similar to: AGI Script for CID Rewrite and CID Name lookup

For those folks who are still using it -- I updated the cid_rewrite script. I noticed that two of the providers were "iffy" and one had changed format a little while ago. It's working again. http://muware.com/asterisk has the latest (1.2.0) Enjoy, -- JM

Hi folks, I think my little agi script is ready for the big one-oh-oh. Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This agi-script does the following: - Standardize incoming caller-id numbers to adhere to US dialing code; NANPA numbers are reformatted to 1+10, international numbers become 011<country-code><number> (this is customizable with a little PHP knowledge).

A few months ago, the friendly folks from liveVOIP went under. We had some discussion on how to limit our losses, and my recommendation was a chargeback, since "FTTP Services" -- their CC merchant -- wasn't affected by the bankruptcy, as far as we could tell. Today, I received this from my CC company: http://muware.com/asterisk/livevoip.pdf Anyone else got lucky?

Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks

Are there affordable DIDs (preferably IAX) available anywhere outside the US? I want to use it to meet ICANN requirements for providing a valid phone number, yet pre-empting some of the telemarketing calls my domain registrations generate. (Yes, I asked a similar question about 900# availability before). I'd prefer to have a number somewhere outside the NANP, preferably an asian country.

I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars -

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

I want to put a single voice-mail box on a remote server, where I have metered bandwidth. Before I do this, I want to make sure it's feasible. Could someone confirm the following math for me? G.711, at 64kpbs has a rated network load of 88kbps. So for each second of conversation, about 11KB are crossing the wires in each direction. That means for a minute of two-way conversation, 1.3MB

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

Whoa, talk about flying under the radar. I got a few DIDs *months* ago from Sixtel (or iax.cc). Initial respone was great, but then it seemed that the only tech-support person had fled the country. No responses, bad responses, poor call-quality. I had a few $s left in the balance and kind-a just forgot about them. Not worth 10 minutes of my time to get $10 of my money back, quite frankly.

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

I tried asking in another thread this week, but I'm not sure people saw the actual subject of the question. Does anyone know where to find documentation of xPL, the home automation interface? Specifically for integrating it with Asterisk. xPL is part of Trixbox, so it's being used, but where is some expertise for using it without Trixbox? -- (C) Matthew Rubenstein

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Hello All, My asterisk installation has now been running for over two months without a hitch, and I've decided it's time to move things around a bit. It's currently installed on a 2.7GHz Celeron under RH9 installed on a 10GB "leftover" drive. Thanks to the strange marketing method called "Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to

Hi: I have a dataframe containing comma seperated group of words such as milk,bread bread,butter beer,diaper beer,diaper milk,bread beer,diaper I want to output the frequency of occurrence of comma separated words for each row and collapse duplicate rows, to make the output as shown in the following dataframe: milk,bread 2 bread,butter 1 beer,diaper 3 milk,bread 2 Thanks for help! deb

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax