similar to: Please help me finding good A-Z provider

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of

So I have my asterisk box up and working internally at home and all is good so far. The next thing I wanted to do was make and recieve calls to regular land lines now. I don't have a POTS line and was looking for probably a SIP trunk. I have seen mentions of Skype integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

I have had good success with my efforts to send faxes over voip using ulaw, surprisingly, and I want to move it from testing to reality. I have an account with Teliax, who have been very good. For voice I use g729 and ulaw, but for faxing I can only allow ulaw. However, Teliax only sets the codec preferences by account. I have another account, but I can't see a way to register two accounts

After loading the RODBC package version 1.1-7, Ctrl-C changes its behaviour and is quitting R and returning to the (unix-)command prompt on the solaris2.8 platform here. Here's what happened before and after loading RODBC > for (i in 1:10^5) rnorm(10) ^C > library(RODBC) > for (i in 1:10^5) rnorm(10) ^C bash-3.00$ platform sparc-sun-solaris2.8 arch

I am shopping around for an IAX provider that provides both outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX. http://teliax.com/ Does anyone have any experience with them? What codecs do they support? Inband or 2833 dtmf? Voice quality and reliability? Any comments and/or other recommendations would be appreciated. Thanks

I've had some serious issues with Teliax as of late with their "new" Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on "1.2 vs 1.4 asterisk compatibility issues". I have had zero problems with their old servers. Voicepulse has been WAY better, but no flat charges, no 729. Frankly, even my broadvoice (yikes!)

I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype

I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple

Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone.... Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

AXvoice server hacked. Here are few working accounts USE XLITE to make calls.... Registrar/Proxy magnum.axvoice.com:9060 Free Sample account.... username=xMaxwellSmartx secret=thanksapache username=woodsy type=friend secret=haramikuttasala username=wumingzi type=friend secret=kickyourass Enjoy! B.R BaBa Jigger -------------- next part -------------- An HTML attachment was scrubbed... URL:

Using AIX 5.3 and gcc-4.2.1 for building R-2.7.1 with the following configure flags OBJECT_MODE=64 CC="gcc -maix64" CXX="g++ -maix64" F77="gfortran -maix64" FC="gfortran -maix64" CFLAGS="-g -O3" CXXFLAGS="-g -O3" FFLAGS="-g -O3" FCFLAGS="-g -O3" LDFLAGS="-L/$HOME/usr/local/lib/ppc64

Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080306/f14d382f/attachment.htm

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from