Currently using Sipstation, but with the ups and downs with the primary and secondary routes, high ping times and other issues, I am now looking for something comparable.
Can anyone suggest something the has USA local and long Distance for a fixed rate? I do not do too much calling and receive a fair amount, but the sip provider must be able to port my numbers over.
Don’t need international, as I use 2 other providers for that.
Thanks in advance.
Guys
I wanted to inform everyone that the module has been fixed and the issue with trunk1 high latency on packets was also resolved. Sorry meant to update people sooner.
So far so good with FlowRoute. I user their .012 per min service which includes free trunks. For us, we would have to make a lot of calls to have .012 add up to 2 or 3 trunks at 14.99 to 29.99 each per month for unlimited.
I use flowroute for my international… May have to look into that, do you pay for incoming?
Have you tried Google Voice? I’ve been using it for a couple months, and haven’t had any problems. Bonus: free, unlimited calling in/out to anywhere in the US.
If you want paid service, I’m a fan of Veracity. Unfortunately, they’re geared toward business use and are therefore somewhat on the expensive side.
I could never get google voice to work right, never could make an outgoing call. Kept getting error messages, not sure why but I think I tried so many different versions of making it work, everything got confused. I am willing to try google voice again, do you have a link to a version that works? or the one that worked for you?
Many thanks for the help, it is not that I can’t find this information. There is just too much out there with different ways of making it work.
There’s a google voice module for freepbx that I used. It’s called “Google Voice” and it’s by POSSA.
The main thing is to create a NEW account, and immediately log out of it and NEVER LOG BACK IN FOR ANY REASON. If you do, say in a browser, cell phone, whatever… calls won’t go to your PBX.
Also, in Google Voice itself, set it to forward to Google Chat and uncheck all other phones.
If it helps, I’m using the standard FreePBX distro, and I don’t think I had to install anything other than the module in FreePBX to get it working.
I would encourage you to give us a bit on this before switching. Schmooze Com just aquired the SIPStation service today and I would be happy to assist you with your issue and make sure its resolved. Please open a support ticket at the new SIPStation support center at support.schmoozecom.com and I will look into it this weekend for you and make sure its resolved immediately. We are also working quickly over the next month to add some new services to SIPStation and some cool features that blend the PBX and the SIP Service together.
Tony, I know it is not your fault, but there has been a serious lack in communications with Sipstations customers. We had to find out the hard way something was going on, and with an 1100ms ping time on my primary, it is way too high, but that is another story. Sipstation module is broken and has been for a long time with no upgrades, I hope now that Shmooze has acquired the service, you can make it more functional.
I am certainly going to try the google voice some more, but I want to continue to support FreePBX for as long as I can, and continue using Sipstation, I have faith that you guys will get sipstation back up and working correctly.
Myself and Philippe will be looking at and working on the SIPStation module this weekend for you. If you have not seen http://www.freepbx.org/news/2013-02-22/freepbx-enters-its-next-exciting-phase
Thanks FluffyNinja, I will give it a go…
Thanks Tony, I will check that link out… it has been a while since I have been around. My FreePBX system is doing a fantastic job and never has any issues. I hope shortly, we can get the Sipstation issue resolved.
Many thanks again.
I’ve had Sipstation for about 1 week and freePbx for 3-4 weeks now
I spent all day adding rules to Linux firewall to allow sipstation
to auto configure trunks with there access key provided i pretty much
opened port 443 and forwarded port 5060, and 10000-20000 with no success to auto config sipstation.
Would it be easier just to manually configure trunks or just
go with google voice if the above mentioned advice about login
correct google’s trunks?
Google Voice is fun to play with, however it’s another of Google’s forever BETA products, and the only revenue they are currently generating on it would be on international. Although it is currently free in North America (until the end of the year) They have not released any future plans for it, and could kill it off next week if they realized they have no use for it, or decide the experiment is over. (Anyone Remember Goog411?) Also, Google Voice does not provide any level of e911 service. Google also can make changes at any time (and have) that will break your PBX’s connectivity to the service, it is a bit of a moving target to keep the PBX modules updated for Google. I would never trust my business numbers on just GV, nor would I trust my home PBX to VoIP service without e911.
Jeff, your ports might not be the issue. Do you have a domain name or are you using dyndns to resolve your IP address?
This is what I have set on my routers for open ports.
port range 5060-5061 UDP
port range 10001-10501 UDP <---- Unless you expect a lot of media channels, you do not need 1000 ports opened up.
They are the only ports I have open at this time.
Inside Asterisk SIP Settings, I have the following.
Nat=Never
IP Configuration=Dynamic IP
Dynamic Host=yourname.dyndns.com
Refresh=30
Localnetworks= Hit the auto configure, or fill in manually.
Mine works fine, but as I have said or noted, mine is on a dynamic IP, if you have a static IP, change it to static and input your Static IP.
Ohh, and do a reboot after.
I was unaware you needed a Dynamic Host.
I have a website. will that work as one?