similar to: Bellster - cool :-)

Steven P. Donegan wrote: > I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the

It looks like DTMF codes are not properly transmitted by bellster. For example, you can try the toll-free number 33800123456, which asks you to press *. When I tried that yesterday, the connection got dropped. Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

>Shoval Tomer wrote: >> As far as I know it's not legal to join bellster in Israel. >> >> It means that you're reselling the minutes you buy from the telco >> company. > >Wouldn't you need to be selling them to be reselling? > >Does that make DISA illegal, and VoIP connections between offices if you >dial out the other end? Well, a thing

As far as I know it's not legal to join bellster in Israel. It means that you're reselling the minutes you buy from the telco company. It also means that you need a permit from the Israeli ministry of communications cause you're acting as an international call provider. Can't be done here. -----Original Message----- From: Geoffrey S. Mendelson [mailto:gsm@mendelson.com] Sent:

None of the unlimited packages you get plan on you using them up. It's like a couple of years ago when the long distance carriers gave us the first phone call of the holiday for free, and people let the phone off hook for the entire week end. It's illegal for sure, and for a couple of reasons. a. you're not allow to resell anything someone else sold you and not pay taxes on it. b.

Hi, > In France, the second most important ADSL provider (named "Free") > offers a phone line (which uses VoIP but can only be used as a FXS) > with unlimited free calls to landlines. I was wondering if I would use my Free phone line with Bellster as well, but I am not sure this is authorized by the ISP : http://adsl.free.fr/hd/cgv.html [in French] En particulier,

> > Great service. Hope it catches on. > > > > In time there will probably have to be some adjustments: > > > > - ratio 10/1 - this means everyone can make 10 calls for every call > donated. > > This doesn't add up in the long run. > > - I get credited 10 calls for every call I make to my own dialplan. This > > should be fixed. > > -

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or on the list. Just curious.

Can Asterisk light the message waiting light on a Grandstream BudgeTone phone? If so please reply with any related configlets :-) Thanks!

Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails in my zaptel build trying to find a Makefile in the /lib/modules/2.6.9/build directory - thanks.

I recently swapped 2 FXO modules on to what had previously been a 4 FXS version of the TDM400 board. The FXS ports are recognized - the FXO ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both say channel 1 isn't there). Has anyone experienced this problem? All software is current as of this AM. If the old FXS modules are re-installed all works just dandy (other than

Hello, I''m Alexandra Well, i have been working with cbq around 3 month ago without any problems, what i do is control the bandwith to 64Kbit for input and output and it works well. But now I don''t want control the Bandwith, i only need prioritize udp trafic over tcp trafic Can somebody to help me with my requirement? Thanks Alexandra

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

This post is a FWDout specific one, but may be of interest for many Asterisk users and may even make more of them use FWDout for the good of everyone :) On the one hand, I have seen many reports of people using the FWDout (http://www.fwdout.net/) service who don't get credits because the prefix they provide calls for to is not popular. On the other hand, I get plenty of unused credits, much

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

The below is a asterisk message when I try to call from a callerid blocked phone into a SIP (Sipura 3000) FXO gateway - and I have not consciously put any restrictions on incoming calls... Dec 29 10:23:44 NOTICE[2745]: chan_sip.c:7486 handle_request: Failed to authenticate user WIRELESS CALLER <sip:A714XXXXXXX@1.0.24.5>;tag=1a6833c3913bcb6o1