HT503-freepbx assistance request


I have asterisk 13 with freepbx 13. I have recently purchased a HT503 to act as an FXO gateway.
I have outgoing calls working through a trunk, but I can not get the incoming calls to work.

I keep getting the error:
res_pjsip/pjsip_distributor.c:246 Request from ‘“pstn” sip:2106643452@’ failed for ‘’ (callid: ZTNhYjsjdhsjfifewnoBlYTE3YzgwZTU4MGY.) - No matching endpoint found

I have tried so many variations, and followed so many guides, and nothing seem to work.
My main source was

You are my last hope of salvation.
Please help!!!




In order to assist you to solve this issue, kindly help to open a ticket in our helpdesk portal via the link below :



I already have,
Ticket #: 20160126105730

i was hoping for some experienced advice, or someone that has done this before.


Have a look here, see if it helps.


and thank you for your reply,
I have been following this, but to no avail.
with freepbx 12 the grandstream to freepbx works.
with freepbx 13 it does not!!!


if you have created a chan_sip trunk on freepbx, remember to point the unconditional ptsn call forward to the correct port… that should be port 5160 for chan_sip, instead of 5060 that is used for pjsip.


I don’t know if it’s of any interest to me