similar to: Re: [Asterisk-biz] bellster.net - GREAT advance

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

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,

I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free T?l?com comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO

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.

I have a problem with ZAP interface bridging in France (FXO interface): hangup is detected through a busy tone (no polarity inversion or whatever). When I dial out from a zap line when I receive an incoming call on another zap line (for example to redirect calls to my office when I'm not home), caller hangup is not detected because Asterisk seems to put itself out of the voice path because it

Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.

merlin# rm GENERIC merlin# cd /usr/share/examples/cvsup/ merlin# cvsup stable-supfile Connected to cvsup.uk.FreeBSD.org Updating collection src-all/cvs Checkout src/sys/i386/conf/GENERIC Finished successfully merlin# cd /usr/src merlin# make buildkernel KERNCONF=GENERIC -------------------------------------------------------------- >>> Kernel build for GENERIC started on Sat Jul 26

I have an asterisk servicer (1.0.5) with 3 X100P cards. Everything is working fine but two days ago I implemented call forwarding using the example from voip-info wiki. Now when I enable call forwarding on my phone and a call comes in it gets redirected to my cell and everything is apparently working. The problem is that when we hang up both Zap interfaces (the one where the original

Hi. I am looking for an IPv6 capable CVSUP mirror. I found a discussion from one year ago where it was stated that CVSUP was not IPv6-capable. Does anyone know if this has changed? Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

I try to compile zaptel, without much success. I followed the guidelines in http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation making dependencie results in: asterisk:/usr/src/linux # make dep *** Warning: make dep is unnecessary now. and make tells me make[1]: *** No rule to make target `modules'. Stop. asterisk:/usr/src/zaptel # make clean rm -f torisatool makefw tor2fw.h rm -f

OK, I have done all the stuff at my end and at Bellsters end to add 21 new area codes (all of california) to the Bellster dial plan. Pretty cool deal! I hope others go for this quickly - as it could be a really nice co-op. I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk match to make sure that someone can't run their credits sky-high by making calls through

Dear List, My problem is that the incoming Caller Id is not displayed on the local analog phones (connected to a TDM400 card). I receive the CID correctly from my telco, but when I place the call to the internal analog line, the CID is not propagated. An interesting point: when I try to place a new call to an already bridged line, I see the second call with the CID on the analog phone. The

>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

> > 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. > > -

I am trying to register my Asterisk server to a SIP server which doesn't accept an Expires: field smaller than 1800 seconds and indicates it correctly with a Min-Expires: in an error response when Asterisk tries to use its default of 120 seconds. Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the

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

Hi. Is there a 802.1x implementation (client and server) for FreeBSD -STABLE? Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

Internet telephony (VoIP): Regulators and industry debate 'irreversible' trend The Commission is weighing up its options on how to regulate internet telephony. Major telecoms operators are already proposing services to avoid being squeezed out of the market. More at: http://www.euractiv.com/cgi-bin/cgint.exe/1?204&OIDN=1507828&-tt= Cheers, Philipp