Starting Aug 1, $35/yr for a Premium account with unlimited VSPs and dialplans, or $99/yr for a Professional account which also includes business hours email support. Otherwise your account will revert to the Free account with one VSP and one dialplan. Blog: http://sipsorcery.wordpress.com/2011/06/25/premium-accounts-now-available-for-purchase/ Plans and Pricing: http://www.sipsorcery.com/mainsite/Home/Pricing

has anyone been using line2 for iphone? it seems like a great dea and so far working fine for me. I would love hear others thoughts

I installed CentOS and Asterisk 1.8.4.2 on a laptop and was able to get "CLI" prompt, I have read the Asterisk E-book edition 2. When I tried some commands, such as "sip show peer" I got "command not found" errors. I searched Asterisk forum and one user posted he had to use "module load chan_sip" to get "sip" commands, I could not find that information from the ebook. After I started Asterisk by "asterisk -r", and tried to register Xlite from another PC using IP address of the Asterisk laptop, I did not see anything in Asterisk command window and could not have Xilte registered. I did create simple "extensions.conf" and "sip.conf" files according to the E-book. I can log in the Asterisk laptop using ssh and ping it. Did I miss something? Do I need to setup DHCP and other servers on the Asterisk laptop? Thanks for any help.

I am having problems with choice one communications DID's started after 5pm est. Anyone else having problems?

Hi Guys, I am looking to set up VOIP calling via my google voice and sipgate account. I am in the process of receiving my WRT54GL and Linksys PAP2T-NA. Anyone have any guides as to what procedure I need to follow so that I can make calls through my ATA. I understand I cannot directly connect my ATA for a Google voice outgoing call. I've read many a places about setting this up but I ended up pulling my hair at the end and am not certain I got anything out of it. Here I am trying to start afresh. So bring it on guys. THanks. -- Chirag's Website

+800-1234567 Is there a way to call the above toll free number in Germany? I use Rynga.com (a betamax service) for all my outgoing calls.

I have tried 3 different popular softphones (LinPhone, Ekiga and Twinkle) on my laptop with Ubuntu. It seems these softphones behave very strange. In my house (Comcast cable), Twinkle works fine with my asterisk server, Sipsorcery and other sip providers; Ekiga can not register to my asterisk server, but can register to Sipsorcery; LinPhone can register, but sometimes it has one-way audio problem. In my friend's house (ATT DSL), Twinkle can not register to any sip servers (it keep saying "your network is behind symmetric NAT"), LinPhone can not register to my asterisk server, while Ekiga works fine. There is no problem with softphones in Windows version, such as x-lite, PhonerLite etc. It looks like the registration setting for those softphones in Linux are very different from those in Windows system. Anyone know why these linux softphones behave so differently and is there any good one for recommendation? Thanks.

Hi, I'm traveling abroad and I am looking for an EASY service that will forward my calls to an international cell (from the USA) and that when I pick up I need to press a number to accept.. (In case I miss a call I don't want them to get voice mail or something else that will be in a foreign language..) An added bonus would be to allow me to make calls from my cell (perhaps setting them up online) so that the people I call won't see I'm overseas.. many thanks

Hello, I have a cordless phone attached to an SPA2102 ATA using freephoneline.ca. The phone lights up for an incoming phone call but does not ring. The phones ringer is on and the volume is turned up. Can you please help me as this is very annoying to keep missing calls I should be able to hear. Thanks

I'm trying to install a new phone jack in my room so I can hook up my new voip service from voip.ms. The phone jack outlet I got from Home Depot has red, green, black, and yellow terminals. I have two wires in my wall and the wires inside it consist of six colors. I checked my other phone jack and it seems like the two blue are connected to one terminal and the two white/blue is connected to the other terminal. So I connected the two blue wires to the red terminal and the two white/blue to the green terminal. There's a dial tone and I can call out but for some odd reason when I try to call in the phone doesn't ring but I can still pick it up and answer sometimes. Often times I call the home I get a busy signal. I'm not sure whats going on here. I've tried connecting two blue wires to the green and the two white/blue to the red but it still didn't solve the problem. I've also tried connecting the orange to the black and the white/orange to the yellow and still no go. I'm pretty sure I got the wiring wrong because when I connect my phone directly to the ATA device it works perfectly. Can someone please help me with this.

I'm looking at porting over my existing number to freephoneline's service (or voip.ms as a fallback). I will be purchasing the Cisco PAP2T-NA ATA adapter to use with my regular analog cordless phone system (with multiple handsets that link to it). I can't seem to find a listing though of what calling features will work with either service. The 3 that I need the most are caller ID, voicemail and call waiting. 3 way calling would be nice but isn't a deal breaker like the other 3. I called FPL and one person said yes call waiting works and called a couple days later and the other person said it didn't... So for folks with this service or who happen to know, do those 3 features work for you once you have it setup properly with a ATA and regular analog phone setup? Thanks in advance for any assistance!

I just received this email from Voxbeam. Please note these are the rates for the Premium service, not the cheaper Standard service. Direct CLI route to Canada $0.0025 As part of our latest premium price file issued for today, Voxbeam is pleased to announce a significant rate reduction to the rate of $0.0025 for all Canadian Direct CLI Destinations listed below. We are also pleased to announce that we have partnered with our direct vendor in Canada to provide increased capacity to help maintain more reliable and stable routes for our valued customers. 1289 CA Canada 289 Ontario $0.0025 1403 CA Canada 403 Alberta $0.0025 1416 CA Canada 416 Metro Toronto $0.0025 1418 CA Canada 418 Quebec City $0.0025 1438 CA Canada 438 Quebec $0.0025 1514 CA Canada 514 Quebec $0.0025 1519 CA Canada 519 Ontario $0.0025 1604 CA Canada 604 British Columbia $0.0025 1613 CA Canada 613 Ontario $0.0025 1647 CA Canada 647 Ontario $0.0025 1705 CA Canada 705 Ontario $0.0025 1778 CA Canada 778 British Columbia $0.0025 1780 CA Canada 780 Alberta $0.0025 1819 CA Canada 819 Quebec $0.0025 1905 CA Canada 905 Ontario $0.0025 -- iNum +883 5100 XXXX XXXX - Anveo - VoIP.ms - Callcentric - Rebtel - Localphone - PBXes - MAXroam - Linksys IP Phone SPA962 - Gigaset S685IP - Ubuntu Linux 11.04

If you consider going to the EU state of Denmark and wish to use WiFi for your voip calls you are soon in for a challenge. Since a couple of years the government monitors and registers everything people are doing on their internet connections. Data about every click on the web, every email sent and received are stored in huge data bases. In addition to the internet traffic detailed information about every phone call (landline, cell and voip) are also registered by the government. However, the government has a problem. There are open WiFi at places like McDonalds, Starbucks, pubs, coffee shops and similar. Currently the government does not know the exact identity of the users of these often free WiFi and that is a nightmare for the government. To solve this problem the government is preparing a new law making it illegal to use WiFi unless the government knows the exact identity of the user. The new law will require anyone wishing to use an open WiFi to register with the government using a personal ID. When the government has verified your ID you will be issued with a unique password which can be used with any open WiFi connection. The new law enables the government to monitor your use of any open WiFi connection in great details. Denmark is also known as the George Orwell state. :huh: -- iNum +883 5100 XXXX XXXX - Anveo - VoIP.ms - Callcentric - Rebtel - Localphone - PBXes - MAXroam - Linksys IP Phone SPA962 - Gigaset S685IP - Ubuntu Linux 11.04

Hello fellow bloggers, I once wrote replies to some people looking for DIDs in New Brunswick, unfortunately the post is now too old to reply to so I thought I'd start a new thread. It has come to my attention that Fredericton, Moncton and Saint-John is now available from a couple of VoIP resellers. VoiceMeUp.com now has Moncton in stock and is currently taking orders for Fredericton and Saint John. I would love to hear where else it is available. Cheers

Hello all, I just got an Anveo account, and plan to set it up with my data-only plan on a HTC Desire Z / G2. I got a DID with SMS. So far so good getting voice to work - it works fine. However, I am quite confused about how to make SMS work. Anveo appears to support SIP MESSAGE and Invite+headers (I think I want the former). As well, CSipSimple appears to support SIP messaging of some kind, though it's not clear to me which. Sending and receiving SMS appears to fail via CSipSimple and through the Nokia N900's "conversations", both of which should work. For sending, the error in CSipSimple is "Method not supported". For receiving, I'm not sure what the error is. Simply no message arrives when I try to forward via SIP MESSAGE, either to a blank URI or to a URI of myaccount@sip.anveo.com or myaccount@sip.anveo.com:5010 or myaccount@anveo.com or myaccount@anveo.com:5010, or of course all of the latter preceded by "sip:". Any ideas? I am also messaging the Anveo support people. If I get a response there, I will post it here. Tarek

I saw one of these systems in a thrift shop. I can't say it is the exact same model, but the picture on the box looks just like this: http://www.aitelephone.com/packet8-virtual-office.html . Are either the phone or the VOIP adapter reprovisionable for another application, or do they have to used together and only with the intended service? If it makes any difference, I don't know if the system was ever activated with the company.

Anyone else having issues with FPL since apprx 11:30 AM today? I cannot make any outbound calls. Incoming is busy. Also my PAP2 says its registered but it shows OFFLINE when I login to their website

Is there a way to disable call waiting?

I tried a google search and had no luck so far. I am using a PAP2T and was wondering, if I ever get the budget, is there an ATA that supports more than 2 ports/lines that is affordable? Also, is there any way to use a logical/software/whachamacallit, rather than hard wired, control for what port is used? One more question: I remember it being discussed that IPV4 ATAs were soon to become obsolete, unusable or something to that effect. How imminent is it that we will need to buy new ATAs, and are they being made yet? -- Join Teams Helix and Discovery. Rest in Peace, Leonard David Smith, my best friend, you are missed badly! Rest in peace, Pop, glad our last years were good. Please pray for Colin, he has ependymoma, a brain cancer, donate to a children's Hospital.

Excerpt: If Google and allies succeed in establishing the technology and building support into multiple browsers, that would mean anybody building a Web site or Web application could draw upon the communications technology. In other words, anyone could build a rival to services such as Skype, with just a Web application. MORE: http://news.cnet.com/8301-30685_3-20072845-264/google-building-skype-alike-software-into-chrome/