No incomign calls after 30 minutes of making a call

[daveguy]

This problem started out of nowhere about 1 month ago. I can make calls just fine however, I get no incoming calls after about 30 minutes of phone being unused.
If I make an outgoing call or reset PAPA2, I do get incoming calls for like 30 minutes and then it stops again, it's like making a call registers my ip for 30 minutes or something.

I am using a PAP2 box. The lights are on normally. I have tried replacing the PAP2 box with a new one, replaced the phone, all the cabling but the problem still remains.

Any help would be appreciated.

[Liptonbrisk]

Beware fake PAP2s: http://genuinelinksys.blogspot.com/2011 ... nuine.html
https://www.ukvoipforums.com/viewtopic.php?t=565.
Fakes are junk and often found on eBay or Kijiji.

PAP2s have been discontinued for quite awhile now.

Preliminary Steps

i) What brand and model modem are you using?

ii) What brand and model router are you using?
If you have your own router, ensure that the modem/router combo issued by your ISP is in bridge mode. Contact your ISP if you need help enabling bridge mode.
For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem
For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/


iii) If you don't have your own router, ensure SIP ALG is disabled in whatever modem/router combo your ISP issued.
For example, Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combos from Rogers (and possibly other ISPs) requires users to do the following:
Open the web browser, and login at 192.168.0.1. Default username is cusadmin.
Select the “Basic” tab and disable “SIP ALG.” Click the “save changes” button.
Contact your ISP if need be.


A. Ensure, after logging in at https://www.freephoneline.ca/showSipSettings that

i) SIP Status shows "connected", and
ii) SIP User Agent reflects a device that own and recognize. If you don't recognize the SIP User Agent, chances are you've been hacked.

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 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. 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.



Regular Steps

Follow these steps carefully, step by step, down the list.


1) Disable DMZ and all port forwarding in your router. Port forwarding is a security risk and should only be used if all else fails.

2) Disable the SIP ALG feature in whatever router or modem/router combo it is that you're using:
visit https://www.obitalk.com/info/faq/sip-alg/disable-alg for an example

DLINK router users may need to log into the admin page of their router, click the "Advanced" tab and then "Firewall Settings",
navigate to "Application Level Gateway (ALG) Configuration", and uncheck SIP.

Refer to SIP ALG issues in the post below.

3). Login to your ATA. Specify a high random SIP port in your ATA between 30000 and 60000.
Navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, change SIP Port to a random number between 30000 and 60000.
Do not use the same random SIP port for any other Line. Always choose a different random local SIP port for each Line you're using.

Using a high random SIP port may help to bypass SIP ALG, 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. Refer to the final point in the post below this one.

4. In the ATA, navigate to Voice-->SIP tab-->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

e) click "Save Settings" button

This helps to ensure the RTP audio stream is being sent to your WAN IP as opposed to your LAN IP.

5. Navigate to Voice-->Line (whichever you use for FPL)-->NAT settings
a) NAT Mapping Enable should be yes
b) NAT Keep Alive Enable should be yes
c) NAT Keep Alive Msg should be $NOTIFY

d) click "Save Settings" button if changes were made

6. Navigate to SIP tab-->NAT Support Parameters category-->NAT Keep Alive Intvl should be 20

a) click "Save Settings" button if changes were made


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

Click "Save Settings" button if changes were made

https://support.freephoneline.ca/hc/en- ... /212430746

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

8. In your ATA, navigate to Voice-->Line-->Proxy and Registration-->Register Expires needs to be 3600 seconds (it probably already is set to 3600)

9. Save settings. Turn off both router and ATA. Turn on router. Wait for router to be fully up and transmitting data. Turn on ATA.
If the problem is solved, don't continue.

10. Try Proxyserver voip4.freephoneline.ca:6060
visit http://forum.fongo.com/viewtopic.php?f=15&t=16294 (look at page 6 of the .pdf)

voip4.freephoneline.ca:6060 is a SIP server whose purpose is to help those with SIP ALG issues (can't disable it in the user's router, for example).


Refer to SIP ALG issues in the post below.


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

12. Please take the time to update the router field in your forum user profile: http://forum.fongo.com/ucp.php (select the profile tab). Thank you.

[Liptonbrisk]

(Generic info)

Typically, for VoIP SIP services, especially for freephoneline, you want

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

Visit https://www.think-like-a-computer.com/2 ... es-of-nat/.
Mango from the Obitalk.com 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:
http://www.dslreports.com/forum/remark,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” 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
https://www.voip-info.org/routers-sip-alg/ (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 voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 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 https://www.voipmechanic.com/qos-for-voip.htm.
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) < 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, 17, 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: https://forums.redflagdeals.com/recomme ... #p28056619.

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 17 for UDP Unreplied Timeout and 117 for UDP Assured Timeout.

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

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT



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 http://forums.redflagdeals.com/please-s ... r-1993629/. For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.