[Asterisk-Users] Speakeasy VOIP + Asterisk?
Has anyone tried getting Speakeasy VOIP to work with Asterisk? I just got Speakeasy DSL and am thinking of trying out their VOIP [1] with the hope that the quality/stability will be better than broadvoice. I searched in the usual places (voip-info.org, the asterisk users mailing list archives [2], dslreports.com) and couldn't find anything. Any other places I should look? Thanks, Simon [1] http://speakeasy.net/home/voip/ [2] http://www.google.com/search?q=site:lists.digium.com+voip+speakeasy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lingo and *
On 6/18/04 11:27 AM, Andreas Schiffler [EMAIL PROTECTED] wrote: Hi, just found out about the great lingo.com service offerings. Could this be used with Asterisk? I have a couple of Sipuras on the LAN and would like to use * to route this to Lingo or my POTS adapter. People report that Lingo is using SIP although they say it can only be used with their ATA. They claim PBX compatibility on their website though. Regards Andreas Hi Andreas, I just received my Lingo ATA yesterday and plan on tackling the same question. From what I understand, Lingo uses MGCP, not SIP. Where did you read that they are using SIP? So far, Lingo has been great, I just made a 2 hour call to Germany from the U.S. Call quality was no different from POTS to my ear. I will post a thorough review of Lingo after using it further and after I try to figure out how to get it working with asterisk. Also if you decide to sign up and you feel like giving me a $25 credit to my account (that would be very nice! :D), enter my name and email when you sign up: Simon Dorfman simon (AT) simondorfman.com Thanks, Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service
Hi all, I just saw this article about this new offer from Lingo.com: http://www.techweb.com/wire/story/TWB20040607S0008 $20 monthly plan with unlimited local and long-distance calling in North America (US Canada) and Western Europe. Plus first three months free and free equipment. It doesn't say what hardware they send you. Sounds like a very good deal. I searched the list and voip-wiki and couldn't find any reviews about their service. Has anyone tried them? How is the service? Does it work with *? What codec are they using? Thanks in advance for any answers, Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Primustel a.k.a. Lingo $20/month unlimited service
Thanks Wojciech and Stephan for the info. Maybe it's silly to reply to my own post, but I have since found much more info on the subject. 1. There is a thread about Lingo's service at dslreports here: http://www.dslreports.com/forum/remark,10442660~mode=flat Most info below is gathered from there. 2. It looks like the hardware they send you is: D-Link DVG-1120 VoIP gateway 3. Codecs used: G711, G729, and G723 -Controllable T.38 fax supported 4. Western European = UK, Germany, France, Italy, Ireland, Netherlands, Belgium, Denmark, Switzerland, Sweden, Finland, Austria, Luxembourg, Vatican City, Norway, Portugal, and Spain. Calls to mobile phones are not included. 5. Price: $19.95/month give unlimited calls to U.S.A., Canada, Western Europe (see #4). Their current offer give the first 3 months for free. But there is a $29.95 activation fee and a $9.95 shipping fee. So basically the activation fee plus shipping is about $40. So in effect, you are really only getting one month of free service. If you enter a referrer's name and email address while signing up, both you and the referrer get a $25 credit to your account. I wish I would have known this when I signed up last night. But feel free to use my name and email while signing up so we can both get a $25 credit: Name: Simon Email: simonATsimondorfman.com (replace AT with @) 6. You can try the service for 30 days and get a full refund but you have to pay for shipping to send back their hardware. If you cancel service after 30 days but before 1 year, they will charge a $39.95 cancellation fee. But if you send back the hardware, they will waive the cancellation fee. I'll let y'all know how my experience goes with this service as soon as I get my hardware (3-5 biz days). Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip device discussion and reviews (Snom 190 request)
I'd love to hear a review of any Snom Phones. I'm waiting for the Snom 190 before I buy my first hardware VoIP phone. It's supposed to be around $150 or less. I've already read what voip-info has to say: http://voip-info.org/tiki-index.php?page=Snom%20Phones http://voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom ...as well as searched this lists archives, but I'd still like to hear more feedback about Snom phones. And if anyone has managed to get a Snom 190 already, please do tell where and how much. Thanks, Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] e164.org
So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a non-profit public DNS root designed to map phone numbers to Internet protocols. Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand this. Let's say I configure my asterisk box to use e164 and then I try to call a phone number in Germany. I'm in the U.S.A. So if the number I'm calling in Germany is registered in e164's dns, would my call be routed directly via their voip provider? Or directly to their asterisk box? And would it be free? If that's the case, it sounds kind of cool, but probably won't be much use until lots of people sign up. Any explanation appreciated. Thanks. Simon in New Orleans P.S.- Yes, I did read their FAQ. http://wiki.e164.org/moin.cgi/FrequentlyAskedQuestions ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Number portability
Voicepulse connect doesn't yet offer LNP (local number portability). They said in an email that they will have LNP in 1-3 months. Does anyone know of any voip companies that DO have LNP (for US area code 504)? Thanks, Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Psssst. The US is asleep - let's talk intern ationalization !!!
On 5/14/04 4:19 AM, Robinson Tim-W10277 [EMAIL PROTECTED] wrote: There should probably be en_uk, en_us, en_ca, en_za, en_nz, en_oz, en_ie and en_in etc to allow each English-speaking country to localise prompts. To further complicate things, there really should be a few more categories for all the different kinds of English spoken in the us: en_us_tx (Texas) en_us_mn (Minnesota) en_us_ny (New York) en_us_ca (California) etc... ;P Simon in N'awlins P.S.-who says us Americans sleep?! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: [ISN] Voice Over IP Can Be Vulnerable To Hackers, Too
On 5/14/04 9:02 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Folks seem to have forgotten that the original hackers were hacking stable and secure traditional PBXs with captain crunch whistles! Mitnik ran wild through PBX's and mobille networks. Let's work to set up secure VOIP, but don't let anyone kid you about the golden days when telephones were secure! (for extra points, why's the hacker mag called 2600?) Extra points please: because 2600Hertz is the frequency of the tone required on the old phone system to get free calls. There was a whistle that came in a captain crunch box that happened to produce this exact frequency. Or something like that. I'm too young to know this stuff first hand... I suppose I could look it up... Ah yes, google reveals this: http://whatis.techtarget.com/definition/0,,sid9_gci211496,00.html -- 2600 is the frequency in hertz (cycles per second) that ATT formerly put as a steady signal on any long-distance telephone line that was not currently in use. Prior to widespread use of out-of-band signaling, ATT used in-band signaling, meaning that signals about telephone connections were transmitted on the same line as the voice conversations. Since no signal at all on a line could indicate a pause in a voice conversation, some other way was needed for the phone company to know when a line was free for use. So ATT put a steady 2600 hertz signal on all free lines. Knowing this, certain people developed a way to use a whistle or other device to generate a 2600 hertz tone on a line that was already in use, making it possible to call anywhere in the world on the line without anyone being charged. Cracking the phone system became a hobby for some in the mostly under-20 set who came to be known as phreaks. In the 1960s, a breakfast cereal named Captain Crunch included a free premium: a small whistle that generated a 2600 hertz signal. By dialing a number and then blowing the whistle, you could fool the phone company into thinking the line was not being used while, in fact, you were now free to make a call to any destination in the world. Today, long-distance companies use Signaling System 7, which puts all channel signals on a separate signaling channel, making it more difficult to break into the phone system. -- Simon in New Orleans ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
