Registrations with other VSPs
 

I think it has been hinted at before, but I would like to know if VoXaLot will be adding the feature of being able to register with other VSPs?

i.e. being able to receive incoming calls from DIDs with other VSPs (like the Oztell WebPABX) whilst my ATA is registered with VoXaLot.

Hi moollar,

This enhancement is on our to-do list. At this stage we dont have a planned release date though.

Cool! This would solve some things for a lot of people. :)

Edit: Make sure to let us know as soon as you do add this feature. :)

Yeah have to agree this would be a killer feature, especially for those of us with 'dumb' ATA's that can only register with one server. Currently voxalot is great for outgoing calls, but if it can route incoming calls as well it would be perfect.

In my case I use voxalot with siggate (with a did), freedigits, astratel and ipkall and I'd like all the DID's to be active instead of just the ones that can be forwarded.

Depending on your VSP, it may already be possible to do that. I have an account with SIPME (and a DID that they provided at $5/pm) and they kindly routed the DID to voxalot rather than my SIPME account. So inbound calls to my DID get routed to Voxalot, then to my chosen VSP provider, which is a combination of SIPME and AstraTEL. My ATA is registered with Voxalot. So, inbound works, outbound works, voxalot dial plans work (Yah!) and e164 also works!!!

Quality is pretty much ok (have noticed a few things but it might have been SIPME or AstraTEL problems).

Have you managed to get that one forwarded to Voxalot?

We were going to implement this feature at a later date, however due to popular demand I am pleased to report development has begun.

It would be less of a big deal if more people had DID that could be repointed to a SIP Broker alias or a Voxalot address, but clearly many people use DIDs that are 'tied' to an outbound calling account as well and so find it hard to use Voxalot with their current setups.

Voxalot is certainly compelling, with its open SIP peering, free voice mail, and built ENUM and toll-free calling. But problems remain, like the question of latency between North America and Australia if the VOIP stream is not handed off immediately after setup negotiation directly between myself and the PSTN interconnect I am using, and then there is the whole major problem of three-way calls being presently unavailable for most users. Still, in my opinion, Voxalot has already exceeded FWD in value, and things can only look brighter as more features are rolled out and the bugs are fixed.

That's great news Martin. Happy to do any testing for you :)

Hi ctylor,

Why would the audio stream not be handed off? I thought the way it worked was the SIP connection went via voxalot but the RTP audio stream was direct between the calling parties.

Otherwise wouldn't voxalot end up with outrageous bandwidth bills as their userbase grew?

Darrin

BTW Martin - thanks a lot for getting the development of this feature underway.