volenin wrote:
Has anything changed recently on the Freephoneline side?
No.
A. Visit
http://status.fongo.com/ to check server status.
B. If the service status website doesn’t note any issues, then chances are the problem is on your end. In
your ATA, navigate to Voice-->Line (whichever one you use for FPL)-->SIP settings, and change SIP Port to a random number between 30000 and 60000.
If changing the local Sip Port works, you were dealing with a corrupted NAT association in your router.
Possibly a NAT router connection was never disconnected or never timed out properly. And, then, the
ATA keeps the corrupted connection in a persistent state over and over again. (Credit goes to Mango for
this information). Possibly, this problem is due to the router's UDP timeout being in excess of the ATA's
Failure Retry timer. With FPL, that's 120 seconds. In your ATA, navigate to Voice-->SIP-->SIP Timer Values (sec)-->Reg Retry Intvl should be 120 seconds
Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working
properly with a VoIP SIP provider like Freephoneline, in particular after power failures, the following
conditions must be met:
UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires) < UDP Assured Timeout (in your router) < SIP Registration Failure Retry
Wait Time (or RegisterRetryInterval in Obihai ATAs)
“<“ means less than.
When a modem leases a new IP address, a problem can arise where prior associations using the old IP
address are maintained in the router. When the ATA attempts to communicate using the old IP address,
the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval
(and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where
the corrupted connection persists. If UDP Unreplied timeout is, for example, 10, and the NAT Keep
Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be
created, and everything will work fine.
Another problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often
180 by default in consumer routers): the NAT hole will close due to the ATA not communicating
frequently enough with the SIP server. In turn, incoming calls may, intermittently, not reach the ATA.
Again, X_KeepaliveExpires is supposed to be 20 with FPL
Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers
may be difficult, if not impossible.
Asuswrt-Merlin, third party firmware for Asus routers, does offer
easy access to these two settings, which are found under Tools-->Other settings. Also, my understanding is that third party Tomato firmware has these two
settings as well. So if your router supports Tomato firmware, that may be another option. Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well:
https://forums.redflagdeals.com/recomme ... 2115672/2/.
The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 10 for
UDP Unreplied Timeout and 117 for UDP Assured Timeout.
C. Double check your Registration timers.
https://support.freephoneline.ca/hc/en- ... redentialsi) Voice-->Line(FPL)-->Proxy and registration-->Register Expires should be 3600
iii) Failed Registration Re-Try Interval: 120 seconds
Voice-->SIP-->SIP Timer Values (sec)-->Reg Retry Intvl should be 120
If your ATA makes more than 5 registration attempts in 5 minutes,
you may end up being temporarily IP banned by the specific FPL server the ATA was sending
registration requests to (each time you reboot the ATA, it's attempting to register with FPL). If you're temporarily IP banned, you could then try switching Proxy to a different FPL server than the one you were previously using (voip.freephoneline.ca, voip2.freephoneline.ca, or voip4.freephoneline.ca:6060), unless you need to use voip4.freephoneline.ca:6060 because you have SIP ALG forced on in your router. The purpose of
voip4.freephoneline.ca:6060 is to circumvent SIP ALG features in routers.
https://community.freepbx.org/t/trunk-s ... ca/22479/8"As May 2013, our servers will rate limit REGISTER requests to a maximum of 10 requests per 5 minutes. Each authentication round usually consumes 2 requests (digest auth), so it is a fair number given our guidelines. Also, it does not affect INVITES (which are also authenticated)…
This rate limit is applied per IP address as our service is tailored to residential Canadian users (ADSL/Cable)."
D. If the ATA loses registration for any reason, incoming calls won't work on it. 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. 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 or with another device). Registration is required for incoming calls. It is not required for outgoing calls. If you simply want to make outgoing calls using your FPL number, configure, but don't register the account, on the SIP app being used. 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. Always check registration status in the ATA and also your SIP status after logging in at the above link. If you see a device listed under SIP User Agent that you don't recognize, you've either been hacked or someone else is using your Freephoneline SIP username and SIP Password.
E. Navigate to Voice-->Line (FPL)
i) NAT Keep Alive Enable should be Yes
ii) NAT Keep Alive Msg should be $NOTIFY
F. Navigate to Voice-->SIP
i) RTP packet size should be 0.02
ii) NAT Keep Alive Intvl should be 20
iii) Handle VIA received should be yes (for 1 way audio issues)
iv) Handle VIA rport should be yes (for 1 way audio issues)
v) Substitute VIA Addr should be yes (for 1 way audio issues)
G. SIP ALG in some routers has been known to cause problems. You may want to try disabling it:
https://www.obitalk.com/info/faq/sip-al ... ter-failed.
If you were issued a modem/router combo or gateway by your ISP, please contact your ISP for assistance or refer to the device manual.
Typically it's best to have your own router and stick the modem/router combo into bridge mode.
H. Latest ATA firmware is found here:
https://www.cisco.com/c/en/us/support/u ... -downloads.
I. I don't know whether this T1 bug has been fixed in the latest firmware for SPA112/SPA122 ATAs: https://community.cisco.com/t5/atas-gat ... -p/1930651.
Basically firmwares were using 10% of the value for T1 that was listed.
a) If that bug still exists, navigate to Voice-->SIP-->SIP Timer Values-->T1
Change SIP T1 to 10.
b) If that bug doesn't exist, change SIP T1 to 1.For an explanation of this issue, visit
http://www.dslreports.com/forum/r294151 ... SIP-T1-bug, and read Mango's posts.
J. You can also try rebooting your modem–>router (wait for for Wi-Fi SSIDs to appear first; wait for the router to be fully up and running)–>ATA (in that order). This is always the proper device reboot order.