Fongo Community Forums

Thanks for letting me know. I would give voip4.freephoneline.ca:6060 a try. If it doesn't work out you can switch back to voip2.freephoneline.ca. I'm sure we can get this solved for you.

Please verify that your device is set up as per http://forum.fongo.com/viewtopic.php?f=15&t=16294 . In particular, be sure to set Register Expires to 3600. I would like to make some changes from the official guide. On the SIP tab, please set Reg Retry Intvl to 120. Since you are a Rogers custome...

mmcgarri, did you solve your problem yet? If not, please try to set the following: ADVANCED SETTINGS STUN server is: stun.3cx.com FXS PORT2 NAT Traversal (STUN): Yes If that solves the problem, remove the port range forwards that I suggested earlier and verify that things are still working. If that ...

with a Rogers cable modem and a Linksys WRT54GS upgraded to DD-WRT v24-sp2 firmware. Is your DD-WRT router receiving a public IP address from Rogers (i.e. modem is in bridge mode)? And if not, are you using the special Rogers proxy? I bought the second VIP key today and try to set it up with my new...

Dang - I forgot that one!

1) Has this ever worked or is this with a brand new new unlock key? 2) Are you able to make outgoing calls? 3) On the status page of your ATA, does the ATA show that it is registered? 4) If you change your ATA's SIP Port to a random number between 20000 and 65535, does that solve the problem? 5) Did...

If you do receive that result, it means your firewall is either not configured properly or insecure. Having port 5060 open leaves your equipment vulnerable to hacking. If your VoIP service won't work any other way, then you should set your port number to a high/random number between 20000 and 65535.

If changing the device's SIP port causes it to register, your router's NAT connection became corrupted. You created a new NAT connection and allowed the old one to time out. There are some settings you can look at to try to mitigate this problem in the future. You may wish to disable your router's S...

Since your DSL provider is TekSavvy, you should expect the worst case scenario. I would not even consider the possibility of TekSavvy orchestrating a seamless switchover.

The rate centre is called Kitchener-Waterloo and it is portable. If you wish, enter your area code and the next three digits of your phone number at https://www.telcodata.us/search-area-code-exchange-detail and verify that you are in fact in the Kitchener-Waterloo rate centre. (It displays as "...

On the FXS PORT1 page of your ATA, please verify that NAT Traversal is set to Keep-Alive. From your description, I believe it may be configured to use STUN. If it was already set to Keep-Alive, leave it that way, and please try the following, if you have not already done so: Set up a static IP for y...

Change your SIP Port to a random number between 20000 and 65535 to test if the problem is caused by your router.

On the DD WRT setup, there are a check box to enable to disable QoS. I have check the box to enable. This by itself will not do anything. You also need to configure the uplink and downlink speeds of your internet connection, minus 20%. Then you need to assign your OBi's MAC address high priority.

QOS is a waste of time and bandwidth Does your VoIP sound as good as or better than a landline 100% of the time? If yes, then your statement I quoted above is correct. However, the vast majority of VoIP users require properly configured QoS. Limiting your individual bandwidth use is a great idea, b...

That's good to hear. 0.2Mbps is indeed low, and you may want to talk with your ISP as a matter of course. If you had faster speeds, there would be more room for general internet use, while a VoIP call was in progress.

Your problem is caused by some other device on your network using too much bandwidth. You can solve the problem by setting up QoS on your router. You'll need to configure it with the maximum upload speed of your internet connection, less 20%. I've never done this with your router, but it's in the us...

If you disable the SIP ALG on your router, does that solve the problem?

http://www.3cx.com/blog/docs/disable-si ... n-thomson/

You'll need to reboot your router after making the change.

Hi JR, What is the model number of ATA and router that you are using? What symptoms are you having that you intend to correct? Just a note with regards to the post you quoted. DMZ should never be used as it presents a serious security risk. Also, the SIP Port should not be forwarded unless things wo...

SIP TCP is the wrong setting. I believe on your ATA it is called SIP Port. If that and your own router does not help, you may want to ask FPL for support as you seem to have exhausted the possibilities of a configuration issue causing your problem.

Try the following: 1) Disable your router's SIP ALG, if there is a setting for this. 2) If such a setting does not exist, set your SIP Server to voip4.freephoneline.ca:6060, and also your SIP Port to 6060. 3) If that does not solve the problem, use your router to set up a UDP port forward of your AT...

This problem is best solved with multiple approaches: 1) If your router is configured to forward your ATA's SIP port, you should remove this port forward unless it won't work any other way. If your ATA is in your router's DMZ, this should be immediately removed. 2) If your router has an option for &...

When you use the Rogers proxy, don't forget to also change your SIP Port to 6060. If that doesn't work, set Symmetric RTP to Yes and test. If THAT doesn't work, return Symmetric RTP to the defualt of No, make a note of your ATA's RTP Port Min and RTP Port Max (by default 16384-16482), and use your r...

my alarm system wire is unplug from my phone line. i have done like 10-15 call without any fax sound :) Booyah! :mrgreen: All someone would need to do is call your house before they broke in and the dialler would not be able to get out. Also if you are on the phone when you need to, say, hit a pani...

It's unfortunate that the VoIPDIY site seems to have a lot of incorrect information :( If you still have one-way audio after setting the OBi up as per Fongo Support's suggestion, this guide may help: http://www.3cx.com/blog/voip-howto/pfsense-firewall/ You will need to know your OBi's RTP port range...

Navigate to your PAP2T's admin/advanced configuration. On the Regional tab, remove the code beside Cfwd All Act Code which by default is *72. If you wish, you can also remove Cfwd Busy Act Code and Cfwd No Ans Act Code, if you do not intend to use these features. Let me know if that solves the probl...