similar to: Using queue app with external members/destinations

Hello, I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does anyone have any wonderful or horrible things to say about it? We are thinking about using them because they have many more programmable buttons than the Snom200 phones and are about $70 cheaper. Thanks, MATT---

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

Cross posted from asterisk-biz: > > Is anyone willing to host/manage a website that people > > can simply browse that lists all current DID providers > > and their coverage areas? > It's a good idea and probably not too hard to implement, > it's just a case of deciding how far you want to go.. are > areacodes good enough? or do you need to go to NPA-NXX >

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hi List, Just wondering where the documentation exists for the statistics which makeup the bwplot. I'm guessing that if R is like similar products that the graph is constructed as The median is the filled circle. The box surrounding the filled circle depicts the 25th and 75th quartile. The range of values is given by the dotted lines (?whiskers?) outside of each box, and possible

I don't know if this is already common knowledge, and it's not specificly for Asterisk, but if you are using Cisco phones and want to roll XML applications, make sure you have "Connection: close" in your HTTP header. Without it screen loads are very sluggish. In PHP, do: header("Connection: close"); I whipped up quick-and-dirty PHP/MySQL/Cisco XML directory and

I have a question hoping someone can help me with. I just purchased a 350 from http://lktelecom.zoovy.com/product/HPT350. I have been reading some posts and heard that these phones won't work with Asterisk.. Is this true? I had a previous Nortel Phone working in this enviroment. Well anyway here's my problem. The phone keeps ringing.. It won't stop.. Which makes the phone

Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten => 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of "5551234"? Thanks, Songtao -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Colleagues, I would like to provide a CITATION file for my package nat.nblast [1]. I have the correct citation in BibTeX format [2]. How can I convert this BibTeX to the format needed by R for a package CITATION file (I have a lot of other packages needing citations ...). I think what I need is the opposite of RefManageR::toBiblatex [3]. This seems like it should be a common need, so I

I've seen the question asked a few times already, along the lines of "Who's really using Asterisk out there, and what for?" or "Would Asterisk be right for me?" etc. I've been thinking about it more lately as I may have the opporunity to consult and/or install a system in the near future. I'm a bit apprehensive, as installing a system with a T1 PRI and a couple

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

I have currently following the instructions at- http://www.voip-info.org/wiki/view/QoS+Cisco I am using a Cisco 877 router with an ADSL2+ connection. My upstream bandwidth is around 1Mbps, and my downstream around 4-5Mbps. I have two PSTN side interfaces - atm0 and dialer0 (dialer0 is where the ISP details are configured and the interface that the WAN side IP is applied on) Now which

Hi all Sorry folks, I won''t be at RubyConf this year. I burned up the travel budget with a trip to Thailand, a makeup trip to Florida, and I''m headed to Florida again this Christmas. At least this gives me more time to work on Wuby. :) Regards, Dan

Hi All, I have a 1.8GB PST file that is backed up every night from a cygwin install) and every night rsync transfers about 1.8GB of data for that file. It changes a bit, but for all intents and purposes the entire file is transferred. There are other files on this system that rsync more typically, so I'm fairly certain this isn't a switch or command error on the calling end. I have also

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

Hi, I've had some strange recent behaviour from ssh (OpenSSH_3.9p1 Solaris 5.8). I have an application that runs a script that run a remote command over ssh, however rather than running the command ssh tried to run openwindows (an xsession app, i think). Every time the application ran the command the same issue would occur, until the application was restarted. The command it should have been

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

On Thu, Feb 1, 2018 at 1:29 PM, Duncan Murdoch <murdoch.duncan at gmail.com> wrote: > On 31/01/2018 6:59 AM, Duncan Murdoch wrote: > >> On 30/01/2018 11:39 PM, Hadley Wickham wrote: >> > [ lots deleted ] > >> Personally, I don't find writing in comments any harder than writing >>> in .Rd files, especially now that you can write in markdown and

I setup a central Fedora 11 printer server in a big office for 80+ windows clients due to the max 10 connections limitation of windows 2000/xp. 2 physical printers are connected to the server, and I configured 3 CUPS printers(1 printer with 2 different printer drivers) in Fedora 11. Is there a console tool to 1. Monitor the print queue(s) in real-time like windows system? A history windows hold

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is