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...nvent the wheel totally from scratch, is there anyone out there that has data in electronic form that they use already for this sort of thing? I'm looking for country code listings, area code listings, NPA-NXX to city name listings etc. Replies to the list, or forward data files to web-dids at ivrl.com Cheers Paul

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

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

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

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 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 --

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

I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #.

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

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