HT503-freepbx assistance request


#1

Hi,
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@192.168.4.6’ failed for ‘192.168.4.7:5060’ (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 http://wiki.freepbx.org/pages/viewpage.action?pageId=33293313

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

Lazaros


#2

Hi,

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

https://helpdesk.grandstream.com

Thanks.


#3

I already have,
Ticket #: 20160126105730

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


#4

Have a look here, see if it helps.
http://wiki.freepbx.org/pages/viewpage.action?pageId=33293313


#5

Hi,
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!!!


#6

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.


#7

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