similar to: List of VoIP provider codes

Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Hope someone can help :) I am testing 4 PSTN termination providers. 3 SIP and 1 IAX IAX and 1 of the SIP providers work fine. Now the wierdness: 2 SIP providers I can only get oubound calls to ring at the destination and then nothing more. 1 gets as far as SIP code 183 (and ringing on the src handset ...yay) the other doesn't get past 100. Added to this inbound calls

Hi List, I want to set my caller ID and name with asterisk. So that when I make outgoing calls then destination end will see my name with number. from asterisk end I set both the things into dialplan. --------------- -------------- exten => _X.,n,Set(CALLERID(num)=9172341457) exten => _X.,n,Set(CALLERID(name)="Virendra Bhati") But when call reach to destination number then only

I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will

From the bend me over news department. 2 March 2006 Mr. Leonard Katz Executive Director Broadcasting and Telecommunications Canadian Radio-television and Telecommunications Commission Ottawa, Ontario K1A 0N2 Dear Mr. Katz: Associated with Bell Canada Tariff Notice No. 6929 1. Attached for the Commission's approval are proposed revisions to Bell Canada's Access Services Tariff Item

Anyone out there using VOIP for business class inbound/outbound services? I've found my VOIP provider to be less than reliable, SIP registrations timeout, calls drop, they claim IAX2 is too buggy (I find that hard to believe), and pretty much blame all problems on other circumstances and don't actually fix their problems. Their auto rollover doesn't even work right, so if my PBX is

Hi please excuse if this seems obvious, but I am new to this and the SIP section in the Asterisk handbook do not give any clues nor do the SIP examples in there seem to represent real-world situations. I am using Nikotel as a VoIP provider (for now) and I would like to configure Asterisk to sign on with Nikotel so that I can use the telephones connected to Asterisk to make calls using the

Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month.

I'm looking for a way to detect when a VOIP provider is unable to complete a call and thus try another VOIP provider (failover/backup type situation). using qualify is NOT sufficient, since the provider could very well be reachable but not be able to complete the call for other reasons. A perfect example: setting my caller ID number to my real number and calling a local number causes the

I have been looking around for VoIP providers but have not found a good listing. Is there no "yellow pages" for VoIP providers? Google mostly returns services like Vonage, Packet8, NuFone, ect. None seam to be very reseller friendly and none offer LNP or local DID's for my area. Anyone know of a list (even a partial one) Jeromie Reeves

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

Just setting up Asterisk. I'd like to be able to dial out through VOIP providers and have customers type in a code in response to a prompt. So far, I've been able to set things up to make the call and play the prompt. However, my problem now is the DTMF tones; they don't register when I call get_data. When I make a call in person (from the DIAX softphone, through a VOIP provider,

Hello we are a small call center with only 8 lines we use max4 and the 2-2 port gateways from net2phone . There equipment is good but we are getting hit by lower cost competition. We need to be able to compete. We have a couple of providers who are 50% less in some cases even more. So it makes sense that we would like to be able to compete . Since we have spent quite a bit of money on existing

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK