[Solved]LAN IP change/Cannet CGN3+OBi100+STUN=15 minute drop

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[Solved]LAN IP change/Cannet CGN3+OBi100+STUN=15 minute drop

Postby sincere » 05/11/2019

I bought the VOIP unlock and have used VOIP for long. But now I find the IP address of my VOIP device often changes and I sometimes I can't get calls or call out. I now use Cannet cable internet, with the CGN3 modem. I already read and did as the official article said: https://support.freephoneline.ca/hc/en- ... /212430746 But don't know if it is related with my device ip address.

Why does my device address often change between: http://192.168.0.12 and 192.168.0.13 ? I don't change its port on the cable moderm. How can I make the VOIP device address fixed at one address, say http://192.168.0.12? Thanks.
Last edited by Liptonbrisk on 04/21/2020, edited 4 times in total.
Reason: new issue added in title
sincere
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Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 05/12/2019

Assigning static IPs has nothing to with FPL. The device’s IP is being assigned dynamically by your router. That’s why the IP changes.
Contact the company or ISP that provided your router and ask them to help you assign a static LAN IP for your ATA or IP Phone if that’s what you want. That’s a router function. https://www.howtogeek.com/184310/ask-ht ... my-router/

Ensure SIP ALG is disabled in whatever modem/router combo your ISP gave you. Alternatively, use voip4.freephoneline.ca:6060 for the proxy server.

Proper device boot order is always modem—>router (wait for it to be fully up and running first)—>ATA (or IP Phone).

If you need configuration help, specify the brand and model of the VoIP device you’re using, or visit viewforum.php?f=15.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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SIP Device Name: OBi202
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ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 05/12/2019

Liptonbrisk wrote:Assigning static IPs has nothing to with FPL. The device’s IP is being assigned dynamically by your router. That’s why the IP changes.
Contact the company or ISP that provided your router and ask them to help you assign a static LAN IP for your ATA or IP Phone if that’s what you want. That’s a router function. https://www.howtogeek.com/184310/ask-ht ... my-router/

Ensure SIP ALG is disabled in whatever modem/router combo your ISP gave you. Alternatively, use voip4.freephoneline.ca:6060 for the proxy server.

Proper device boot order is always modem—>router (wait for it to be fully up and running first)—>ATA (or IP Phone).

If you need configuration help, specify the brand and model ATA you’re using, or visit viewforum.php?f=15.

Thanks. I don't mind the change of IP address of my ATA, but when it changes, I can't call out or in, I find. So I have to make it fixed. I will try to find a way.

I can't find the SiP ALG tab in my CGN3 modem. Why use voip4.... can solve the problem? Thanks?
sincere
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Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 05/12/2019

sincere wrote:I can't find the SiP ALG tab in my CGN3 modem.


That's because the option to disable SIP ALG in that modem/router combo has not been made available to CanNet users.
Consequently, that device (unless it's in bridge mode and is solely being used as a modem) is not good choice for SIP services in general.
Your Hitron CGN3 isn't just a modem. It's a modem/router combo or gateway. The router portion (SIP ALG is a router feature, as is assigning/leasing LAN IPs) of that device is what's causing problems.





Why use voip4....



See point B below (but if you really want to learn, read the whole post). SIP ALG features in routers typically listen to traffic on UDP port 5060 (and then corrupt/mangle SIP headers), which is the UDP port voip.freephoneline.ca and voip2.freephoneline.ca use, whereas voip4.freephoneline.ca uses UDP port 6060. Using voip4.freephoneline.ca:6060 circumvents SIP ALG features in routers. Using voip4.freephoneline.ca:6060 has nothing to do with Rogers, regardless of what you may have read. voip4.freephoneline.ca:6060 has everything to do with circumventing SIP ALG in any router, regardless of what company issued it.


(Generic info)
Typically, for VoIP SIP services, especially for Freephoneline, you want

A) 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.

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) < SIP OPTIONS 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 ... 2115672/2/

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.



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 (Asuswrt-Merlin is one exception), and stick whatever modem/router combo (typically garbage, especially for SIP services, like Freephoneline) 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.

Again, proper device boot order is always 1. modem—>2. router (wait for it to be fully up and running first)—>3. ATA (or IP Phone).
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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SIP Device Name: OBi202
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ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 05/13/2019

You are so knowledgable and nice. Thanks. I am ok with the voip.freephoneline.ca., not voip4.free.... Strange. But hope it can hold on. THanks again
sincere
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Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 05/13/2019

sincere wrote: I am ok with the voip.freephoneline.ca., not voip4.free....



You need to specify 6060 for the UDP/registration port when using voip4.freephoneline.ca.
In most ATAs, for the proxy server, you would just enter voip4.freephoneline.ca:6060
Otherwise, UDP port 5060 is used by default, which won't work with voip4.freephoneline.ca.

Then you need to reboot both your router (wait for it to be fully up and running first) followed by your ATA to help ensure NAT associations aren't corrupted.



1) Always check in your ATA, IP Phone, or SIP app to ensure you're registered. In some (Obihai) ATAs, this information is found under "SIP Status". In other ATAs, it may be found under "Registration State."

2) Also always check registration status after logging in at https://www.freephoneline.ca/showSipSettings. SIP status needs to show "connected." If SIP User Agent is anything that you don't recognize, you've been hacked.

Registration is required for incoming calls. It is not required for outgoing calls.


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, with Acrobits Groundwire, for example, on a smartphone (there is no way to do this with the Freephoneline desktop application; the Freephoneline desktop application doesn't offer to option to not register). 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 https://www.freephoneline.ca/showSipSettings. 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.

Each time you reboot the ATA, it's attempting to register with Freephoneline again. If you attempt more than 10 registrations in 5 minutes (this is why the registration interval is important),
you may end up being temporarily IP banned by the specific FPL server the ATA (and/or desktop app) was sending
registration requests to. If you're temporarily IP banned, you could then try switching ProxyServer 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 buggy 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)."

That's why the registration interval of 3600 seconds and failed retry timer of 120s in the ATA are important.
If the ATA loses registration for any reason, incoming calls won't work on it.


What is the brand and model of the VoIP device that you're using?
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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SIP Device Name: OBi202
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ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 05/13/2019

Liptonbrisk wrote:
sincere wrote: I am ok with the voip.freephoneline.ca., not voip4.free....

What is the brand and model of the VoIP device that you're using?


Thanks. I have been using the Obihai 100 for many years, and ok.

Now, I configure like below, and it's ok now, though not sure next time:
ProxyServer voip.freephoneline.ca
ProxyServerPort 5060
ProxyServerTransport UDP
RegistrarServer voip.freephoneline.ca
RegistrarServerPort 5060

Or do you think it's better to be below?
ProxyServer voip4.freephoneline.ca
ProxyServerPort 6060
ProxyServerTransport UDP
RegistrarServer voip4.freephoneline.ca
RegistrarServerPort 6060

Can the Proxy Server and RegistrarServer be different like this?
ProxyServer voip.freephoneline.ca
ProxyServerPort 5060
ProxyServerTransport UDP
RegistrarServer voip4.freephoneline.ca
RegistrarServerPort 6060

Tks.
sincere
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Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 05/13/2019

RegistrarServer shouldn't be entered at all (should be blank), and, consequently, RegistrarServerPort can be ignored despite the PDF guide listing 6060 for it (doesn't matter).

If you specify "voip4.freephoneline.ca:6060" (without the quotation marks) for ProxyServer, you don't even need to change ProxyServerPort (can be left at default). The UDP port specified after the colon for ProxyServer takes precedence the last time I tested an OBi100.


Or do you think it's better to be below?
ProxyServer voip4.freephoneline.ca
ProxyServerPort 6060
ProxyServerTransport UDP


That's fine.

RegistrarServer voip4.freephoneline.ca


There's no need to specify RegistrarServer at all. Blank that out (which should be the same as having the default box checked).

RegistrarServerPort 6060


There should be no need to specify anything here if RegistrarServer is blank (both the setting and its port aren't needed when configuring FPL), but if you are going to specify something then enter 6060 when using voip4.freephoneline.ca.

ProxyServer voip.freephoneline.ca
ProxyServerPort 5060
ProxyServerTransport UDP
RegistrarServer voip.freephoneline.ca
RegistrarServerPort 5060


Don't use that for SIP ALG problems, which is a problem you may be experiencing if you can't disable SIP ALG in your modem/router combo.

Can the Proxy Server and RegistrarServer be different like this?
ProxyServer voip.freephoneline.ca
ProxyServerPort 5060
ProxyServerTransport UDP
RegistrarServer voip4.freephoneline.ca
RegistrarServerPort 6060


Definitely don't do that.

Follow the steps in the order listed:


1. If you used the Obitalk web portal (http://www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. Never use both methods.


2. Use this PDF guide: download/file.php?id=1704. Again, that guide really has less to do with Rogers than it has to do with using a router with SIP ALG forced on (applies to you). Ensure that you're using voip4.freephoneline.ca:6060 for the proxy server.

3. Ensure that you (this is missing from the PDF guide)
Navigate to Service Providers-->ITSP Profile (FPL)-->SIP
RegisterRetryInterval should be 120 seconds.
RegisterRetryInterval is the Failed Registration Re-Try Interval listed at https://support.freephoneline.ca/hc/en- ... /212430746

4. Navigate to Voice Services-->SP(number that you're using for FPL) Service-->X_UserAgentPort
Change this local SIP (UDP) port to something random between 30000 and 60000. Just pick a number in that range.
(Changing this helps thwart SIP scanners/hackers and may also help with SIP ALG issues as well involving certain routers). It's a variation on what Mango recommends (same thing, basically).

5. "Also from within Voice Services >> SP(FPL) Service, set X_InboundCallRoute to {>1xxxxxxxxxx:ph}
If SIP scanners do find you, this will cause your OBihai ATA to reject the call. (Note: replace 1xxxxxxxxxx with your FPL number.)" -- Mango

6. Submit/save.

7. Reboot modem/router combo. Wait for it to be fully up and running. Then reboot your ATA.

8. Test with an incoming call from a regular cellphone or landline.

If the person speaking to you is using a VoIP or SIP service provider, the problem could be on the other end of the call.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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Posts: 1247
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SIP Device Name: OBi202
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ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 05/13/2019

Tks. I don't use its web portal, but configure using the IP address. I don't know if I have the SIP ALG issue or not, but I still do as you said to use voip4. It is ok now. Thanks.

You are very knowledgeable. Do you work for FPL, teacher?
sincere
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Posts: 25
Joined: 05/11/2019

Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 05/13/2019

sincere wrote: It is ok now. Thanks.


You're very welcome!

You are very knowledgeable. Do you work for FPL, teacher?


Thank you for the compliment. No, I don't work for FPL. I believe all the moderators here are volunteers.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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Posts: 1247
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 05/13/2019

Liptonbrisk wrote:
sincere wrote: It is ok now. Thanks.


You're very welcome!

You are very knowledgeable. Do you work for FPL, teacher?


Thank you for the compliment. No, I don't work for FPL. I believe all the moderators here are volunteers.


Nice. I also volunteer a lot in the community. Together we devote love and knowledge to make it a better world. Thanks once again, sir.
sincere
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Joined: 05/11/2019

Caller can't hear me

Postby sincere » 04/13/2020

[quote="Liptonbrisk"][quote="goldenmeadow"]

I still have the one-way voice problem that I can't hear the caller, though he can hear me.

I use the Hitron CGN3 modem. It was ok before, but recent months failed. I tried your suggestion, but can't find "SIP ALG" under the "Basic" tab. Any clue? Seems you are very knowledgeable, so would you please help too? Thanks.
sincere
Quiet One
 
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Joined: 05/11/2019

Re: Caller can't hear me

Postby Liptonbrisk » 04/13/2020

I merged your post into your original thread since it makes it easier for me to see what brand and model devices you're using (easier for me to troubleshoot with that information).

sincere wrote:I use the Hitron CGN3 modem. It was ok before, but recent months failed. I tried your suggestion, but can't find "SIP ALG" under the "Basic" tab.


viewtopic.php?f=8&t=19589#p76453
That's because the option to disable SIP ALG in that modem/router combo has not been made available to CanNet users.
Consequently, that device (unless it's in bridge mode and is solely being used as a modem) is not good choice for SIP services in general.
Your Hitron CGN3 isn't just a modem. It's a modem/router combo or gateway. The router portion (SIP ALG is a router feature, as is assigning/leasing LAN IPs) of that device is what's causing problems.

Follow the steps below, carefully, step by step:

1. If you used the Obitalk web portal (http://www.obitalk.com) to configure your ATA, keep in mind that you must continue using it to configure your ATA. Otherwise whatever settings you change will eventually be overwritten by what you previously entered at obitalk.com anyway. If you wish to disable this behaviour, dial ***1. Enter that IP address into a web browser. Navigate to System Management-->OBiTalk Provisioning-->select Disabled for the method. Save. Reboot ATA. Now obitalk.com won't overwrite whatever changes you make via the device's interface (via IP address).

Pick one method (obitalk.com) or the other (IP address of device) for changing device settings. Never use both methods.


2. Use this PDF guide to configure your service: click download/file.php?id=1704. Again, that guide really has less to do with Rogers than it has to do with using a router with SIP ALG forced on (applies to you). Use voip4.freephoneline,ca:6060 for the proxy server (as outlined in the guide). The original thread can be viewed at viewtopic.php?f=15&t=16196.

3. Ensure that you (this is missing from the PDF guide)
Navigate to Service Providers-->ITSP Profile (FPL)-->SIP
RegisterRetryInterval should be 120 seconds.
RegisterRetryInterval is the Failed Registration Re-Try Interval listed at https://support.freephoneline.ca/hc/en- ... /212430746

4. Navigate to Voice Services-->SP(number that you're using for FPL) Service-->X_UserAgentPort
Change this local SIP (UDP) port to something random between 30000 and 60000. Just pick a new random number in that range.
(Changing this helps thwart SIP scanners/hackers and may also help with SIP ALG issues as well involving certain routers).

5. "Also from within Voice Services >> SP(FPL) Service, set X_InboundCallRoute to {>1xxxxxxxxxx:ph}
If SIP scanners do find you, this will cause your OBihai ATA to reject the call. (Note: replace 1xxxxxxxxxx with your FPL number.)" -- Mango
For this step, login at https://www.freephoneline.ca/showSipSettings. Pay attention to your SIP Username, which is what Mango wants you to use.

6. Submit/save.

7. Reboot modem/router (CGN3) combo. Wait for it to be fully up and running normally first. Then reboot your ATA.

8. Test with an incoming call from a regular cellphone or landline.

If the person speaking to you is using a VoIP or SIP service provider, the problem could be with the user's configuration on the other end of the call.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk
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Posts: 1247
Joined: 04/26/2010
SIP Device Name: OBi202
Firmware Version: 3.2.2 (Build:5921EX)
ISP Name: Cable
Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: Caller can't hear me

Postby sincere » 04/13/2020

[quote="Liptonbrisk"]I merged your post into your original thread since it makes it easier for me to see what brand and model devices you're using (easier for me to troubleshoot with that information).

Thank you, Sir! I have tried your suggestions, and used voip4.freephoneline.ca, 6060, etc, for long. But still can't hear the other party. Can it be the modem combo's problem? If so, as you implied, I may try to use the CGN3 as a modem only and use my old router. But would you pls educate me a bit on how to disable the router part of the CGN3, and then how to configure the modem and router? Thanks.
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Re: Caller can't hear me

Postby Liptonbrisk » 04/13/2020

sincere wrote:
Thank you, Sir! I have tried your suggestions, and used voip4.freephoneline.ca, 6060, etc, for long.


Did you change to a new random X_UserAgentPort just now, and then reboot CGN3 followed by power cycling or rebooting your ATA?


But still can't hear the other party.


Is your other party also using a SIP service provider, such as FPL? If so, the problem could be on their end.

Can it be the modem combo's problem?


It could be, but using voip4.freephoneline.ca:6060 and choosing a new random X_UserAgentPort (between 30000 and 60000) should bypass SIP ALG in the CGN3.

If so, as you implied, I may try to use the CGN3 as a modem only and use my old router. But would you pls educate me a bit on how to disable the router part of the CGN3, and then how to configure the modem and router? Thanks.


Hopefully, this works for you: https://www.rogers.com/customer/support ... emode-cgn3
If it doesn't, you may need to contact Cannet for bridge mode instructions.
I would, after enabling bridge mode, connect your ATA directly to the modem, briefly, for testing purposes.
If that works (after enabling bridge mode), then the problem involves something in the router portion of the CGN3.

If you use your own router, you may then need to check for SIP ALG in your own router if you continue to have one-way audio issues afterwards.

Other things to check include ensuring you haven't accidentally muted audio on your phone (or lowered volume) and checking phone cables. It's also possible the phone is defective (try another phone).

If none of that helps, then, unfortunately, you may have to port forward your RTP (UDP) port range 16660-16798 from your router to your ATA. For reference, that range can be found under ITSP Profile (FPL)-->RTP in your ATA. Then look at LocalPortMin and LocalPortMax. RTP packets need to reach your ATA 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 ATA. 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 ATA.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Re: Caller can't hear me

Postby sincere » 04/13/2020

[quote="Liptonbrisk"]

Thanks for your time.
1) Yes, I have been using 30000 in the random X_UserAgentPort, though not sure what it means, :-) Just now, I tried 40000 too, to no avail, though.

2) I called my own mobile phone to test, so that is not the other party using FPL too. :-)

3) Seems it's too troublesome to bridge the CGN3 to an old router, I'd better not do it now. Yes, I saw you mentioned using voip4.freephoneline.ca:6060 and choosing a new random X_UserAgentPort 30000 should bypass SIP ALG in the CGN3, so the modem problem doesn't really matter, though it doesn't work out for me. Can it be the Obihai 100 being too old? But I don't really think so. I don't want to change it just to find it doesn't matter either in the end, from many other clients' posting of the similar problems.

4) Your last para of port forwarding may be so advanced for me that I can hardly understand. Sorrry to be dumb.

5) As I can still use the soft phone (the FPL software) on my computer, perhaps I can just use the software to call, instead of the more convenient hardware phone. Hope for a day when the problem disappear.

Thanks again, sir!
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Re: Caller can't hear me

Postby Liptonbrisk » 04/13/2020

sincere wrote:
3) Seems it's too troublesome to bridge the CGN3 to an old router,


I would enable bridge mode in the CGN3, briefly, just for testing purposes. Connect your ATA directly to the CGN3 after bridging. Try to make a call. If the problem disappears, then the issue is the CGN3.
I feel that's a worthwhile troubleshooting step.



Can it be the Obihai 100 being too old?


I doubt it. But it wouldn't hurt to try another phone just to rule the phone out as being the cause.

Your last para of port forwarding may be so advanced for me that I can hardly understand. Sorrry to be dumb.


There's no need to apologize, and you're not dumb. This is a just a new concept for you. And we'll cross the port forwarding road if we need to. Right now, I'd just like to know if the router portion of the CGN3 is the problem.

As I can still use the soft phone (the FPL software) on my computer, perhaps I can just use the software to call


Do you mean the FPL desktop app doesn't exhibit the same issue for you?
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Re: Caller can't hear me

Postby sincere » 04/13/2020

[quote="Liptonbrisk"][quote="sincere"]

I tried to bridge based on your that Roges link, but I couldn't even log onto the CGN modem's interface, nor could make a call. I had to reset the modem to go back to before, without bridging any more.

The Obihai 100 is the ATA box, not the phone. The phone is perfect with both mic and speaker/handset tested.

I don't understand the port forwarding concept, or how to do it yet.

Yes, the FPL desktop application is ok without any problem. I just can't use the hardware phone with the ATA box to hear a caller, while it was ok months ago.
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Re: Caller can't hear me

Postby Liptonbrisk » 04/13/2020

sincere wrote:
I tried to bridge based on your that Roges link, but I couldn't even log onto the CGN modem's interface


You're going to have to get in contact with Cannet and ask about placing that device in bridge mode.

The Obihai 100 is the ATA box


I am aware an OBi100 is an ATA. I own an OBi202.

not the phone. The phone is perfect with both mic and speaker/handset tested.


As long as you're positive the phone is working properly


Yes, the FPL desktop application is ok without any problem.


That's pretty bizarre. If the desktop app works while the computer is connected to the CGN3, your ATA should work as well.

1) Ensure any computer running the FPL app is turned off (or ensure the app is not running at all).

2) What happens if you use voip2.freephoneline.ca:5060 for the proxy server?
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Re: Caller can't hear me

Postby sincere » 04/13/2020

Liptonbrisk wrote:
That's pretty bizarre. If the desktop app works while the computer is connected to the CGN3, your ATA should work as well.

1) Ensure any computer running the FPL app is turned off (or ensure the app is not running at all).

2) What happens if you use voip2.freephoneline.ca:5060 for the proxy server?


Thanks for your patience. I thought the software FPL is different from the hard FPL. The fact that the soft phone can work on computer doesn't mean the hard phone with the ATA will work, right?

I don't usually open the soft phone at all. I just tried again with voip2.freephoneline.ca:5060, (without the soft FPL of course) but still can't hear anything from the other side.
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Re: Caller can't hear me

Postby Liptonbrisk » 04/14/2020

sincere wrote:
Thanks for your patience. I thought the software FPL is different from the hard FPL. The fact that the soft phone can work on computer doesn't mean the hard phone with the ATA will work, right?


They should be connecting to the same servers (voip.freephoneline.ca or voip4.freephoneline.ca:6060). My experience has been, with one or two exceptions, that if the FPL desktop app works, an ATA should just as easily on the user's LAN. In fact, the FPL desktop app is generally used to test whether the service will work before the customer pays for a VoIP unlock key. Usually, the FPL desktop app is more difficult to get working properly because its SIP settings are less customizable than an ATA's.

RTP stands for Real-time Transport Protocol, which defines the audio packets, or audio stream. The stream is transported to your ATA via UDP ports 16660 to 16798 as defined in your ATA under under ITSP Profile (FPL)-->RTP-->from LocalPortMin to LocalPortMax. RTP packets need to reach your ATA in order for you hear incoming audio.

I see four possible causes for this issue: 1. SIP ALG in the CGN3 is modifying (mangling) SIP headers so that RTP traffic isn't reaching your WAN IP. SIP ALG features in routers typically listen to or monitor traffic on UDP ports 5060 and sometimes 5061. So, if you're using voip4.freephoneline.ca:6060 and also using a random X_UserAgentPort, SIP ALG should be bypassed, at least, in theory. 2. CGN3's NAT firewall is blocking incoming RTP packets for the ATA for some reason. 3. For some reason a SIP header in your ATA is now specifying a private LAN IP (the one you use to log into your ATA) instead of your public WAN IP (your actual public WAN IP address) to send RTP traffic, and in turn, RTP traffic isn't even reaching your CGN3. 4. Your phone, your phone cord(s), or your ATA is defective.

I'm still interested in whether your ATA works (switch back to voip4.freephoneline.ca:6060, please) when the connected directly to the CGN3 in bridge mode.

Also, while I'm usually hesitant to suggest this (because I'm uncertain there would be any improvement), perhaps factory resetting the ATA and reconfiguring it from scratch might help.

COVID-19, makes this next suggestion impossible, but it would be interesting to see whether your ATA works at a friend's place that doesn't use Cannet for an ISP.

When you use voip4.freephoneline.ca:6060 and make a test call to 416-342-9562 (http://thetestcall.blogspot.com/), can you hear anything? I'm wondering if the problem involves all phone calls or just a single number.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 04/14/2020

You could submit a ticket at https://support.fongo.com/hc/en-us/requests/new and ask if they'll do a "force registration" for you. Explain your lack of incoming audio issue in the ticket and mention that after server migration you started experiencing this problem. They don't offer free technical support, but they might make an exception due to the server migration. However, if the problem is the CGN3, a forced registration isn't going to help.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Computer OS: Windows 10 x64 Pro
Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby Liptonbrisk » 04/14/2020

I also have something else for you to try.

In your ATA, navigate to

ITSP Profile (FPL)-->SIP

Does 1) disabling X_UseRport make a difference? You will need to uncheck the default box on the right, and then uncheck the box under the value column. Reboot ATA. Test.

Does 2) disabling both X_UseRport and X_DiscoverPublicAddress make a difference? Reboot ATA after changing settings. Test.

If neither 1 nor 2 helps, re-enable both settings.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Router: Asuswrt-Merlin

Re: why does my VOIP IP address change often?

Postby sincere » 04/14/2020

Liptonbrisk wrote:I also have something else for you to try.

In your ATA, navigate to

ITSP Profile (FPL)-->SIP

Does 1) disabling X_UseRport make a difference? You will need to uncheck the default box on the right, and then uncheck the box under the value column. Reboot ATA. Test.

Does 2) disabling both X_UseRport and X_DiscoverPublicAddress make a difference? Reboot ATA after changing settings. Test.

If neither 1 nor 2 helps, re-enable both settings.


Thanks. I tried both of your suggestions, but didn't work out either. Thanks anyway.
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Re: why does my VOIP IP address change often?

Postby sincere » 04/14/2020

Liptonbrisk wrote:You could submit a ticket at https://support.fongo.com/hc/en-us/requests/new and ask if they'll do a "force registration" for you. Explain your lack of incoming audio issue in the ticket and mention that after server migration you started experiencing this problem. They don't offer free technical support, but they might make an exception due to the server migration. However, if the problem is the CGN3, a forced registration isn't going to help.


OK, I can try to submit a ticket for that, though not sure what "force registration" is. Thanks.
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