similar to: SIP <--> PSTN gateways

Just received: Cognigen is very proud to announce the official launch of Broadvox Direct, a new VOIP service. Broadvox Direct offers unlimited calling plans to anywhere in the US and Canada for a low monthly payment starting at $29.95 and basic accounts as low as $12.95. http://cognigen.net/broadvox/?mu Anyone know who's behind that? It's not BroadVoice, is it?

Guys, For what it's worth, after months of trying to troubleshoot issues with them, and after paying them around $2500 for setup and a down payment (it's unclear what of that will be refunded, if any) BroadVox -- http://www.broadvox.net/ -- decided to terminate our contract without any valid reason, and the only explanation they could cite was "it's because of the software

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

I have a NY 212 packet8 service if you would like to work with me to set this up on my A@H service, I'm happy to test this with you. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Adam Megacz > Sent: Sunday, 7 August 2005 5:29 PM > To: asterisk-users@lists.digium.com

I'm wondering if any other Asterisk users have a recommendation for a reliable SIP Trunk provider that supports Asterisk and offers decent support. I've worked with Coredial, Broadvox, and Broadvoice and have had some bad experiences with each of these providers. Broadvoice offers low cost service, however I have constant issues with Broadvoice blocking my customers due to Asterisk

Hello all. I don't know how I got onto this list, but now I'm here I'd like to ask a question: Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? Is this even the right place to ask such a question? Thanks in advance and apologies if I'm in the wrong place. Cheers. Paul. -- Work: https://paulc.es

So, I've been able to receive faxes quite reliably through teliax with g711 so far; I think I can live with it. For outbound faxing, I'd really like to get a service that lets me send faxes, but doesn't charge me a monthly fee (I don't send enough faxes to justify it). T.38 is a requirement; I need to know that a fax has gone through at the time I send it (store-and-forward,

Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But

Apologies if this is an FAQ, I wasn't able to find an answer googling: Will any of the public PSTN/VoIP gateway services (Vonage, Packet8 etc) interoperate with * ? I'd like to deploy a box which provides PBX service for analog handsets, and handles inbound/outbound calls via both analog PSTN lines, and, say Packet8 VoIP service. I understand that I can do this by connecting the analog

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

On 10/07/15 04:26, Charles Lepple wrote: > On Jul 9, 2015, at 9:43 AM, paul at paulcarmichael.org wrote: > >> Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? > > Not sure. There was a thread in February talking about an Ovislink Chrome 1000, but I don't think we ever resolved what was causing the "Device busy" error: > >

On 10/07/15 04:26, Charles Lepple wrote: > On Jul 9, 2015, at 9:43 AM, paul at paulcarmichael.org wrote: > >> Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? > > Not sure. There was a thread in February talking about an Ovislink Chrome 1000, but I don't think we ever resolved what was causing the "Device busy" error: Is there an

Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote

I have been transcoding a group of panoramic videos using ffmpeg2theora 0.26 i just completed transcoding Naples Cycle HDV.vwmovie the resulting video has some strange colors in it the source video can be got from http://www.eyesee360.com/videowarp/examples/ the resulting images and the theora file can be download from http://www.lightningridgecommunity.com/images/ tom at

Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102 (yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of

I found some old messages regarding a possible pkt8 DTA "bypass". Anyone is using Packet8 with Asterisk? ========== http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat Got Softphone Working with Packet8 Friendly name: {Anything you'd like.} SIP domain: packet8.net SIP proxy: packet8.net Leave everything else at the default. When you login, it will ask for a username

Noob here, my apologies in advance. Recently my employer decide to stop paying for my home POTS line, so I ordered packet8 for a home line instead of another POTS line, since I really dislike my local phone company, and the POTS line without any long distance would cost more than the $20 to packet8 with unlimited US calling. Anyway, this whole thing got me started thinking about VOIP a lot

I received a message from Packet8 last night telling me they added some new area codes. I took a look at their new area code/rate center finder page and it looks like the added a LOT of new area codes. They now have phone numbers available in almost every state. The URL for finding out which area codes and rate centers are available with Packet8 is at http://www.packet8.net/about/areacodes.asp

Hello, I have a linux box with the following S/W versions: CentOS release 5.2 (Final) Linux ns.hostname.ro 2.6.18-92.1.13.el5 #1 SMP Wed Sep 24 19:32:05 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Version 3.0.28-1.el5_2.1 My server has 2 IP interfaces, eth0: with the public ip address and br0: for the private subnet. Br0 being a bridge ip interface between eth1 and tap0 ( for openvpn ). Now my