search for: chuji

...gt; integration. I'm sure > >you could dig up more. Hell, I'd probaby chip in 500 or so. > Most of it > >is already written too. > > > >There you go all you programmers. Want to make some cash? > Pretty good > >bounty on this one! > > > >-Chuji > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listin...

Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ ;Announcement to play to the Agent answering strategy=ringall ;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them

The Karma Hall Of Fame is now available at: http://bugs.digium.com/karma_halloffame.php Users with negative karma aren't named'n'shamed.. YET.. but congrats to all the users with positive karma on the current list! The list shows all users holding the top 10 karma scores in the system. Right now you need a 12 or above to feature on the list, and there's a boat load of people just

Hi All, I have a question about Fastagi because I can't get it to work for some reason. Everytime I execute the fastagi command, i get an error: my extensions.conf: .. exten => 1000,1,agi(agi://some_ip_address) .. output from asterisk console: -- Executing AGI("Zap/1-1", "agi://some_ip_address") in new stack -- Launched AGI Script

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? __________________________________________________________________ Anton Krall

...t to the log, but it seems like it is much better suited for inside app_queue. Is this by design? Would anyone want this as a feature? It seems like an easy thing to do, I'm just not up to the challenge of doing it. I'm sure that I could find someone who would though. Any ideas? Thanks, Chuji

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

I'm interested in setting up Asterisk as a voice mail system for our Toshiba Strata CTX100 PBX. Our current voice mail system connects to the PBX with four analog extensions. Does anyone have information on how to integrate this kind of system with Asterisk? I've looked at the legacy integration section of the Wiki, but it doesn't seem to have anything about Toshiba systems. ---

Is anyone else experiencing 404 errors on outbound dial with Broadvoice? I've followed the instructions posted by Broadvoice to configure sip.conf, and inbound calling works fine. Every time I try to dial out, I get a 404 "Not Found" error. Here are the relevant sections from my configs. sip.conf: context=broadvoice-in register =>

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Hi all, I have a potential client interested in an Asterisk PBX, which will allow them to improve their virtual office services, among other things (it's a Business Centre). They currently have a Toshiba DK-40 and 19 DKT 2010-SD phones. The owner has told me that the phones are digital (not in the VoIP sense, obviously). Is anyone here familiar with this particular PBX? Any chance that it

Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. Simply using check_udp to port 4569 yields no usable answer and Asterisk complains about receiving a midget packet or something like that :) jens

May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris

*sigh* Ok, I have fought and fought with this. I have read all the FAQ's, I have scanned the list archives. I can receive calls on * from my Broadvoice acct, but I cannot place calls... I have the 'World Unlimited' plan, and like 5 area codes that are local to me in Dallas. Can anyone help me? here are my config files... - sip.conf - [general] context=default ;

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel

Hi asterisk users! Here?s my issue, I?ve deleted the ?s? extension cause I don?t want any action to be taken on incoming calls as my pbx is for home use, but I would like to ring all my VoIP extensions at the same time the PSTN line rings and to be able to pick up the call in any extension, honestly I don?t know if this is possible, some ideas ??? Thanks in advance! -------------- next

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working.