Horrible quality with one phone

The carrier you are referring to hides the topology from you. SIP telephony is no different than landline other than the Internet is the. transport.

The network is cooperative and made up of many different peering points. In a major market like Toronto there could be 100’s of media gateways. What number you dial, time of day, load and least cost routing all factor into the routing decision.

If you want to see how this works look at your RTP streams when a call is up to see thr location your media is anchored to.

Certainly we want you to have a positive experience with our trunking product. We only peer with top quality carriers.

Certainly we can help you understand SIP trunking in a forum however I think the constructive dialog is to see what can be done to improve your call quality.

Hi @SkykingOH

Thanks for helping me clarify the situation with general RTP handling. I have a CallCentric trunk as well and they behave in the way you suggested.

However, VoIP.ms does not operate that way. I have been monitoring the call signalling and media streams with them for over a year. They both originate and terminate at the same IP. Port 5060 for the signalling and then a random port between 10000-20000 for RTP in my current situation. I just placed a call again an monitored and it still holds true.

Once one of VoIP.ms’ servers reaches a preset number of assigned trunks they do not allow further assignment of trunks to that server. New customers are added to newer servers until they are full as well before more are deployed.

I understand that after that point where my trunk terminates the call routing is invisible to me and I’m under the assumption that they are doing the best to route the call efficiently.

I have NOT completely ruled out ever using SipStation again. I tried it, made some calls at random points of the day, did some echo tests for comparison and found more latency and choppiness than what I was hoping for.

I did like the SMS integration in UCP. I also liked how I just had to enter my user name and password and then everything trunk side auto-configured.

I’m a very small business right now so the pay per minute pricing offered by other providers is attractive and saving me money versus an unlimited trunk.

As my business grows an unlimited trunk would be useful. The porting fees and the slightly higher DID fees are mainly holding me back from giving SipStaion a production run at the moment. I DO think they’re competitive I just err on the belief that, “if it ain’t broke don’t fix it.”

Thanks again for following up with me.

Sipstation also makes you pay for support even if you are a partner. Never seen any other carrier do this. They are a phone carrier that has a listed 800 number that doesn’t even work on their website. You enter s ticket and can’t get support. I think there are maybe 3 techs. Support is a major problem. Even if you put I a ticket emergency, you have to call, and sort through sales since sipstation tickets aren’t tied to Sangoma. The support guys are good but you have to have credits to keep going. Horrible

Good luck growing your business.

Aren’t voip.ms servers on the West Coast of the US? Your experience is interesting.

The tight integration with FreePBX, integrity of our media behind NAT thanks to our Sangoma Netborder SBC’s and support of the Open Source project are all parts of the SIPStation value proposition.

Montréal (Québec, Canada, 8 servers), Toronto (Ontario, Canada, 8 servers), Vancouver (British Columbia, Canada, 2 servers), Atlanta (Georgia, United States, 2 servers), Chicago (Illinois, United States, 4 servers), Dallas (Texas, United States, 2 servers), Denver (Colorado, United States, 2 servers), Houston (Texas, United States, 2 servers), Los Angeles (California, United States, 2 servers), New York (New York, United States, 8 servers), San Jose (California, United States, 2 servers), Seattle (Washington, United States, 3 servers), Tampa (Florida, United States, 2 servers), Washington (DC, United States, 2 servers), Amsterdam (Netherlands, 1 server), London (United Kingdom, 1 server), Paris (France, 1 server), Melbourne (Australia, 1 server).

NB: The number of servers are at this time, they keep on adding new ones.

I use 4 providers, them being the main one…

Of those for only one of them the trunk is just for signalling, for the other 3, it is for both signalling and media.

Some of them provide features the others don’t have (like T.38) and can be used for (outbound) redundancy, others are cheaper for some destinations (outbound) or have more interesting pricing plans for inbound, etc…

They all have their uses…

Have a nice day and Season’s greetings!


Actually that is in correct. Their is no option in SIPStation to buy support. If you try and open a ticket with SIPStation that is PBX or non SIPSTation related you will be told to use PBXact or FreePBX Support.

As far as 1800 number not sure what you are referring to. The number we publish for support is 920-886-8130.

Also with SIPStation their is no need to open a ticket first or when you call in to enter a ticket.

Based on your input I think you are confusing SIPStation support with FreePBX support.

Four servers in Toronto when I started over a year ago. Demand has grown to the point where they’ve added three more just in Toronto.

This is the toll free number on the sangoma site that does not work. 800 388 2475

Like I said, the support guys are good. Who do I contact if we think it’s a PBX issue? We have exhausted every resource. Replaced all hardware. Even phones. Issue with multiple clients. The only thing the same is sipstation.

After 1-2 rings:

“The number you have reached is not in service. This is a recording.”

(English only… Somehow I was expecting this to be played back in multiple languages…)

Have a nice day and Season’s greetings!


Where do you see this number? That number is not anything I show we own so not sure where you are seeing it but if its on our website then someone fat fingered something and it has never been brought up before that I am aware.

As far as support the SIPStation level will not get into your PBX. If our logs and pcaps show its not a problem on our side that is where we end support on the SIPStation free support and where you have to get into PBX support.


and multiple PDF files…

(one of which is hosted on Sangoma’s site, http://www.sangoma.com/assets/docs/datasheets/en/Lync_Products_Brochure.pdf . You can’t see it in the documents (with my PDF editor I see it’s hidden under “sangoma.com/lync” but Google does pick it up from that official Sangoma document… That PDF should be removed or have this reference edited out properly (pretty easy to do with a PDF editor…)).

Ya I just found that. Seems it was a Pre Schmooze/FreePBX SIPStation number that nobody in the IT side knew we had. The joys of a large organization. Anyways we will get the website fixed. Wish I had known of that issue sooner. Cant believe nobody every saw or caught that.

Tony, I just updated my post…

There’s even one on the FreePBX site… Considering it’s the “legal” page it was probably copied verbatim from something provided by your lawyers…

Have a nice day and Season’s greetings!


Ya our support page has it correct

Seems someone put in some random number as a holding place and nobody ever caught it. Getting it fixed up now.

For testing, can I lock down which codec is being used?

Yes that is controlled as outlined here. http://wiki.freepbx.org/display/PHON/Codecs

Also make sure your trunks have just ulaw enabled. Also make sure you are on updated EPM and Firmware for your phone.

how do you check the firmware? Doesn’t EPM do that automatically?

You manage firmware upgrades in EPM under Firmware Management. We do not upgrade firmware automatically. That is for you to decide and control

looks like I’m running 1.21, how do I upgrade to 1.26. When I click on it, nothing happens