Recommendation for SIP trunk to replace GV

With Google Voice ending XMPP support next month, I need to find a SIP trunk service.  It seems like finding a good SIP trunk service is difficult.  Obivoice was good but looks like they are about to go out of business.  Here is my situation:

  • I'm in the USA (California)
  • I need 3-4 DID
  • I use about 500 minutes a month, between all lines, equally divided between incoming and outgoing.
  • I need something that is solid and reliable on the inbound.  GV has been HORRIBLE with dropping the inbound connection about every day or two.  (You get what you pay for.)
  • I don't need international long distance
  • Rarely call out of my state.
  • Most of my incoming and outgoing calls come from a 100 mile radius.
  • I think a cap of $10 a month for this is realistic.  Some SIP trunk providers I have seen charge $20-40 a month.  If I had a lot of minutes, then maybe, but I don't.

Thank you in advance!

The server in question
I have FreePBX Core 2.11.0.20 with Asterisk 11.2.1 (Asterisk 11.2.1 built by root @ 237-138-19-10.digium.internal on a i686 running Linux on 2013-03-06 18:56:51 UTC) The only editing I have done at the CLI level was to some of the .conf files, as directed by instructions, to get the Digium Free Fax License to work.

My background
Although I have 25 years experience in I.T. and my own I.T. corporation, I just got into FreePBX in January 2014 using the FreePBX distro. Despite being new, I have probably accumulated about 200 hours of experience on the FreePBX system. I have set up two FreePBX servers (one 32 bit and one 64 bit) and learned about every feature of the base FreePBX, a few add-on supported modules, and even set up the free Digium Fax License on both servers (that by far was the most difficult process I had to do on FreePBX.) I have set up custom inbound routes that route my cell phone to a "the system is working" message so I can check on my systems. The only thing I have not gotten into is auto provisioning. Finally, I have set up about 20 Yealink T20 and T28 phones. (BTW, I really like the Yealink phones on both the user and admin level. Other than setting the custom graphic on the T28, these phones are super easy to configure.)

I would suggest that anything you find below your pricepoint is probably also soon going out of business. For multiple DID’s expect to pay one to two dollars a month retail for each DID, calls on a per minute basis should be less than 2 cents you can usually get two concurrent calls without paying for extra trunks, If you want quality service as you say you get what you pay for, if you have fax needs, then many VSP’s will disappoint you, given your location and needs you could look at Vitelity, they are reasonably priced, do faxing over g711 without problem, and are very reliable with a track record of both endurance in the market and reliability there are cheaper vendors but not by much.

Try flowroute.com

+1 on Vitelity. I have 160 DIDs with them and they are rock solid in my opinion.

Yeah for supporting the project guys :frowning: Not a single mention of SIP Station?

Well I am biased but I would recommend SIPStation and it has full T38 for inbound and outbound faxing and works very well. As a bonus its run and owned by the FreePBX crew so its what pays our bills here.

My reply to the OP was to his usage and requirements:-

I think a cap of $10 a month for this is realistic. Some SIP trunk providers I have seen charge $20-40 a month. If I had a lot of minutes, then maybe, but I don’t.

I believe that would preclude Sipstation. Do you guys do pay-per-minute?

After shopping around, it seems that finding a good, reliable SIP provider is as easy as finding a honest politician. I am not about to sign up for two years (paid upfront) with a company that hasn’t been around for two years. No way.
Although the price point of SipStation seems high, if it is reliable, I can swing that.
So if I order one 2-way trunk from SipStation and I am on the phone when someone calls, will they receive a busy signal? I have heard that in reality, no they won’t, they will ring through/go to my FreePBX system.

We were happy with SIP Station in the beginning. However, use in production revealed several issues that put me on a look out for another provider.

  1. Use limit. You might not realize that ‘unlimited trunk’ is limited to 3,000 minutes. The cap seems more restrictive than for most providers , 5K is more typical

  2. Support. We had an issue where our trunk started returning congestion messages. Things happen, right? Tried to reach support on Sunday - noone is available. Once reached on Monday, the guys kept pushing this back on us saying this is our problem and their system shows an active call. Overall it did not seem like they were focused on resolving the issue. The focus was on finding a reason not to resolve. I made a suggestion to restore the service while they are investigating and the answer was “you can buy another trunk”. This turned out to be a case of a “stuck call”. SIP station support blames our side for never sending a session termination packet. I checked the service and there were no crashes. So, how can one be sure this is a client issue? You might have missed the packet due to network outage. Besides, clients do crash, power goes off, Internet connectivity gets impacted, you name it.

I do not see a problem with service occasionally not working. The problem with SIP Station is that there is no support to provide service SLA outside of business hours. Support personnel does not seem to be interested in resolving the issue.

Bottom line - while the sevice worked it was OK except for the low minute limit cap. When service was not working, there is no support to restore the service afterhours.

anatolyk, We appreciate your use of SIPStation for your business communication services, I’ll address a couple of your concerns below.

  1. Most SIP providers actually have a very vague “Fair Use Policy” statement that they in their sole discretion will determine what is considered typical business usage, we actually define a number for clarity and to provide clear expectations of our service parameters.

  2. We monitor the system for outages 24 x 7, we had no system wide outages this weekend. The ticket that you opened on Sunday was for a different account than the one you were having issues with, the accounts were not linked. Technicians reviewed that account when you opened the ticket and saw no issues with it. The error message that you stated in the ticket was related to getting a “Trunk Concurrency Limit Reached” message. Meaning that you didn’t have enough call paths for the number of calls your were trying to make. Monday morning was the first time that you clarified that this ticket was for a different account than the one you opened it for.

You and I spoke for 35 minutes this morning, during which time I walked you through the process of logging into your account, and then we had our support department merge your two accounts to make it easier in the future.

It’s unfortunate that you had a bad experience this weekend, after our call this morning I assumed everything had been cleared up. Had your ticket been submitted under the correct account the tech that reviewed it when it came in on Sunday could have quickly resolved the issue for you.

For future reference:

Schmooze/SIPStation support is available Monday thru Friday 8:00AM to 6:00PM Central Time.
Support tickets are monitored after hours for service outages and will be responded to in a timely fashion. Other issues submitted after hours will be responded to during regular business hours.

http://wiki.freepbx.org/display/ST/Support