sip disconnected on Grandstream HT-286

[Roger]

Hello

for about 2 weeks I can no longer send or receive calls with grandstream ht286. It was working before I have not changed anything in the configuration. I hear a dial tone and dial a number but nothing happens afterwards. I am went to my freephoneline account and in sip settings it says sip disconnected.internet is working fine.

[Liptonbrisk]

Click viewtopic.php?f=8&t=20199 (also refer to Points B and D from the second post below in that thread).

Follow the steps, step by step, down the list:

A. What brand and model modem are you using? Ensure it's in bridge mode if it's a gateway, hub, or modem/router combo. This can be important. Contact your ISP if you need help enabling bridge mode.

B.Visit http://tplinkwifi.net, and log in with the username and password you set for your router.
i) Navigate to Security--> Basic Security --> ALG
ii) Disable "SIP ALG".
iii) Click "Save".


C. Login to your ATA.

i) Select "Advanced Settings" tab
ii) Change SIP Server to "voip4.freephoneline.ca:6060" without the quotation marks (voip4.freephoneline.ca:6060 helps to circumvent potential SIP ALG problems; and if you were temporarily IP banned, changing to a different sip server will resolve that problem, at least, temporarily)
iii) SIP Registration should be "Yes"
iv) Register Expiration needs to be 3600 seconds (otherwise you risk temporary IP bans)
v) Use Random Port should be "Yes"
vi) Change SIP Registration Failure Wait Retry time to 120 seconds (otherwise you risk temporary IP bans)
vii) NAT Traversal should be "Yes". No stun server needs to be entered.
viii) Keep Alive Interval should be 20 seconds

D. Update settings and Reboot.

This is always proper device reboot order:

a.Turn off modem, router and ATA (or IP Phone or close SIP App).

b. Turn on modem. Wait for modem to be fully up and running.

c.Turn on router.
Wait for modem to be fully up and transmitting data before turning on router.

d. Turn on ATA (or IP Phone or open SIP app) only after the router is fully up and running.

The SIP device (ATA) should be the last device powered on in the device chain.

E. Log back into your ATA. Check registration status. Navigate to the Status tab. Do you see "Registered"?

F. What does the LED light pattern on the ATA indicate at this point?
https://www.voipsupply.com/downloads/dl ... _guide.pdf (page 14)

RED LED indicates not normal status

Button flashes every 1 second If Ethernet link is down
Button flashes every 2 seconds. (if DHCP is configured) DHCP Failed or WAN No Cable
Button flashes every 4 seconds. (if SIP server is configured) HT–286 fails to register
Button flashes every 6 seconds. Firmware Upgrading
Button flashes briefly No STUN responses
Red light steady. Device Malfunctions

GREEN LED indicates normal status

Button flashes every 2 seconds. Message Waiting Indication
Button flashes at 1/10 second. RINGING
Button flashes every second. RINGING INTERVAL
Green light steady. In Conversation

--

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 FXS Port or Line on your Grandstream ATA). 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 with a short interval 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).

[Liptonbrisk]

Also, ensure you can ping or reach FPL's SIP servers. If you can't, you (or your ATA) may be experiencing a DNS issue.

---
"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)

[Roger]

Thank you for the support

my router tplink tl-wr940n in standard mode wireless router.1-disable sip alg router.2-login ata and done all 8 steps.3-update settings and reboot,done all 6 steps.4-led light not flashing and sip registered on status in ht286. I looked in my freephoneline account and sip is always disconnected and user agent nothing written. I ping 200 times voip,voip2,voip4 freephoneline.ca the result is less than 30 ms. tries to dial a number I hear the dial tone he dials the number but he says data is not valid and the light is green on ht286.

[Liptonbrisk]

my router tplink tl-wr940n in standard mode

I'm more worried about the modem B2B2C gave you. Do you know the brand and model of it?

I looked in my freephoneline account and sip is always disconnected and user agent nothing written.

It may take 10 minutes for SIP Status to update properly at https://www.freephoneline.ca/showSipSettings because of https://status.fongo.com/incidents/cjE1TACuHrD3.
Updates or setting changes made in your ATA will take awhile to appear.

Hopefully, scheduled maintenance will fix the status update taking too long to appear: https://status.fongo.com/maintenances/Tpzi6pRncwOz.

he says data is not valid

Are you sure you heard a male voice? And are you sure you heard the word, "data"?

1) Do you hear a female voice say, "This account number is not valid?"

a) Login at https://www.freephoneline.ca/showSipSettings and check SIP Username.

b) Login to your ATA. Navigate to Advanced Settings-->SIP User ID. Ensure you have entered in your FPL SIP Username (don't copy and paste) carefully, without extra spaces before the 1 or after the final digit. Re-enter it.

Be careful not to use an automatic form filler or password manager.

c) Do not enter anything for Authenticate ID. Leave it blank.

d) Update and Reboot ATA.

2) Do you hear a female voice say, "That password is not valid?"

a) Login at https://www.freephoneline.ca/showSipSettings and check SIP Password.

b) Login to your ATA. Navigate to Advanced Settings-->Authenticate Password. Ensure you have entered in your FPL SIP Password (don't copy and paste) carefully, without extra spaces before the first letter or after the last letter. Manually re-type your SIP Password

Be careful not to use an automatic form filler or password manager.

c) Update and Reboot ATA

If you hear a female voice say, "This account number is not valid" after doing steps 1a,b,c,and d (type your FPL SIP Username carefully), then submit a ticket: https://support.fongo.com/hc/en-us/requests/new.
Choose "My Account Inquiry" for the final issue type. Let them know the error message you're hearing. Let them know the problem is happening after the Call Platform Migration: https://status.fongo.com/maintenances/KSqP5KV3SjTd.

[Liptonbrisk]

I've never understood why registration doesn't fail with Freephoneline when SIP Username or SIP Password is entered incorrectly.

For what it's worth, your ATA's firmware version appears to be old.
You can find firmware updates here (under "End of Life Products"): http://www.grandstream.com/support/firmware.

[Liptonbrisk]

I looked in my freephoneline account and sip is always disconnected and user agent nothing written.

It may take 10 minutes for SIP Status to update properly at https://www.freephoneline.ca/showSipSettings because of https://status.fongo.com/incidents/cjE1TACuHrD3.
Updates or setting changes made in your ATA will take awhile to appear.

Hopefully, scheduled maintenance will fix the status update taking too long to appear: https://status.fongo.com/maintenances/Tpzi6pRncwOz.

Today's maintenance work seems to have fixed the lag with SIP Status. I'm seeing "connected" immediately when registering with voip4.freephoneline.ca:6060 at this time after logging in at https://www.freephoneline.ca/showSipSettings. "disconnected" appears immediately too.