"4 ) If you do have enough World Credits, my guess would be the issue being either due to a lack of registration in your ATA (test with an incoming call) or a SIP ALG issue in your router. There's a RE-INVITE that occurs at the 15 minute mark, and if there's no ACK (acknowledgement) received, the call will drop. However, it doesn't make sense that all local calls work fine if the problem is due to the Re-INVITE. If you want to pursue this possibility (albeit, unlikely) further, then I need the information requested from viewtopic.php?f=8&t=20199."
I have this issue where my calls are disconnecting at the 15 minute mark. I did just install a new router so what you've said make sense. However, I cannot find a SIP ALG setting in my router.
Would it go by another name?
I do have my ATA in the DMZ.
thanks
[Resolved] outgoing calls drop after 15 minutes with SPA112
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- Just Passing Thru
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- Technical Support
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- ISP Name: FTTH
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: calls drop after the 15 minute Re-INVITE
As your issue doesn't appear to involve World Credits, I moved your post into its own separate thread.marcaccioc wrote: I have this issue where my calls are disconnecting at the 15 minute mark. I did just install a new router so what you've said make sense. However, I cannot find a SIP ALG setting in my router.
Would it go by another name?
I do have my ATA in the DMZ.
A.Using DMZ is a huge security risk. It leaves your ATA completely unprotected. Don't use DMZ unless you have no other choice. Similarly, don't port forward unless you have no other choice.
B. I need the information requested from viewtopic.php?f=8&t=20199.
Please provide the information from #1, #2, #3, and #5 requested from that link.
In order for calls to not drop at the 15 minute Re-INVITE, the ATA should be registered and not fighting for registration with another device, app, or line (only one line on an ATA may be registered per FPL account at any given time). Only one registration per FPL account is permitted at any time. SIP ALG (also called SIP Passthrough in Asus routers) should be disabled (refer to B from viewtopic.php?f=8&t=20199#p78976). Any modem/router combo, hub, or gateway should be in bridge mode if you're also using your own additional router.
Registration timers are also important, especially the 3600 second registration interval.
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/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at http://forum.fongo.com/viewforum.php?f=15.
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- Just Passing Thru
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Re: calls drop immediately after the 15 minute Re-INVITE
thank you.
I've determined the call drop only happens on outgoing calls. Incoming calls are fine.
March 8th is the last time an outgoing call lasted over 15 minutes.
March 10th is the first time an outgoing call was disconnected at 15 minutes.
1 - SmartRG SR516ac is my modem/router.
2 - I've now plugged the SPA112 directly into the SmartRG to take the other router out of the equation.
3 - Cisco SPA112
4 - voip2.freephoneline.ca and I've tried voip.freephoneline.ca
5 - Registration Status = Registered SIP status = Connected
I've determined the call drop only happens on outgoing calls. Incoming calls are fine.
March 8th is the last time an outgoing call lasted over 15 minutes.
March 10th is the first time an outgoing call was disconnected at 15 minutes.
1 - SmartRG SR516ac is my modem/router.
2 - I've now plugged the SPA112 directly into the SmartRG to take the other router out of the equation.
3 - Cisco SPA112
4 - voip2.freephoneline.ca and I've tried voip.freephoneline.ca
5 - Registration Status = Registered SIP status = Connected
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- Technical Support
- Posts: 3069
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: FTTH
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: calls drop immediately after the 15 minute Re-INVITE
I suspect #2# & #3 below are the cause of the problem, in particular if someone in your household dialed *67, even accidentally.
Follow the steps, step by step, down the list:
1. a) If you're using the SmartRG 516ac in addition to some other router you haven't named, make sure the SmartRG516ac is in bridge mode.
Visit https://www.reddit.com/r/teksavvy/comme ... &context=3
Contact your ISP for assistance. Alternatively, and I find this is usually better, perform PPPoE login using the additional router (make sure your ISP allows multiple PPPoE sessions). Contact your ISP for further assistance (you will need your PPPoE username and password). Then make sure the additional router has SIP ALG/SIP Passthrough disabled (if you need help disabling SIP ALG, then I need the brand and model of the additional router).
Or
b) If you're going to going to use SmartRG 516ac without another router, then make sure SIP ALG is disabled in the SmartRG.
Visit https://speedtouch.ca/wp-content/upload ... manual.pdf
Refer to pages 71 and 72. Uncheck the "SIP Enabled" box shown on page 72. Then click "save/apply".
In some ISP's firmware, it's possible this setting is enabled and hidden to the user. Contact your ISP for help.
2. Dial *68 to remove caller ID blocking on all outbound calls.
Especially, don't use "Anonymous" for display name in your ATA. Don't use or accidentally dial *67.
3. Login in to your ATA (as an admin); navigate to Voice-->Line (whichever you use for FPL)-->Supplementary Service Subscription-->Block CID Serv:
a) change to no
b) Click "submit"
Having Block CID Serv enabled was a reported issue with SPA112s dropping calls after 15 minutes with FPL.
4. 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
This helps to ensure the RTP audio stream is being sent to your WAN IP as opposed to your LAN IP.
d) NAT Keep Alive Interval--> 20 seconds
e) click "Save Settings" button
5. 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 avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potential corrupted NAT association that developed between your router and ATA.
6. Navigate to Voice-->Line 1 (or whatever you're using for FPL)-->Proxy and Registration-->Proxy
a) use "voip4.freephoneline.ca:6060" without the quotation marks
The purpose of voip4.freephoneline.ca:6060 is to circumvent faulty SIP ALG features in routers.
If you go through this list and find after doing #3 that your problem is fixed, then you can probably use whatever proxy server you want.
b) Ensure Register Expires is set to 3600 seconds
7. 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
8. 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.
9. 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. Please reboot or power cycle your devices now in that order.
10. Test calls.
Your ATA’s setup guide is here: viewtopic.php?f=15&t=16206.
Follow the steps, step by step, down the list:
1. a) If you're using the SmartRG 516ac in addition to some other router you haven't named, make sure the SmartRG516ac is in bridge mode.
Visit https://www.reddit.com/r/teksavvy/comme ... &context=3
Contact your ISP for assistance. Alternatively, and I find this is usually better, perform PPPoE login using the additional router (make sure your ISP allows multiple PPPoE sessions). Contact your ISP for further assistance (you will need your PPPoE username and password). Then make sure the additional router has SIP ALG/SIP Passthrough disabled (if you need help disabling SIP ALG, then I need the brand and model of the additional router).
Or
b) If you're going to going to use SmartRG 516ac without another router, then make sure SIP ALG is disabled in the SmartRG.
Visit https://speedtouch.ca/wp-content/upload ... manual.pdf
Refer to pages 71 and 72. Uncheck the "SIP Enabled" box shown on page 72. Then click "save/apply".
In some ISP's firmware, it's possible this setting is enabled and hidden to the user. Contact your ISP for help.
2. Dial *68 to remove caller ID blocking on all outbound calls.
Especially, don't use "Anonymous" for display name in your ATA. Don't use or accidentally dial *67.
3. Login in to your ATA (as an admin); navigate to Voice-->Line (whichever you use for FPL)-->Supplementary Service Subscription-->Block CID Serv:
a) change to no
b) Click "submit"
Having Block CID Serv enabled was a reported issue with SPA112s dropping calls after 15 minutes with FPL.
4. 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
This helps to ensure the RTP audio stream is being sent to your WAN IP as opposed to your LAN IP.
d) NAT Keep Alive Interval--> 20 seconds
e) click "Save Settings" button
5. 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 avoid SIP Scanners (or hackers).
Also, changing local SIP port will reset a potential corrupted NAT association that developed between your router and ATA.
6. Navigate to Voice-->Line 1 (or whatever you're using for FPL)-->Proxy and Registration-->Proxy
a) use "voip4.freephoneline.ca:6060" without the quotation marks
The purpose of voip4.freephoneline.ca:6060 is to circumvent faulty SIP ALG features in routers.
If you go through this list and find after doing #3 that your problem is fixed, then you can probably use whatever proxy server you want.
b) Ensure Register Expires is set to 3600 seconds
7. 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
8. 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.
9. 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. Please reboot or power cycle your devices now in that order.
10. Test calls.
Your ATA’s setup guide is here: viewtopic.php?f=15&t=16206.
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/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at http://forum.fongo.com/viewforum.php?f=15.
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- Just Passing Thru
- Posts: 12
- Joined: 01/02/2013
Re: calls drop immediately after the 15 minute Re-INVITE/SPA
thanks very much - i took the simplest approach and did the *68. That has solved the problem.
I will go through your other suggestions and make sure my settings are correct. I've been an FPL user for so long, I can't believe this has never come up before. LOL
Really appreciate your efforts.
thanks.
I will go through your other suggestions and make sure my settings are correct. I've been an FPL user for so long, I can't believe this has never come up before. LOL
Really appreciate your efforts.
thanks.
-
- Technical Support
- Posts: 3069
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: FTTH
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: calls drop immediately after the 15 minute Re-INVITE/SPA
This situation with *67 with FPL and the SPA 112/122 ATA series began in 2020.
I’m glad the problem is fixed.
Thanks for following up.
I’m glad the problem is fixed.
Thanks for following up.
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/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at http://forum.fongo.com/viewforum.php?f=15.