Incapable de recevoir des appels mais capable d'en envoyer

[Tranthom]

comment as tu regler le probleme ?

j'ai un probleme similaire.

merci

[Tranthom]

J'ai un FPL sip sur un PAP2T

je suis capable de faire des appels sortant (il y a un delais enorme avant que ca sonne)

je ne recoit aucun appel de l'exterieur et ceux-ci tombent sur la boite vocale apres 30-45 seconde de "roaming/recherche"

CEPENDANT, mon test le plus recent avec un CELLULAIRE virgin+ les appels rentrent. mais tout les autre (rogers, videotron, autant cell que fillaire) ne rentrent pas.

autant sur le PAP2T que sur l'interface web FPL me disent que je suis REGISTERED (donc le user/sip password ainsi que l'addresse du serveur sont donc bonne.

le registration expire est a 300.

j'ai essayer les protocols G711u et G729a (presentement sur le G729a)

J'ai configurer un QoS sur mon routeur sur l'ip locale fixe de mon ATA ainsi qu'un DMZ .

je suis a cours d'idée.

le numero est edited

merci de me repondre rapidement

[Liptonbrisk]

comment as tu regler le probleme ?

j'ai un probleme similaire.

merci

Hello,

Sorry, I am not fluent in French.
Désolé, je ne parle pas couramment le français. J'utilise Google Traduction.


You posted in a thread that is from 2014. The problem is not the same.
Vous avez posté dans un fil qui date de 2014. Le problème n'est pas le même.

I've merged your posts here.
J'ai fusionné vos messages ici.

[Liptonbrisk]

le registration expire est a 300.

I see that you have posted in English before.

Register Expires should be 3600.
https://support.freephoneline.ca/hc/en- ... redentials

Your configuration guide is located at http://forum.fongo.com/viewtopic.php?f=15&t=16294. I would make the changes I listed below.

The problem with Rogers incoming calls is due to not getting true bridge mode from the router being used and/or from having SIP ALG enabled.
A number of people using Asus routers have had to disable SIP Passthrough, which is the setting for SIP ALG in Asus routers, to get incoming calls from Rogers working:
http://forum.fongo.com/viewtopic.php?f=8&t=20211#p79016
viewtopic.php?f=8&t=20182#p78916 (Fido is Rogers)


Rogers numbers have no issue calling me on my FPL accounts.

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


1. What brand and model modem are you using?

a) If you're using a modem/router combo, gateway, or hub issued by your ISP (and are NOT using your own additional separate router), contact your ISP to ask for assistance for disabling SIP ALG in the modem/router combo, gateway, or hub. Disable SIP ALG.


Here are two examples for disabling SIP ALG in gateways:


i) Hitron CGN3ACSMR and CODA-4582 series gateway modem/router combo from Rogers (and possibly other ISPs)


Open your 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.


ii) Arris XB6 from Rogers

Open your web browser, and login at 192.168. 0.1
Navigate to Advanced-->Options.
Uncheck the SIP box.
Click "Apply".

2. What brand and model router are you using?

If you are also using your own separate router in addition to the one supplied by your ISP, then you should be enabling bridge mode instead in the modem/router combo, gateway, or hub issued by your ISP.
For Bell Hubs, visit http://forums.redflagdeals.com/please-s ... r-1993629/. For Bell and Virgin Hubs, I find it's often simpler to perform PPPoE login in your own router (this is PPPoE Passthrough) and disable Wi-Fi in the hub. You will need the PPPoE Username and Password from Bell or Virgin.

For Rogers, visit https://www.rogers.com/customer/support ... ridgemodem.
Shaw users will have to call Shaw to enable bridge mode at the time of this post.


3. Afterwards, disable SIP ALG in your own separate router if you own one. Here is an example: https://www.obitalk.com/info/faq/sip-alg/disable-alg.
In Asus routers, SIP ALG is called SIP Passthrough. In Asuswrt-Merlin, the ALG is "+NAT Helper". Just make sure SIP Passthrough in Merlin does not show "+NAT Helper".


4. Check to see whether you've accidentally enabled Caller ID block on your ATA. Dial *68 to remove caller ID blocking on all outbound calls.
Don't use "Anonymous" for display name in your ATA. Don't use or accidentally dial *67.

This not related to your specific issue, but others have reported issues on outbound calls when *67 was enabled.

5. Login to your ATA. Select the admin menu. In your ATA, navigate to Voice-->Line (whichever you use for FPL)-->Supplementary Service Subscription-->Block CID Serv:
a) change to no
b) Click "submit all changes"

This not related to your specific issue, but others have reported issues on outbound calls with Block CID Serv enabled.

6. Navigate to Voice-->Line ( (whichever you use for FPL)-->Proxy and Registration-->Proxy

Use "voip4.freephoneline.ca:6060" without the quotation marks.

voip4.freephoneline.ca:6060 is used to bypass SIP ALG in routers. Refer to point B in the post below.


7. Navigate to Voice-->Line (whichever you use for FPL)-->SIP settings.
Specify a high random SIP port in your ATA between 30000 and 60000.
Change SIP Port to a random number between 30000 and 60000. Choose a number in that range.
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 avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potential corrupted NAT association that developed between your router and ATA.

Click the "Submit all changes" button.

8. Navigate to Voice-->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

e) click "Submit all changes" button

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

If people calling you are also using Linksys or Cisco ATAs, check to ensure they’re using those settings as well.

9. 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 "Submit all changes" button if changes were made

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

Click "Submit all changes" button if changes were made

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

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

11. Click https://www.freephoneline.ca/followMeSettings and login. Is Follow Me enabled? Ensure "Follow Me" is disabled while testing.

12. Check the registration status or "SIP status" after logging in at https://www.freephoneline.ca/showSipSettings

Please note that if "SIP User Agent" does not reflect a device you're using, someone else is using your Freephoneline VoIP unlock key.
Only one device or Line registration is permitted at any time per VoIP unlock key. Registration is a requirement for incoming calls but not for outgoing calls.


13. 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; wait a few minutes)-->ATA (wait for router to be fully up and running before turning on ATA). That's always proper device reboot order. Please reboot your devices now in that order.

14. Test incoming calls

[Liptonbrisk]

(Generic info)

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

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

Visit https://dh2i.com/kbs/kbs-2961448-unders ... -punching/.
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.

B) 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.

C) 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 D) 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_KeepAlivesExpires (SIP OPTIONS Keep Alive Interval) 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 (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 ... #p28059363.

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

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT


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