Now look what at what they are doing....

[tbrummell]

It seems there are multiple servers in the mix today....

My registration server:

Code: Select all

FPL/1613729XXXX            208.65.240.165                                             5060     OK (30 ms)

An incoming call....

Code: Select all

[2013-01-22 15:49:17] NOTICE[3425] chan_sip.c: Sending fake auth rejection for device "CallerName"<sip:81977XXXXX@208.65.240.170>

I'm registered to .165, yet the call is coming from a server at .170. Needless to say, my Asterisk server is rejecting it. Thanks for the heads up you're making changes IDIOTS.

For you Grandstream owners that have "Only accept invites from SIP proxy" or something like that, this will affect you too. The incompetence seems to be expanding.

[tbrummell]

Now, after a sip reload, we are registered to the new server. I wonder where this all leading? An explanation from someone @Fongo would be nice, but it seems someone has broken their fingers and they have no STT technology there. I wonder if they know that Google has a free to use STT API?

Code: Select all

FPL/1613729XXXX            208.65.240.170                                             5060     OK (28 ms)

[Jake]

I have two trunks registered and they are using 208.65.240.165 and 208.65.240.160

this will confuse things.....

[tbrummell]

Yeah, it wouldn't be bad if they send the call to the server that you are registered to, but they didn't appear to do that today. If that continues I'll have to port.

[Jake]

Well as I said in another thread, I think they have set up load balancing servers, or some sort of CDN cloud, and that calls are going to be coming from any one of them.

ATA users need to go back to a one server setup like we had before. The app may be able to cope with all the different servers changing all the time. Delicate ATA/PBX registrations are going to struggle unless we want all and sundry calling us day and night.

[FONGO_steve]

It's always advised to use our hostname SIP servers (sip.fongo.com/voip.freephoneline.ca) and to not setup devices to bind to the IP addresses that a server may resolve to at any given time. As we continue to grow and scale as a company, so does the architecture that drives everything behind the scenes. Just like any large scale server deployment, be it VOIP, email, DNS, etc. the SIP proxy may resolve to any given node at any given time, and a different node another time; this makes for a more disaster proof setup, among many other advantages.

Our own ATAs, softphones and apps will deal with this seamlessly and don't have any troubles, but any BYOD users who have setup a more delicate or locked "down device" configuration above what we recommend (for devices where recommended configurations are available) may experience trouble until those restrictions are loosened up a bit. I'll suggest some better documentation in the online FAQ to better explain this for BYOD users too and encourage community members to help contribute to our "Device Configuration Sharing" section so we can build an accurate and robust library of configuration for third-party devices.

We always do our best to communicate major, planned maintenance and our new service status page should also be beneficial in communicating service impacting problems or maintenance: http://status.fongo.com/

However, one thing to keep in mind is that just like any company, we are continually growing and enhancing our network and small changes and enhancements are being made day in and day out and aren't necessarily communicated unless they are deemed to be of high importance or service affecting. If we were to start communicating every little tweak being made then users would eventually start to ignore said communication and it would be more difficult to make sure the important messages are in fact not ignored.

More updates on the recent changes will become available in the near future.

[Jake]

We always do our best to communicate major, planned maintenance and our new service status page should also be beneficial in communicating service impacting problems or maintenance: http://status.fongo.com/

You sneaked that one in there, why wasn't it announced (or is this the announcement)? May I suggest you put links in the footer of the support pages that point to this new page.

I hope it actually gets updated!

[FONGO_steve]

We always do our best to communicate major, planned maintenance and our new service status page should also be beneficial in communicating service impacting problems or maintenance: http://status.fongo.com/

You sneaked that one in there, why wasn't it announced (or is this the announcement)? May I suggest you put links in the footer of the support pages that point to this new page.

I hope it actually gets updated!

It's hot off the press Announcement coming very soon!

[tbrummell]

It's always advised to use our hostname SIP servers (sip.fongo.com/voip.freephoneline.ca) and to not setup devices to bind to the IP addresses that a server may resolve to at any given time. As we continue to grow and scale as a company, so does the architecture that drives everything behind the scenes. Just like any large scale server deployment, be it VOIP, email, DNS, etc. the SIP proxy may resolve to any given node at any given time, and a different node another time; this makes for a more disaster proof setup, among many other advantages.

Our own ATAs, softphones and apps will deal with this seamlessly and don't have any troubles, but any BYOD users who have setup a more delicate or locked "down device" configuration above what we recommend (for devices where recommended configurations are available) may experience trouble until those restrictions are loosened up a bit. I'll suggest some better documentation in the online FAQ to better explain this for BYOD users too and encourage community members to help contribute to our "Device Configuration Sharing" section so we can build an accurate and robust library of configuration for third-party devices.

We always do our best to communicate major, planned maintenance and our new service status page should also be beneficial in communicating service impacting problems or maintenance: http://status.fongo.com/

However, one thing to keep in mind is that just like any company, we are continually growing and enhancing our network and small changes and enhancements are being made day in and day out and aren't necessarily communicated unless they are deemed to be of high importance or service affecting. If we were to start communicating every little tweak being made then users would eventually start to ignore said communication and it would be more difficult to make sure the important messages are in fact not ignored.

More updates on the recent changes will become available in the near future.

/TL;DR
I do use host voip.freephoneline.ca, above is what it was resolving to at the time I posted that, before reloading SIP on the Asterisk box.

Now I'll go ahead and read your post.

As for the status page, I doubt it'll be accurate, but time will tell.

[tbrummell]

Please go ahead and post the IP block that *may* tentatively send me a call, and I can go ahead and open up my configuration for that block.

You better have all those Grandstream users remove that option from their ATA's as well. Don't forget to explain to them why they are all of a sudden getting various calls at all hours of the night again too. You are opening a big can of worms for your support people.

[Jake]

The CIDR for those servers seems to be 208.65.240.0/21

I have seen 4 servers in use so far, 208.65.240.175, .170, .165, .160