Hi all
I am struggling to get the Grandstream 813 fxo port to communicate through freepbx
Currently not getting either incoming or outgoing calls
Please help 
First of, you need to help, at least post some inkling of what you have so far failed on
i expect its not authenticating. Heres what i have in the trunk
type=peer
host=192.168.1.101
username=01283734683
secret=01283734683
context=from-trunk
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
And that is what i am unsing in the grandstream sip authentication details, pointing to the ip of freepbx
As the ATA is on the LAN I would use type=friend with no user name or secret , similarly on the ATA route
Always use sngrep for the 10000 foot view.
ok changed those. And seeing a SIP flow viewer. i am unable to seethe ip of the ATA there though.
Ok, so remove the SIP user and authenticate ID from the ATA?
You said
host=192.168.1.101
If you don’t see that IP then something very basic is misconfigured.
Yes, not there at all…
Start over . . . (after the FM bit is completed )
Getting a few ‘call setup’ through with a random phone number method ‘invite’?
Theres nothing much to start over - server ip and user ID seems all there is to it?
Sorry, i have no idea what you did or what you are posting
The INVITES you are seeing are almost certainly random callers from Iceland or the Netherlands (or somewhere else) knocking at you door but that is your firewall also misconfigured.
When you see an INVITE from 192.168.1.101 after you call your phone number,then you have got your ata to know where your PBX is until then you need to read the GS/FreePBX resources related to this particular hardware. When you see that then next steps . . .
okie dokie, no worries.
Thanks for your help. i’ll get further now i expect!
1 Like
Did you get this working? I have a working config that I can share. Although my device does not detect disconnection, so please let me know if you’re in the UK and you get that working.
No, not got it working yet, but yes, in the UK.
Fire it this way, and if/when i can get disconnection working ill let you know 
I recommend having the 813 register to the PBX.
Peer details:
type=friend
host=dynamic
username=01283734683
secret=01283734683
context=from-trunk
dtmfmode=rfc2833
disallow=all
allow=ulaw
User details: (leave blank)
In the HT813 FXO page:
Account active: yes
Primary SIP Server: 192.168.1.100:5160
SIP User ID: 01283734683
Authenticate ID: 01283734683
Authenticate Password: 01283734683
(Replace 192.168.1.100 with the IP address of the PBX. Replace 5160 with the bind_port for chan_sip, if you set it to something other than the default.)
Does the Status page of the HT show registered? If so, report what happens on attempted incoming and outgoing calls. If not, report what, if anything, appears in the Asterisk log for the registration attempts.
Ive changed all those. But doesnt show registered on the grandstream device.
I can see it in the asterisk log as unspecified with the telephone number as the name/username.
When i call in it does as before, justs rings and doesnt connect
Please post the error message(s) from the log related to the attempted registration.
Says not registered on the Grandstream
under asterisk info:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
01283734683/01283734683 192.168.1.101 Yes Yes 5060 Unmonitored
in the log: [2020-04-03 18:32:22] VERBOSE[3876][C-00000011] pbx.c: Executing [[email protected]:3] Set(“SIP/01283734683-00000007”, “TECH=SIP”) in new stack
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] pbx.c: Executing [[email protected]:4] Set(“SIP/01283734683-00000007”, “SIPHEADERKEYS=”) in new stack
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] pbx.c: Executing [[email protected]:5] While(“SIP/01283734683-00000007”, “0”) in new stack
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] app_while.c: Jumping to priority 13
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] pbx.c: Executing [[email protected]:14] Return(“SIP/01283734683-00000007”, “”) in new stack
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] app_stack.c: Spawn extension (from-trunk, 07970462989, 1) exited non-zero on ‘SIP/01283734683-00000007’
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] app_stack.c: SIP/01283734683-00000007 Internal Gosub(func-apply-sipheaders,s,1(2)) complete GOSUB_RETVAL=
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] app_dial.c: Called SIP/01283734683/
[2020-04-03 18:32:22] VERBOSE[3876][C-00000011] app_dial.c: SIP/01283734683-00000007 is ringing
So the system knows but Grandstream doesnt
The log you just posted appears to be from an attempted call, not registration.
Confirm that in the FXO page, SIP Registration is set to Yes.
Reboot the HT so it attempts to register and without trying any calls, see if anything gets logged by Asterisk. If so, paste the relevant section of the log at https://pastebin.freepbx.org and post the link here.
SIP registration Yes
Rebooted, and nothing in the log about attempted connection
Please post a screenshot of the FXO page.