Fixmyphone wrote:Thanks for trying to help.
Can you at least tell me whether your Rogers modem/router combo is in bridge mode and whether you have SIP Passthrough in your Asus router set as I've described?
What is your router's firmware version? Are you using Merlin? Do you have UDP timeouts set properly?
I wouldn't put it past my wife's work, municipal town job, to have some half set up corporate VOIP being mishandled by their IT department. My wife is checking to see if they are using some VOIP system through Rogers or similar provider.
Are you unable to enter the main business number at
https://freecarrierlookup.com/? Enter the main business number (without the extension). Answer the question on the website, and click "submit".
Who is the carrier, and do the results indicate it's a wireless carrier (probably not)?
If it's a toll-free number, try here instead:
https://www.800forall.com/SearchWhoOwns.aspx.
This issue 100% originates from just her work and just calls from her work show up in the call logs at freephoneline as 1:00 minute calls
Freephoneline call logs always round to the nearest minute now, regardless of call duration:
viewtopic.php?f=8&t=20404#p79878.
What is the listed "disconnect reason"?
but do not show in the call logs on my OBI
Okay, I was wondering whether OBi202 was dropping the call, but it's not even reaching your ATA.
so something between freephoneline and my OBI doesn't work for just calls originating from her work.
Yes, it could be a SIP ALG issue if you didn't stick the Rogers device in bridge mode and if you still have SIP Passthrough enabled in stock (official) Asus firmware.
viewtopic.php?f=8&t=20182#p78916Your ATA could also be, randomly, losing connectivity if your internet service is unreliable.
The instructions below are for testing to FPL's proxy servers, but I would
also test to 208.85.218.149 and 208.85.218.150, which are the RTP IPs at the moment for FPL.
That's where the RTP audio packets (audio stream) come from.
"Test pings and jitter (you want little to no variation between pings) to the specific Freephoneline SIP servers you plan on using.
Use winmtr:
https://sourceforge.net/projects/winmtr/. Ping about 200 times to each server.
My pings to
-voip.freephoneline.ca average 11 ms.
-voip2.freephoneline.ca average 12 ms
-voip4.freephoneline.ca average 27 ms
If you're using a Macintosh, maybe this helps:
https://www.reddit.com/r/TagPro/comment ... tr_on_mac/When using WinMTR, look at the very last hop or line. Look at your average ping and then maximum ping. Although WINMTR doesn't provide a jitter value, you can get an idea of what yours is by subtracting maximum ping from your average. Jitter is the difference between each successive ping. The bigger the difference, the bigger the problem.
Same with ping, which represents lag or delay. The lower your ping and jitter, the better.
You do not want high pings and lots of jitter (you do not want a lot of variation between each ping). If you get horrible results (pings over 200ms), to any server, you probably don’t want to use that server. So you would want to give that server the lowest priority.
I get between 11 (voip.freephoneline.ca and voip2.freephoneline.ca) and 24ms (voip4.freephonline.ca) on average, depending on the server I'm testing to. Preferably, you want pings below 100ms.
Anything over 200ms is unacceptable.
What you don't want to see is 40, 45, 50, 35,
500, 40, 30, 45,
700. That's bad jitter.
You want relatively consistent pings without a lot of variation.
One reason why jitter can occur is due to other devices on your LAN (local area network) using bandwidth. That’s why properly enabling QoS in your router for your ATA is always a good idea. Refer to point C from
viewtopic.php?f=8&t=20199#p78976.
Bad jitter can produce broken-up audio or choppiness during phone calls. Severe jitter (or large ping spikes) can cause calls to drop (and incoming calls won’t arrive while the ping spike is occurring). Ping affects delay.
I recommend testing pings/jitter between 8 p.m. and 11 p.m. to see if local congestion is a factor (this often is your ISP's fault). Sundays are the best days to test (because that's when most people in your area will be home). 8 p.m. - 11 p.m. is prime time. During prime time (between 8 p.m. and 11 p.m.) cable internet nodes may be oversubscribed in your area and face congestion issues (and congestion can also exist with DSL). So I suggest testing services between 8 p.m. and 11 p.m., particularly on Sundays, when everyone in your area will be home.
Ping is a measurement of data packet transmission, and ping does affect delay or lag. All gamers know, almost inherently, that lag affects them negatively. A PC gamer will pound his or her keyboard in hope that a character will respond on his or her monitor, quickly, but when there's a delay or lag, reality doesn't meet expectation. A gamer can see this problem visually. Over VoIP, anything over 200-210 ms, you will typically start to encounter crosstalk due to increased delay, even if the untrained ear doesn't notice. All VoIP services are subject to the same scientific principles including the fact that speed of transmission affects delay, and Freephoneline is not some magical service that is somehow exempt from issues arising from high pings and jitter. When pings and, especially, jitter are high, it's a pretty horrible experience, just as it would be with any other VoIP service. When pings and jitter are fine, Freephoneline is great.
Lastly, anyone using any communication service (or even when playing online games or using other online services) should understand that the longer the path to the server being used, the greater the potential exists for a problem to occur somewhere along that path. Freephoneline’s SIP servers are located in Ontario."
-- from
download/file.php?id=2195 (pages 16 and 17)
Also, please note that only one registration per FPL account is allowed at any time. When there are multiple devices/softphones using the same account, only the most recent registration is valid. The previous device will lose registration (despite what the ATA's registration status indicates, since 3600 seconds is a long time to update registration status), and incoming calls will not work on it. Registration is required for incoming calls but not for outgoing calls.This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at
https://www.freephoneline.ca/showSipSettings (or if you're trying to register your FPL account with a smartphone SIP app, with another device, or, in this case with an additional SP on your OBi202). Registration is required for incoming calls. This is also important to consider if you're using Freephoneline's desktop application (don't have it running while using your ATA with the same FPL account). Additionally, keep in mind that if someone else is also attempting to register the same SIP credentials on another device where you live, too many registration attempts can result in a temporary IP ban. If you ever see a SIP user agent that you don't recognize after logging in at the above link, someone else is using your credentials (possibly, you've been hacked in that scenario).
One of my routers is the same one that you're using, except I use Asuswrt-Merlin. I have UDP timeouts set properly.
My ISP's gateway/hub is being used in bridge mode (so I don't have to worry about its router features interfering with anything).
Obviously, I'm not experiencing incoming call issues from anyone that I know (Telus, Rogers, Bell, Freedom, etc.), or I'd be posting about it.
The SIP service customers that I know are all able to call me without any issue, although I can't say that I know many.
My OBi202 is configured exactly as outlined in this PDF guide:
download/file.php?id=2195.
Not sure if it is related but she was always able to call before, but as of September 7th, her calls stopped coming through.
Something probably changed. There's a number of possibilities including intermittent problems with your internet service.
The call is reaching Fibernetics. It's in Freephoneline's call logs for your account. Unless your ATA is losing registration, a UDP timeout issue is causing intermittent problems, your Rogers internet service is unreliable at your location, there's a mangled SIP header involved, or something weird going on with the switches FPL uses, I'm not sure why the call wouldn't reach your router. That doesn't make sense if your ATA is registered, if SIP Passthrough is set properly in your router with bridge mode enabled in your Rogers modem/router combo, and if UDP timeouts are set properly in your Asus router. Make sure to enable X_UsePublicAddressInVia in your ATA.
I have also had to restart my OBI several times around this date for about a week as calls were not connecting from freephoneline
voip.freephoneline.ca wasn't working for incoming calls on September 15th for some accounts. Otherwise, there weren't reported issues.
My first inclination is to suspect a problem with Rogers in your area (which you can try to confirm by following the WinMTR instructions in the PDF guide when a problem occurs), unless, of course, you haven't switched to Asuswrt-Merlin and set UDP timeouts properly.
I wouldn't put it past my wife's work, municipal town job, to have some half set up corporate VOIP being mishandled by their IT department
If that's the case, it seems strange that the outgoing call is directed all the way to your FPL account but not to your ATA.
If they're using a SIP service provider on their side, I wonder whether SIP ALG is enabled on their end.
I was thinking of asking you to collect a syslog from the OBi202, but that’s not useful since the call isn't reaching your ATA anyway.