No ring tone when I Dial

[pinipogue]

Hi,
I`m fairly new in this VIOP call. I have a linksys SPA2102 and a Assus RT-N66 router. Sometime when I place a call a get no ring tone but the call is being place and the receiver answer the call normally. But for me I ear nothing. I' m not sure if it's not a port setting in my router or a config in the linksys.
Any Ideal ....

Thanks,

[Funkytown]

Hi,
I`m fairly new in this VIOP call. I have a linksys SPA2102 and a Assus RT-N66 router. Sometime when I place a call a get no ring tone but the call is being place and the receiver answer the call normally. But for me I ear nothing. I' m not sure if it's not a port setting in my router or a config in the linksys.
Any Ideal ....

Thanks,

Try unplugging everything and then wait a minute..re-plugging everything back one at a time.. modem..router..ata.

Also if you have a computer plugged into the SPA2102 unplug it now and plug it directly into the ASUS allowing it to re route everything properly from now on.

If the problem persists with the SPA2102 then you can always fix a static ip on the Spa2102 and place DMZ for the Spa2102 on the ASus.

[pinipogue]

Thanks for the quick respond. But my ata is already plug into the router and nothing into the ata. Could you please tell me witch port should be forward in the router.

Thanks

[Funkytown]

Thanks for the quick respond.

Your welcome.

But my ata is already plug into the router and nothing into the ata.

Great to hear

Could you please tell me witch port should be forward in the router.

Thanks

Sure, you can forward the default ports that are required 5060-5061(UCP), & 6060-6061(UCP) & 13000-13001(UCP).

What happens if the above doesn't work for you and you happen to continue to have constant problems with your phone lines using the same ports i described above then you will need to follow below.

Go into phone line 1 and make sure phone line 2 is not using the same ports, for example line 1 is using 5060 automatically uses 5060-5061 so on line 2 phone you need to change this port to 5070 and it automatically reserves 5070-5071 and so on, now continue to check other ports and make sure none are the same on Line 1 & line 2 and if they are make sure to change each of them and that they are totally different by 5 to 10 digits higher.

Take a notepad save all the ports in it so that everything is easy when it comes to configuring the port forwarding for both lines.

Then inside your ATA Voip adapter i recommend to make it a static ip so the internal ip of the voip adapter never changes so that your router can give the go for the sip device and tell it everything is good. Especially saving you all that wasted time from having them headaches of going back and forth into the device everyday.

Important also please make sure on your Sip device all ports remain the same by enabling this setting for each line.

Forward all the ports inside your asus so for example Line 1 , Line 2
Line 1 5060-5061(UCP), & 6060-6061(UCP) & 13000-13001(UCP)
Line 2 5070-5071(UCP), & 6070-6071(UCP) & 13010-13011(UCP)

[pinipogue]

Thanks again.
Do we have to enable the QOS on the router for a better communication.

[Funkytown]

Thanks again.
Do we have to enable the QOS on the router for a better communication.

I wouldn't worry about Qos as this feature only prevents applications or devices hogging bandwidth.

But I do have a question for you, have you gone over everything as explained in my above post to change the ports on Line 1 & Line 2 in your ATA Device and also forwarded those ports inside your Asus?

[hanke]

The SPA2102 has a router built into it with 2 phone line ability. I would be hooking the SPA2102 directly to the modem on DHCP mode, configure the ATA as required by FPL. The SPA2102 will get its own IP from the service provider, then hook up your other router into the ATA to distribute your interent/wireless through out your home as needed. The ATA in this postion would not require any special settings or opening of ports, it will work on its own with the basic FPL settings. So if a reset of the ATA was done, use the handset to get it into DHCP mode with the commands, then connect to it with your computer and configure it accordingly.

Good luck..

PS - If you do, do a reset you will have to ensure the suggested Dial Plan is used from FPL settings as well as all the other required information.

[pinipogue]

Thank and it's working OK. One last thing... I hope
Because some place say use outbound proxy and other not... Could you look at thoses setting and let me know what you think
Proxy and Registration (Should I use the outbound proxy) my anwser in ()

Outbound Proxy: (viop.freephoneline.ca)
Use Outbound Proxy: yes /no (YES) Use OB Proxy In Dialog: yes no (YES)
Register: yes/no (YES) Make Call Without Reg: yes no (YES)
Register Expires: (3600) Ans Call Without Reg: yes no (YES)
Use DNS SRV: yes/no (NO) DNS SRV Auto Prefix: yesno (NO)
Proxy Fallback Intvl: (3600) Proxy Redundancy Method: (Normal)

Best Regards,
Pinipogue

[Fongo Support]

Thank and it's working OK. One last thing... I hope
Because some place say use outbound proxy and other not... Could you look at thoses setting and let me know what you think
Proxy and Registration (Should I use the outbound proxy) my anwser in ()

Outbound Proxy: (viop.freephoneline.ca)
Use Outbound Proxy: yes /no (YES) Use OB Proxy In Dialog: yes no (YES)
Register: yes/no (YES) Make Call Without Reg: yes no (YES)
Register Expires: (3600) Ans Call Without Reg: yes no (YES)
Use DNS SRV: yes/no (NO) DNS SRV Auto Prefix: yesno (NO)
Proxy Fallback Intvl: (3600) Proxy Redundancy Method: (Normal)

Best Regards,
Pinipogue

Hi there,

Please see below:

Outbound Proxy: (blank)
Use Outbound Proxy: yes /no (NO) Use OB Proxy In Dialog: yes no (YES)
Register: yes/no (YES) Make Call Without Reg: yes no (NO)
Register Expires: (3600) Ans Call Without Reg: yes no (NO)
Use DNS SRV: yes/no (NO) DNS SRV Auto Prefix: yesno (NO)
Proxy Fallback Intvl: (3600) Proxy Redundancy Method: (Normal)


reference documentation below:

http://forum.fongo.com/viewtopic.php?f=15&t=310

Thanks,