similar to: Anyone heard of BroadVox direct?

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

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?

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

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in

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

Dear R-users, I have got the following problem. I need to create 4x2 arrays of xyplot's on several pages. The plots are created within a loop and plotted using the print function. It seems that I cannot find the proper grid syntax with my viewports, and the more/newpage arguments. The following script is a simplification but hopefully will suffice to illustrate my problem. Any suggestion

Dear R-users, I am trying to understand what the different padding arguments in trellis.par.set are exactly controlling the space around lattice plots. I have used the following code as a basis for testing but it did not really help me to visualize how the value of each argument changes the margins and the plotting area. I guess a better way to visualize the effects of these padding items

So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if

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

Hi. I'd really like to start using Broadvox or one of the many companies that resell connectivity to their network, since they are the only VoIP provider out there that solidly advertises full support for T.38 (I'd be using the openh323 stuff for faxing, since Asterisk doesn't do T.38). However, I really like using IAX for my voice calls. Is there any way to have both? Ultimately

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 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

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

http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy!

Hello, I have a client that has a fairly small installation (20 SIP Phones) that is running Stable. Asterisk appears to be consuming large quantities of memory, and growing uncontrollably to the point where after about 6 weeks the box starts to swap itself to death. I've been keeping my eye on it today, and in the last 12 hours, it has grown by about 8 megabytes, and there has been

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --------------- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 ---------- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 ----------

I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry