Freephoneline settings on FreePBX

[Juste]

Hello,

Could anyone please remind me the settings Freephoneline needs to work with Asterisk (FreePBX) ?

I tried to replicate whatever has been mentioned on this forum a few years ago, with no luck.

Thanks,

[menext]

[general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=IP.ADD ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
register => 1234567890:SECERETPASSWORD@voip2.freephoneline.ca:5060
acl=tellme
useragent=CISCO-TEL

[office-phone](!) ; create a template for our devices
type=friend ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan
host=dynamic ; the device will register with asterisk
nat=yes ; assume device is behind NAT
; *** NAT stands for Network Address Translation, which allows
; multiple internal devices to share an external IP address.
secret=VERYBIGSECRET ; a secure password for this device -- DON'T USE THIS PASSWORD!
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=ulaw ; which audio codecs to accept from, and request to, the device
allow=alaw ; in the order we prefer


This may be very late for your answer but I am sure that others will be interested. You need to setup SIP not PJSIP. Modify your sip.conf accordingly. Below is an example that works with the proper username and password.

[freephoneline]
type=peer
secret=SECRETPASSWORD
username=1234567890
host=voip2.freephoneline.ca
fromuser=1234567890
fromdomain=YOUR.IP.ADD.OR.YOURDOMAIN
canreinvite=no
insecure=invite,port
qualify=yes
nat=force_rport,comedia
context=from-sip ; this section will be defined in extensions.conf
deny=0.0.0.0/0.0.0.0
permit=162.213.111.22/255.255.255.255

[Liptonbrisk]

Check Settings->Asterisk SIP Settings->Chan SIP Settings (Registration Times) to see if you have the following:

defaultexpiry=3600
RegisterExpiry=3600
MaxExpiry=3600
MinExpiry=3000
registertimeout=120

Settings might be within sip_general_custom.conf file
Or when using FreePBX web UI try Asterisk-->Asterisk SIP settings






This should be under Peer Details (for FPL trunk configuration with FreePBX):

keepalive=20
nat=yes

[ilneofita]

I modified the pjsip in order to have it worked with FPL, the modification is on git asterisk and for sure will be included in the next release

[mkaye]

chan_sip works fine
trying to convert to pjsip
can't seem to get the settings correct
created new pjsip trunk, incoming calls terminate after 30s
upgraded to Freepbx17/asterisk21 and let it convert all my chan_sip to pjsip - trunk won't register

[Liptonbrisk]

created new pjsip trunk, incoming calls terminate after 30s

Call drops due to lack of ACK.

I can't test, but have you taken a look at https://community.freepbx.org/t/incomin ... s/95369/12 or tried asking there?
That person had to set Contact User to 1FPLnumber.

https://github.com/asterisk/asterisk/pull/232

[mkaye]

Contact User solved the pjsip register problem
but have the 30s call terminated problem with pjsip, not with chan_sip

[Liptonbrisk]

Contact User solved the pjsip register problem
but have the 30s call terminated problem with pjsip, not with chan_sip

gideons at https://community.freepbx.org/t/incomin ... s/95369/12 is using pjsip and Freephoneline, and the problem described isn't with registration. It's with incoming calls, which that user fixed. The subject of that thread is "Incoming Call drop after 32 seconds".

I decided to give it a try and add back the Contact User.
And so as I am making the call and following the logs, what a surprise: I couldn’t believe my eyes: an ACK !
Then the call went beyond the 32 seconds."

I would try asking at the Freepbx forums and post your logs there: https://community.freepbx.org/.

Additionally, ensure SIP ALG is off in your router(s).

[mkaye]

i have all my connections working with pjsip, except FPL trunk
with pjsip outgoing calls terminate after 15 min
i had fixed this with chan_sip, but i don't see any parameter in the pjsip menus to help

[Liptonbrisk]

Freephoneline uses 15 minute session timers. It's likely that ACK isn't received after the Re-INVITE.

Provided the trunk is registered, one cause might be a lost NAT association or a timeout issue.

UDP timeout adjustments are shown at viewtopic.php?p=80377#p80377.

I don't know if disabling timers would work. It's usually not advised. Possibly using timers = no
for Freephoneline's trunk endpoint might work.

You could also see if there's any difference for you between using voip2.freephoneline.ca:5060 and voip4.freephoneline.ca:6060

I'm not able to test, and people don't seem to post here often.
Is something preventing you from asking at the FreePBX community forums?
https://community.freepbx.org/

[mkaye]

i had adjusted the UDP timeouts already

i don't see any option in the pjsip advanced parameters to disable timers (FreePBX17)

i agree, it seems to be not receiving the ACK after the re-invite

[Liptonbrisk]

i don't see any option in the pjsip advanced parameters to disable timers (FreePBX17)

Okay, I did a quick search, and I see this:
https://community.freepbx.org/t/disable ... ui/42059/3

We were able to solve the 15-minute hangup problem by using the pjsip.endpoint_custom_post.conf file. By adding “timers=no” to the trunk, the calls stopped timing out. I wish there had been a way to configure this in the pjsip trunk settings.

pjsip.endpoint_custom_post.conf

[mytrunk](+)
timers=no

Notice the (+) after the trunk name. It appears to need that to append to the other settings FreePBX manages.

I think normally you're advised to fix whatever is causing the ACK to not be received (possible NAT issue or maybe something weird happening with the contact header), but if there's no other easy fix, then trying settings timers to no. I'm guessing because I can't test.

If it were me, I would try asking at https://community.freepbx.org/.

[mkaye]

didn't fix it
my trunk is called fpl
i used [fpl](+)

[Liptonbrisk]

didn't fix it
my trunk is called fpl
i used [fpl](+)

That's unfortunate.

I see that you posted at https://community.freepbx.org/t/disable ... ui/42059/7 and, in turn, bumped a 6 year old thread. Your responses there also don't appear to be asking for help or a question, so much as making a statement (that is, it seems to me as though you're saying, "This doesn't work" and not, "This doesn't work; can someone please help me?").

Respectfully, I would suggest creating a new thread at the FreePBX community forums, posting all your trunk, endpoint, related fpl settings, and also a log showing when the Re-INVITE occurs and that no ACK is received (include the contact header too). Then, at the same time, I would ask a question (ex. What should I do to help ensure ACK is received after the Re-INVITE?). Then if no one responds, you know that no one knows the answer or that no one is otherwise willing to help.

I hope you do discover the solution or that someone responds to you with a fix somewhere. I can't test.

If ACK isn't received after 15 minutes, I usually think a firewall is blocking something or that there's a failed NAT association.
Setting qualify_frequency to 20 seconds might help, if it wasn't set to that previously. I know it used to be called "Qualify Frequency" in the GUI for PJSIP (pjsip Settings-->Advanced). Anyway, I can't test, so hopefully someone using pjsip responds to you.

[Julie]

Hello, can Liptonbrisk or mkaye share both your settings (sip and pjsip), for the trunk, extension, inbound & outbound route and the Sip Settings? It would be greatly appreciated.

Currently, I am using chan_sip. Inbound calls worked but not the outbound

Thanks
Julie

[Liptonbrisk]

I don’t use FreePBX or Asterisk.

For troubleshooting FreePBX issues, I suggest creating a new thread at https://community.freepbx.org/

For Asterisk, visit https://community.asterisk.org/