similar to: [Fwd: Re: [Asterisk-biz]bellster.net- GREATadvance]

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:

>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

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

Hi, I hate to have my first post be a bug, but this drove me crazy for a while. I compiled the latest version from source and followed the instalation instructions. I then copied the xinetd.d entries from the Wiki and proceded to test. No matter what I did I could not log on. I found that while I was editing /usr/local/etc/dovecot.conf, nothing was hapening. I then started it manualy and found

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,

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

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

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

> > 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. > > -

Areski, thank for spending the time to write it. Thanks for taking the time to share it with us. But, many special thanks for having to sit this MySQL vs. Postgress bickering WAR. GUYS, this is an Asterisk Users Mailing List. PLEASE, take it elsewhere. I really do want to help newbies. But about 200 messages a day is hard to read through. When you add another 50 to the bunch for no reason it

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I'm having a brain fart.... What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten => _6X.,1,Dial(SIP/3091@205.22.93.1/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

No, its up to them which backend they want to use. Sounds like they think that GCC is super quick compared to LLVM. Looks like another fud fart out of google to me. 2009/11/12 Jon McLachlan <mclachlan at apple.com>: > Any plans to make LLVM work with Google's new language, Go? > >

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ok... I am having a serious brain fart this evening. IIRC, the next sip draft addresses shared lines and I thought I remembered something on the list about support for it in the near future. I also thought that chan_sccp supported it with in asterisk. Am I loosing it here? Is there an implementation for shared line support in asterisk? I know that

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

On Nov 12, 2009, at 10:25 AM, Edward O'Callaghan wrote: > No, its up to them which backend they want to use. > Sounds like they think that GCC is super quick compared to LLVM. Looks > like another fud fart out of google to me. Edward, this is no place for comments like this. Evan > > 2009/11/12 Jon McLachlan <mclachlan at apple.com>: >> Any plans to make LLVM

Erik de Castro Lopo <mle+la at mega-nerd.com> wrote: >> -msse for SSE code, -msse2 for SSE2 code, -msse4.1 for SSE4.1 code > > Yes, that was it. Brain fart. These flags were not needed on x86_64. > > Erik But now all C code is compiled with -msse2 and it won't work on older CPUs. Isn't it better to compile only necessary files with this flag?

Yea, I figured it out...brain fart on my part. Got it working, but getting some sort of TCPSocket error now... still trying. :) [root@jabez etc]# darkice -c /usr/local/etc/darkice.cfg DarkIce 0.6 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: /usr/local/etc/darkice.cfg Using POSIX real-time scheduling, priority

I have an array with dimension of 5001 (rows) by 40 (columns). This array is a series of power output from a wind turbine in kW. The rows correspond to wind speed from 0 to 50 m/s in 0.01 m/s step increments. The columns correspond to the air density values from 0.90 to 1.30 kg/m3 in 0.01 kg/m3 step increments. I have a vector of wind speed and air density observations (typically 8760 records