search for: broadvox

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 you're using". We use asterisk. We've asked about using other products (such as SER), and they say they d...

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. Outgoing has been bad. It seems to work the best...

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. Otherw...

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...

...the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally mis...

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 "registering too often". Support either does not respond to e-mails, hangs up on phone...

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 I...

...Subject: [Asterisk-Users] list of T.38 providers on wiki: please > contribute > > > > Please excuse my ignorance but doesn't the VOIP/PSTN gateway > > (Broadvoice,VP Connect) have to support T38 in order for an T38 > > supported ATA to do any good ? > > Yes. BroadVox supports it. I've started a wiki page to track > providers who do, mainly because such providers are so hard to find, > and the available information out there can be so misleading and > unreliable: > > http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+T.38 &g...

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 tested with Grandstream and SNOM phones and both fail 90% of the time Unidata phones fa...

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 I'm already on a call a...

...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, efax, etc aren't good enough). I don't need a DID. Broadvox offers T.38, but don't offer pay-as-you-go. Grr.... - a -- "I didn't see it then, but it turned out that getting fired was the best thing that could have ever happened to me. The heaviness of being successful was replaced by the lightness of being a beginner again, less sure a...

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 the Mediatrix never recognizes that the * box has picked up and continues to ring and then goes on to voicemail. I am beginning to think that...

...* box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the Mediatrix FXS port, I have VoIP service through Broadvox Direct, to one of the FXO ports on my * box. The issues I am having are happening with both the X100P and TDM FXO ports. The issue I am seeing is that the FXO on * is detecting the ring and * is picking up the line and running through the setup I have in the extensions.conf. But the Mediatrix is n...

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

Hi, Anyone knows a DID provider that can do both outbound and inbound? Regards Nhadie

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

All, I use various IAX providers to terminate my outbound calls. I set the caller-ID to one of several DIDs, based on the called number. There doesn't seem to be any rhyme or reason as to what the called user sees, however. Calls to most cell-phones show *exactly* the number I submit. Calls to land-lines sometimes show, other times not. Calls to other voip-providers most often show, but

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

I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone