similar to: IAX provider using Broadvox's network?

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

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,

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.

Well, in lieu of dropping us, Broadvox has transferred us to their lab switch (keeping our DID's in the process). Now they're complaining that asterisk is sending a Silence-Suppression OFF request of some sort. There's no way to turn this on in asterisk is there? (Yes, I know it will shoot call quality to shit. Otherwise, does anyone know if SER works with silence suppression?

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

Hello, I've been tasked to look for ways to resell to others the service that one of a trunk provides.. In other words, i want to configure my current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act as a trunk to others.. I would provide an IP to them from one of my servers and they will use that IP to connect to me and i will connect them to my trunk/provider. If

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

Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040828/bddb5898/attachment.htm

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

Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have

From: Jean-Christian de Rivaz <jc at eclis.ch> I suspect that some instructions about how to use the diag/mbr was not updated when the source file was renamed to handoff.S. Here is a simple proposition to fix that. Jean-Christian de Rivaz --- diag/mbr/README | 4 ++-- diag/mbr/handoff.S | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/diag/mbr/README

Hi, - I have account on VoIP provider sample.net with auth for example Username: test and Password: test - this provider VoIP use only H.323 and I must use softphone with H.323 support to make phone to PSTN via this provider My idea is configure asterisk to possible make phone from my users SIP softphones to PSTN via this provider sample.net. - I have asterisk with support H.323 ( Ihave use

hello all i am having a problem in compiling openh323. [root@kamran openh323]# ./configure checking for g++... g++ checking for C++ compiler default output... a.out checking whether the C++ compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C++ compiler... yes

Hi, I've installed Centos 5.5 (plus updates) in a machine with INTEL DP43BF motherboard. In order to make Linux detect the PCIs I've added the pci=assign-busses in my GRUB conf. Everything runs fine but within less than 2 days of uptime the machine simply freezes (black console no connectivity). This has happened more than one time so I'm considering to be a problem. The memtest

I'm using ezstream-0.1.2 KJ -----Oorspronkelijk bericht----- Van: Joel Ebel [mailto:jbebel@ncsu.edu] Verzonden: donderdag 3 februari 2005 21:05 Aan: Klaas Jan Wierenga Onderwerp: Re: [Icecast] Stream drops during handoff. Suggestions? Thanks. I'll have to try that. I wonder why ezstream would ever stop sending data for that long though. What version of ezstream are you running? Joel

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but

On 02/02/2017 04:48 PM, Ilia Mirkin wrote: > On Thu, Feb 2, 2017 at 4:43 PM, Phil Turmel <philip at turmel.org> wrote: >> On 02/01/2017 10:40 AM, Phil Turmel wrote: >>> Hi All, >>> >>> I've been running Gentoo on a ZBook with great success for a couple years, >>> but I've been stymied in my attempts to implement SecureBoot by an