similar to: Configure Voipjet.com content in Asterisk

Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India.

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've heard good things about VoipJet here, so I was going to set up an account. Then I noticed their Terms of Service here: https://www.voipjet.com/tos.php Several things there are very concerning to me, and I'm interested in what other people here think of them. * The ToS specifically forbids use for any call relating to medical, financial, or government matters -- as well as any

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

Hi, I just setup a VoipJet test account (one with 25c credit) to test, they seem to offer good rates to 02 Uk mobiles :) Anyway, everything went ok, iax.conf amended and extensions.conf too, however when I try to make a call I see:- rt*CLI> -- Executing SetCallerID("SIP/2008-d747", "4153574000") in new stack -- Executing Dial("SIP/2008-d747",

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 List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

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.

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

Hi All, I'm sure this is something simple that I have missed somewhere. When I make a call using BT100 over IAX2 with Voipjet terminating I don't get a ringing sound whilst I'm waiting to be connected. The destination party can answer the call (they do get ringing) and conversation can take place. I don't get this problem with X-Lite softphone? Any help appreciated -

I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers.

The past two days, I've been having issues with my two VoIP service providers where calls just suddenly hang up. The following is from the log: Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host 64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=250000, seqno=80) Mar 14 13:50:55 DEBUG[10428] channel.c: Didn't get a frame from channel: IAX2/voipjet-3 Mar

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.....

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

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

Whats up to VoipJet.com? Their DNS servers are not reachable. Both primary and secondary are on the same subnet - weird setup. Thanks, David

I try to connect to voipjet, but I get always busy, ... it worked yesterday, ... no changes on my side.... -- Executing SetGroup("SIP/615-829b", "iax-voipjet") in new stack -- Executing Dial("SIP/615-829b", "IAX2/17567@voipjet/011886228357765") in new stack May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to create channel

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

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