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SIP connected, can call out, cannot receive calls

PostPosted: 02/26/2021
by GTA_doum
My line is configured properly, I can see on the web configuration page of my FreePhoneLine that it is connected, but no in calling calls goes to it! I can use the phone to call out.
I tried to use voip and voip2 servers. Using port 5080.
How to resolve?

Re: SIP connected, can call out, cannot receive calls

PostPosted: 02/26/2021
by Liptonbrisk
GTA_doum wrote: can see on the web configuration page of my FreePhoneLine that it is connected

Freephoneline's web portal information or connected status is only updated every hour, which is the registration interval (3600 seconds) for FPL. In between registration attempts, the status may not be accurate.
Try rebooting or power cycling your IP Phone if you want an updated status.

I tried to use voip and voip2 servers. Using port 5080. and only register on UDP 5060

5080 is likely a local SIP port (LAN), which is different (refer to 5d below).

1) What brand and model modem are you using?

2) What brand and model router are you using?

3) Make sure whatever modem/router combo your ISP gave you is in bridge mode if you are using your own router. Call/contact your ISP if you have to. For Bell Hubs, visit ... r-1993629/

4) What brand and model IP Phone are you using? SPA-942? I'm not familiar with it.

I'll try to find a manual online...

a) Login to your IP Phone.

b) Under Voice-->User (the ones that you're using with FPL)-->navigate to Supplementary Service Settings


a) DND setting is set to NO
b) Block ANC Setting is set to NO
c) DND Activated is set to NO
d) Cfwd All Serv: no
e) Cfwd Busy Serv: no
f) Cfwd No Ans Serv: no

5) Under the LINE (FPL) Tab, ensure the following are set:

a)Nat Keep Alive: Yes
b)Nat Mapping/Traversal: Yes
c) NAT Keep Alive Msg should be $NOTIFY
d) SIP Port: a random number between 30000 and 60000. Pick a number in that range.

Using a high random SIP port may help to bypass SIP ALG in routers, and it also helps to avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potential corrupted NAT association that developed between your router and ATA due to UDP timeouts.

e) If you're having problems, I would try Proxy: to avoid a potential buggy SIP ALG in the router you may be using

6) Navigate to SIP-->NAT Support Parameters, and make sure that the following settings are enabled:

a)Handle VIA received-->yes
b)Handle VIA rport-->yes
c)Substitute VIA Addr-->yes
d) NAT Keep Alive Interval--> 20 seconds

7) Line-->Proxy and Registration-->Register Expires needs to be 3600 seconds (it probably already is set to 3600)

8) Navigate to SIP-->SIP Timer Values (sec)
Reg Retry Intvl should be 120 seconds

Click "Save Settings" button if changes were made

Many older guides for FPL don't include this setting.

9) Proper device reboot (or power on) order is always 1.modem (wait for it to be fully up before turning on your)-->2. router (ensure Wi-Fi SSIDs are populated first on your devices)-->3. IP Phone (wait for router to be fully up and running before turning on ATA). That's always proper device reboot order.

10) 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, and, consequently, incoming calls will not work on it. This is especially important to consider if someone else is using your SIP credentials (username and password) that are found after logging in at 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. A more significant concern, though, is that multiple registration attempts can lead to temporary IP bans. The more devices being used can make the temporary ban happen more quickly. Note that each time you reboot or restart your ATA or SIP app, it's attempting to register with Freephoneline again. Multiple registration attempts within a short period can result in temporary IP ban. Each time you reboot your ATA it's attempting to register with FPL's proxy server.

11) Pay attention to point #4 (and also #2) in the post below this one.

12. If none of that helps, then, unfortunately, you're pretty much stuck with port forwarding your RTP (UDP) port range 16384-16482 (I'm guessing that's the range) from your router to your IP Phone. For reference, that range can be found under SIP-->RTP Parameters-->RTP Port Min and RTP Port Max. You're going to want to double check those numbers in your IP Phone. RTP packets need to reach your IP Phone in order for you get incoming audio. Quite often, when the one way audio issue occurs, this is the problem. RTP packets are not reaching your IP Phone. Ideally, one should not have to port forward in order to achieve proper two-way audio, since port forwarding does create security issues. Port forwarding should only be done when everything else fails.

Refer to the port forwarding section of your router manual to learn how to port forward to your IP Phone. If a router was given to you by your ISP, call your ISP.

Re: SIP connected, can call out, cannot receive calls

PostPosted: 02/26/2021
by Liptonbrisk
(Generic info)

Typically, for VoIP SIP services, especially for Freephoneline/Fongo, you want

1) a router that does not have a full cone NAT,

Visit ... -punching/.
Mango from the forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:,22292023. To run it, use stun from a command prompt.”
Essentially, you download the file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit ... c-commands); and type “stun” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed.

2) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use for the Proxy Server. The purpose of is to circumvent
faulty SIP ALG features in routers.

3) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and 4) A router that lets you adjust both Unreplied and Assured UDP timeouts.

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; for Grandstream, the setting is SIP OPTIONS Keep Alive Interval) < 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, 15, 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_Keepalives expires is supposed to be 20 with FPL.

(the above settings are making reference to those in Obihai ATAs)

Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below/less powerful than an RT-AC68U), third party firmware for Asus routers, does offer easy access to these two settings, which are found under General–>Tools-->Other settings. 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. Note that I will not be held accountable any damage resulting from failed firmware updates. Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well: ... #p28056619.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include


The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 15 for UDP Unreplied Timeout and 115 for UDP Assured Timeout.

ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode. For Bell Hubs, visit ... r-1993629/. For Rogers, visit ... ridgemodem.