Multiple trunk from the same provider might limit you to the providers that sell 2-4 channel trunks only. I like Voicepilse for that. Very solid and pretty cheap.
They charge $11 for DID number and the $0.019/min… I think that DID charge covers inbound US/CA minutes (not sure).
Their trunks are 4-channel trunks. You can add more than one DID to each trunk. They provided vast DID selection and you can port existing DIDs to them. You can add more channels to the trunk, but at a pretty staggering cost of $20/channel. Perhaps Voicepulse is not good for 10-channel trunk. Also it is not unlimited minutes as you pay for use. This said… for my office, we are saving money with pay as you go plan as we do not place enough calls to make an unlimited plan cheaper… But we are not placing that many calls, but evaluating your call logs sometimes makes it worth while to go with pay per use plans.
Our main supplier at this point is WiLogic (www.wilogic.com). They are ISP and VOIP provider and more. Not sure if you can use them or not for VOIP in Canada, but their plans are pretty cheap and they can set you up with a single 12-channel trunk and 6 DIDs or whatever you want. With DID that can place international call and DIDs that can not. They are pretty flexible and will make what you want, but I would strongly advise against multiple trunks from the same provider, it just confuses things. If it is one provider, you should be able to do everything you need with just one trunk. This is simple and trouble free way. Once you get in multiple trunks from the same IP to the same IP to your Asterisk behind NAT, you are asking for troubleshooting dropped registration issues.
By the way, if you have technical difficulties with your current SIP provider with the multiple trunk plan, this could be just the problem of multiple trunks. SIP uses pretty convoluted way of keeping the registration alive. Where server send a request and the client sends authentication back. Because it is the same IP where two authentications are requested and received from on the same port sometime the request for Trunk 1 gets response from Trunk-1 and Trunk-2 is such succession that the request for one thing tries to match response from the other thing. Result is incorrect password and dropped registration. Usually takes out all but one trunk. This is more of a how the authentication handling is implemented on the side of the SIP provider and your side. from what I gather it is not a straight forward matter. Voicepulse is capable of dealing with the above no problem, but they have the exact instructions on how to set this up under Asterisk and if that is not done right, same problem is a result.
Well it gets worst because just to have the DID registered, I have to created a number of trunk equivalent to each DID, just so I can send the register string.
That’s the only way to work with this SIP provider and FreePBX / Asterisk (confirmed on this forum and from the provider)… of course they try to sell you their shitty equipment and replace your FreePBX…
I managed to get it working but I keep having service interruption and it is not related to my setup, but rather to their shitty service… customer support is catastrophic too.
I was hoping to get a list of the top provider which offer services in Canada.
8x8 seem to show up a lot in my searches… does anyone have experience with them?
Sounds like BS. You should not need DID to send a registry string. The trunk can have alpanumeric user name and the password. Even in asterisk there are fields for extension number, authorization user name, and password. I think FreePBX is just rolled the two together in the gui so it is a bit more straight forward as most people put the same for both anyway.
I have 12- channel trunk with 3 DIDs ringing into it. Two are phone numbers and one is for fax.
In FreePBX gui I then create three inbound rules. One rings receptionist, another rings IVR where people can pick an extension to bypass receptionist, third one is routed to ATA device which is hooked up to our POTS only fax machine.I manage outbound traffic based on prexixes in the dial plan of each SIP device. I had this setup for about 3 years and have not had any problems (started with Asterisk 2.11 (i think) now I am on the most current freePBX/Asterisk combination).
Good thing about this setup if that we can use as much as all 12 channels for the phone at the times Urtilizing the fax channel) without running out of circuits or needing to do some creative routing.
If ISP is feeding you crap that you need to match number of DIDs with number of channels, they either trying to upsell you the DIDs or they have a system that limits them to this. Either way it is their own unique problem and you should be able to find an ISP that will get you hooked up right. You can have SIP trunk without DID at all (it just there is nothing to call in on, but to call out it will work all day long (including international). You can get trunk like that from voicepulse right now and see for yourself. It is free and requires no commitment. you can set it up in about 2 minutes, with $25 of prepaid minutes and start placing upto 4 simultaneous calls immediately.
Well you’re absolutely right about everything…
Current SIP provider is total BS and they also force a NAT proxy and stuff… anyhow, very unconventional stuff that needs a lot of work from the client when not using their soft phone or their own equipment… I don’t think they’re interested at all in having PBX users, in the end.
So, hence the goal of my post… looking for a new SIP trunk provider.
I’m taking note of VOICEPULSE
If anyone has other recommendations, I’d be happy to hear them.
Back then I had contacted SIP Station and they can’t offer the trunks I’m looking for.
First they were unable to confirm if they could “port” the DID I’m using.
Second, it seems they don’t offer products such as the one I’m using.
Right now I have two trunks.
5 DIDs (Montreal, Quebec, Canada area)
Unlimited in and out calling, local
Unlimited calls in North America
I use the second trunk to pass my calls when my dial plan detects it is going to be a long distance call (or that it’s not a local call).
I also use the second trunk as the “fallback” trunk on my normal outbound route.
Do any of you have anything that looks like these trunks?
I would like recommendation because my current SIP provider is giving me hell and today I got into an argument with them and I am now very much decided to switch as soon as possible.