similar to: T1 passthrough/middleman

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore

Hi all, My office recently purchased an InterTel Axxess system with the IPRC card for VoIP. To our suprise, this card allows the InterTel endpoints and MGCP endpoints to work, but not SIP clients. I was really expecting to get a SIP softphone working with this setup, but that appears to require our vendor to sell us a SIP gateway and licenses at a not yet determined price. With this

Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hello everyone - first time poster, long time lurker. (sounds like a radio morning program, I know). I'm attempting to get my InterTel Axxess (w/v9.0 software) to play nice with my Asterisk implementation. Asterisk 1.2.6 is running on a Fedora Core 4 x86 box. I've tried getting the Axxess to talk SIP to Asterisk, but InterTel's SIP implementation is, well-let's say, incomplete.

Hello - I have (finally) obtained a fair amount of success in connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I can place calls from the Intertel side through the T1, out to an IAX2 softphone and the calls get routed correctly and all of the CID information stays intact. However, when I call from the IAX side to an extension which should route back through to the Intertel

-----Original Message----- > -this is very true, however, the current version of the Axxess software > (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess > upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider.

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

Dear Fil, I am not sure if my answer can help you. I had ever asked a similar question to Daniel and I was using a thirty-party card. As a container uses a shared kernel with the host, so hostdev mode='subsystem' doesn't make sense. Maybe you can try to use hostdev mode='capabilities'. Please see http://libvirt.org/formatdomain.html#elementsHostDevCaps Hope this helps Cheng

Is it possible to call a worker from inside a worker? Right now, if I try, I get a recycled object error.

In the 'me too' vein, I also have an untested 3624 here on the shelf and am waiting on a shipment of T100 cards to play with. Documentation is very hard to come by. Alcatel are certainly the owners of the Mainstreet product line but, without a support contract, any documentation they may have is essentially unavailable as their per-incident fees for support cost more than most of the