. http://asterisk.650dialup.com Since then, I decided I would make a whole new strategy called 'skillbased' I added a new option skillmask to the Queue appl and the AddQueueMember func as well as the queues.conf basicly if the skillmask of the queuemember is