[Resolved]Can only call Fongo numbers using SPA112

[vlasov01]

Hi,

I have a similar issue with making outgoing calls, which started today - August 27th. I'm able to get incoming calls. And I can make outgoing calls only to Fongo numbers.
I'm using Cisco/SPA112-1.4.1_SR5 with freephoneline. I've tried the voip2 sever and have the same issue. I never had this problem before. Reboot didn't help. No errors in the ATA logs. And there is no incidents reported today on https://status.fongo.com/ . It doesn't seems to be a network issue on my side as I can call other Fongo numbers.

[Liptonbrisk]

The thread you posted in applied to some calling card service. FPL to Fongo calls are SIP URI calls (not the same as making a call to PSTN), which, if they work does not necessarily prove your issue is not a LAN issue.
I don't have problems calling anything at the moment.

Are you getting a fast busy on outgoing calls?

If so, 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.

Please follow the steps, step by step, down the list.

1. In your ATA, navigate to Voice-->Line (whichever you use for FPL)-->Supplementary Service Subscription-->Block CID Serv: change to no
b) Click "submit"




2) What brand and model modem are you using?
3) What brand and model router are you using?
4) Describe what happens when outgoing calls fail.
5) When you login at https://www.freephoneline.ca/doGetCallLogs, select the month, and click "submit", are the failed outgoing calls listed?
a) If so, what does the disconnect reason state?


6) Login to your ATA. Specify a high random SIP port in your ATA between 30000 and 60000.
Navigate to Voice-->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.

7. This step is unrelated to your problem. Regardless, in the ATA, 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

d) 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.

8. 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

9. This step is not related to your issue.
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- ... redentials

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

10. 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.

[Liptonbrisk]

This appears to be resolved: https://forums.redflagdeals.com/freepho ... #p33036045