Freephoneline settings on FreePBX

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This forum is for those users who have already purchased a configuration file with the SIP settings needed to configure any SIP compatible device.

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Freephoneline settings on FreePBX

Postby Juste » 06/07/2019


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.

Just Passing Thru
Posts: 10
Joined: 02/29/2012

Re: Freephoneline settings on FreePBX

Postby menext » 04/12/2021

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 =>

[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.

context=from-sip ; this section will be defined in extensions.conf
One Hit Wonder
Posts: 1
Joined: 04/12/2021
SIP Device Name: Asterisk/ATA122/CISCO/GDS3710
Computer OS: FreeBSD
Router: FreeBSD

Re: Freephoneline settings on FreePBX

Postby Liptonbrisk » 02/28/2023

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


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):

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: Visit to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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SIP Device Name: Obihai 202/2182, Groundwire
Firmware Version: various
ISP Name: gateway/hub in bridge mode
Computer OS: Windows 64 bit
Router: Asuswrt-Merlin & others

Re: Freephoneline settings on FreePBX

Postby ilneofita » 08/14/2023

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
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