[Resolved] Freepbx/Asterisk 15 minute call drop

pablob · Post by **pablob** » 02/23/2020

I found I am having the same issue (outgoing calls to landlines/mobiles) dropping after ~15 minutes, but I am not blockimg any CID. Is there anything else that may have changed for you in addition to the CID blocking?

Thanks!

Post by **Liptonbrisk** » 02/25/2020

pablob wrote:I found I am having the same issue (outgoing calls to landlines/mobiles) dropping after ~15 minutes, but I am not blockimg any CID.

SIP session timers (re-invites . . . invite is sent and your device does not respond--or the response is blocked/lost) tend to cause this problem as well as lost or corrupted NAT associations.

I'm not sure if you're still using SIP Sorcery (if so, I can't help you), but in your devices, I would change local SIP port to a new random number between 30000 and 60000 and also use voip4.freephoneline.ca:6060.
Afterwards, reboot modem/router combo (wait for it to be up and running)-->router (wait for Wi-Fi SSIDs to populate first)-->and then reboot SIP devices (ATAs, IP Phones) in that order. That's always proper device reboot order.

mkaye · Post by **mkaye** » 03/17/2020

i have been getting this all day
2 different calls - each lasted 15 minutes (called back each for another 15 min)

mark

slvrsi · Post by **slvrsi** » 03/26/2020

yes. i'm having the same problem.

running an asterisk box on my openwrt router which is in the dmz.

alg is not on and i also tried voip4. still 15ish minute limit...

incoming calls unlimited. outgoing calls 15 mins...

Post by **Liptonbrisk** » 03/26/2020

slvrsi wrote:yes. i'm having the same problem.

running an asterisk box on my openwrt router which is in the dmz.

Using DMZ is a huge security risk. It's better to ensure, if you're using a modem/router combo or gateway issued by your ISP, that it’s in bridge mode. Afterwards disable DMZ in your own router.

I'm not using Freepbx nor Asterisk, so I won't be able to help troubleshoot. I can't reproduce your issue with any FPL server for outgoing or incoming calls.

mkaye · Post by **mkaye** » 03/28/2020

strange, i seemed to fix it on my Grandstream 702 ATA, but not my Grandstream 2-line voip phone (use voip.ms for business & it's OK, not freephoneline)
all settings are identical and it has to be the phone since the ATA works fine
i added session=timers=refuse on the trunk, but that didn't help

mark

mkaye · Post by **mkaye** » 03/30/2020

just checked again
ATA has 15 min problem if long distance - local calls are OK
so, it is a freephoneline problem

mark

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote:just checked again
ATA has 15 min problem if long distance - local calls are OK
so, it is a freephoneline problem

I have no issues calling within Canada, between Ontario and B.C. I'm not going to pay to test calling outside of Canada.

mkaye · Post by **mkaye** » 03/30/2020

London, ON right now
BC a couple of days ago

mark

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote:London, ON right now
BC a couple of days ago

I'm not using Freepbx nor Asterisk, but after the server migration, some Asterisk/Freepbx users did report issues on these forums. I'm not sure if those issues are related.

I have no issues except for voicemail notifications.

For those that want to test, visit http://thetestcall.blogspot.com/
There's a 250 test number in B.C. to try. Call it. Then press #. Then press 4 for music on hold. After one song finishes, press the # again to play the next song.
There's also a 416 test number to try.

I've also spoken to some people in B.C. for over 15 minutes. My call didn't drop.

mkaye · Post by **mkaye** » 03/30/2020

Liptonbrisk wrote:
mkaye wrote:just checked again
ATA has 15 min problem if long distance - local calls are OK
so, it is a freephoneline problem
I have no issues calling within Canada, between Ontario and B.C. I'm not going to pay to test calling outside of Canada.

are you using your own PBX?
i use Freepbx and had no issues up until a month ago

mark

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote: are you using your own PBX?

No. I mostly use an Obihai OBi202 with multiple services configured on it. For my needs at home, I haven't required more than a combination of 4 separate SIP trunks and 8 voice gateways.

i use Freepbx

Yeah, so it is possible that after migration occurred, something changed that's affecting Freepbx and Asterisk users: http://forum.fongo.com/viewtopic.php?f=15&t=19702.
Unfortunately, I don't have the time to setup Freepbx and troubleshoot. But maybe other Freepbx users have suggestions: https://community.freepbx.org/.

mkaye · Post by **mkaye** » 03/30/2020

this is what i see in the log

[2020-03-30 11:05:47] NOTICE[11947] chan_sip.c: Disconnecting call 'SIP/freephoneline-0000001b' for lack of RTP activity in 31 seconds

i get this message @15 min during a call

mark

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote:this is what i see in the log

[2020-03-30 11:05:47] NOTICE[11947] chan_sip.c: Disconnecting call 'SIP/freephoneline-0000001b' for lack of RTP activity in 31 seconds

i get this message @15 min during a call

mark

Okay, that's typically an issue involving NAT firewalls, SIP ALG, or timeouts (RTP keep alive and UDP timeouts). RTP is the audio stream.
In Freepbx, see if there's RTP Keepalive or other timeout values. I'm not familiar with Freepbx.

1) what brand and model modem are you using? Please answer this question.

2) What brand and model router are you using? Edit: I see ubiquiti usg3 in your profile. I need to lookup how to change UDP timeouts for that router.

Edit: 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.

mkaye · Post by **mkaye** » 03/30/2020

Liptonbrisk wrote:
mkaye wrote:this is what i see in the log

[2020-03-30 11:05:47] NOTICE[11947] chan_sip.c: Disconnecting call 'SIP/freephoneline-0000001b' for lack of RTP activity in 31 seconds

i get this message @15 min during a call

mark

Okay, that's typically an issue involving NAT firewalls, SIP ALG, or timeouts (RTP keep alive and UDP timeouts). RTP is the audio stream.

1) what brand and model modem are you using? Please answer this question.

2) What brand and model router are you using? Edit: I see ubiquiti. I need to lookup how to change UDP timeouts.

modem = Smart/RG 808 (bridge mode) - i was using a technicolor tc4350 - ISP sent the Smart/RG Dec 20 to see if it was better - i was having signal issues on the cable

Post by **Liptonbrisk** » 03/30/2020

Liptonbrisk wrote: Edit: I see ubiquiti usg3 in your profile. I need to lookup how to change UDP timeouts for that router.

1) Disable SIP ALG
https://docs.fusionpbx.com/en/latest/fi ... le-sip-alg

"To disable SIP ALG:

Either click on the CLI button from the Ubiquiti Edgerouter GUI or via you favorite SSH client to the Edgerouter.
Then type: configure
Then type: set system conntrack modules sip disable
Then type: commit
Then type: save
Then type: exit
"

Reboot router. Wait for it to be fully up and running first. Then reboot ATAs and IP Phones. That's always the proper device reboot order.

There's a related thread here that might help: https://community.ui.com/questions/Disa ... 60696cbcb6.

Then try testing calls again. If the problem is resolved, don't bother with #2.

2) With respect to UDP timeouts, for your router the commands are

set system conntrack timeout udp stream 115
set system conntrack timeout udp other 15

stream means assured UDP timeout
other is the same as unreplied UDP timeout

These values assume you're using 20s for Keep Alive and 120s for failed retry timer.

Otherwise, you'll probably have to ensure RTP packets aren't being blocked, eventually, for some reason by your router.
Other things to look at include RTP Keep alive values and RTP timeouts. I'm not familiar with Freepbx settings, so it's probably better to ask for help at https://www.freepbx.org/community/.

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote:
modem = Smart/RG 808 (bridge mode)

Okay, I wanted to ensure that was in bridge mode.

I'm not familiar with all Freepbx/Asterisk settings, so I'm going to defer to the Freepbx community (unless someone else here can help you): https://community.freepbx.org/.
Chances are someone there has dealt with similar problems before.

mkaye · Post by **mkaye** » 03/30/2020

i found the UDP settings in the GUI
changed to your numbers
called BC #, listening to music...

mark

mkaye · Post by **mkaye** » 03/30/2020

lost call 15:16...

Post by **Liptonbrisk** » 03/30/2020

Did you disable SIP ALG?

If so, then try asking in the Freepbx forums for further suggestions. You may need to take a look at keep alive timeouts.
I hope you don't have to forward RTP ports in your router, which is a security risk.

mkaye · Post by **mkaye** » 03/30/2020

yes, SIP ALG was disabled

shouldn't have to forward ports as i am originating

mark

Post by **Liptonbrisk** » 03/30/2020

mkaye wrote: shouldn't have to forward ports as i am originating

RTP packets (the audio stream) have to reach you regardless of origination. If the NAT hole closes, or if a NAT association times out or becomes corrupted, the caller will have problems.

SIP ALG was disabled

I suggest asking at https://community.freepbx.org/ for further advice.

mkaye · Post by **mkaye** » 03/31/2020

a friend sent his incoming/outgoing config
i changed a few parameters and now it is FIXED!
not sure which parameter it was

mark

Jos · Post by **Jos** » 03/31/2020

@mkaye I have this problem whith Freephoneline can you share this config with us PLEASE

mkaye · Post by **mkaye** » 03/31/2020

incoming:
disallow=all
allow=ulaw
username=<phone_number>
type=peer
secret=<password>
qualify=yes
insecure=invite
host=voip.freephoneline.ca
fromdomain=voip.freephoneline.ca
dtmfmode=rfc2833
context=from-trunk
canreinvite=no

outgoing:
disallow=all
allow=ulaw
username=<phone_number>
type=peer
secret=<password>
qualify=yes
insecure=invite
host=voip.freephoneline.ca
dtmfmode=rfc2833
fromdomain=voip.freephoneline.ca
context=from-trunk

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