similar to: Which mix of VOIP services do we need?

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone

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

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

I am looking for only a few DID's in the above areas using IAX. If the solution works to my customers satisfaction, I will be interested in getting whole sale prices. CA - California, mostly LA. WA -Washington, mostly Seattle BC - British Columbia, Canada, mostly Vancouver FL - Florida, mostly Miami, Fort Lauderdale and West Palm Beach NY-NY - well, the City.

Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX

Okay, first a little background - I've been with Packet8 since a month after they started. I found that we were outgrowing their services and decided to move to an asterisk box in the office. I found a service provider that offered me a reasonable rate. After a fair ammount of testing I decided to stick with their services and port my 3 primary DID's from Packet8 to the new service.

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

Just an FYI guys we have some of the leading open source developers and projects going to speak/showcase at Cluecon. These include: Mark Spencer - Asterisk Bob Andreasen - SIPFoundry Craig Southeren - OpenH323 David Sugar - Bayonne This should be an exciting event for all. Register Today! hope to see you there! Brian West Asterlink.com _______________________________________________

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

Where can I go to get blocks of DID's or single DID's? CHEAP :) Thank You David

In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-biz]

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/38108fe4/attachment.htm -------------- next part -------------- Hi, This is Shan. I need a suggestion regarding my work. First i'll explain my work and next my problem in the extensions.conf file: Work ---- According to my dialplan every incoming DID's will be diverted to the

Hi, here is my scenario in my voip system(asterisk based) every user has a primary did and 5 secondary did's i.e. all six did's point to a single channel. every user has a blacklist feature and a call filter feature. if blacklist feature is enabled, user has option to include 5 bl numbers. if the user is using all 5 bl numbers then i have to match DNID with every(6) did. im not using

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or

How do you get the called number on incoming SIP calls? I've never had multiple DID's via SIP from one provider before and somehow I never realized that with IAX it just works, and SIP is different. If I don't set an extension in the register command the incoming invite has <sip:s@me.com> in the To field. Now if I have multiple DID's that I want routed to different

I thought maybe one of the providers here could answer this question. When using IAX to make calls, it passes the called number. Being designed to work with a PBX this makes sense. SIP works differently though and I'm curious why providers don't have a way to pass the called number on their DID's, or choose not to do so. Providers that themselves use upstream SIP proxies are

I know I can get free DID's with SIP, is anyone giving out free DID's with IAX? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000