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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---

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

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

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 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

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

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

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

Folks, I fully understand the emotional need to respond to one who throws around terms like "Communist", "Tyrannical", etc. even if ostensibly framed as a rhetorical question. Certain terms and phrases are by nature pejorative and I;m sure the OP knew this when he entered his post. My feeling was the OP is either an ignorant, unappreciative, self-centered, and emotionally

Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how

I'm running SIP 1.5.2 firmware with a 2.6.2 bootrom on a mix of Polycom 500s and 501s. I'm having a problem with NTP and I'm not sure if it's a configuration issue or a bug in the firmware. I've got the NTP server and GMT offset set in sip.cfg. However, not all phones are in the same timezone. I was under the impression that I could override the offset on a per-phone basis

In the 1.4.x firmware release, there was a config file setting that would cause the screen focus to change to a new call that came in whilst a call was in progress. When set, presentation of a new call would cause that call to be selected, and the softkeys would display Answer, Reject, Forward. The 1.5.2 firmware doesn't seem to do this, despite the setting being set. In all cases, it seems

Hi guys I've got a need to do some call queueing, with the slightly unusual caveat that the destination for the calls is not a phone or group of phones connected to my local asterisk box, but an "external" PSTN number. Can I setup a queue in asterisk and make the queue "member" an external address like SIP/5551234@my.pstn.gateway? There will be a smaller number of PSTN

Dear R-devel, Can anyone help me to understand this? It seems that subscripting the rows of a data.frame without actually changing their order, somehow changes an internal representation of row.names that is revealed by e.g. dput/dump/serialize I have read the docs and inspected the (R) code for data.frame, rownames, row.names and dput without enlightenment. df=data.frame(a=1:10, b=1)

I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? The backgrounder behind my question is

On Tue, 23 Jun 2015 14:23:52 -0400 Mauricio Tavares <raubvogel at gmail.com> wrote: > On Tue, Jun 23, 2015 at 1:54 PM, Marko Vojinovic <vvmarko at gmail.com> > wrote: > > On Tue, 23 Jun 2015 11:15:30 -0500 > > Jason Warr <jason at warr.net> wrote: > >> > >> I'm curious what has made some people hate LVM so much. > > > > (3)

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

Our smallest network has just three systems permanently attached to a 100/1000 router.? There is one additional port available to temporarily plug in a laptop.? The largest systems are both Dells, one running Windows 7 and one running CentOS 6.7.? The Windows 7 system supports three flavors of Linux as virtual machines.? The CentOS system is often used for data file import from USB-attached