similar to: experiences with teliax, voipjet or junction networks?

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

One would assume they have better things to do as they are quite busy. I think this is just a proactive measure meaning they say you cannot do it upfront so that in the event of a problem, it was predeclared. As to the rest of the TOS, I could be wrong but I got the impression that the owner of VoipJet speaks English as a second language due to some email exchanges. If that is the case, the TOS

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

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

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

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

No Dice so far, anyone now how to add anIAX trunk? What are the settings exactly? I have added everything but I do not know what are the registration strings? Jonathan

So far my experience with TOS has been that most of them are pretty odd. No one wants the liability of a stock trade gone foul or a call to the doctor that gets disconnected. Essentially, those things in the TOS are just a CYA. They are un-enforced but should someone decide to attempt to sue based upon a financial loss, the ITSP is covered. So, yep. That is weird but not unexpected. Heaven

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

I am running 3.4.3 on a windows server and I ran the code in a new session. I get a warning when running packrat::init() on a project that is located on a windows network drive. The warning I get is Warning message: cannot set reparse point 'U:/packrat5/packrat/lib-R/base', reason 'Access is denied' The error is created based inside the function .Internal(mkjunction(fr,

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

I''m looking into how to implement a relationship between some models, and I think I need to use a junction table. From the documentation I''ve read, a junction table doesn''t require a model. But from my controller, I''m not sure how to get the data I need. Here''s a description of my models and what I need in my controller: Models 1. Event 2. Site 3.

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is

Hi, Has anyone experienced intermittent echo issues with voipjet lately?

Hi, Does anyone know what is going on with voipjet? This morning/afternoon they just seem to have gone down.... no word on their website.....