@keltekfreepbx wrote:
Using Ver 12.7.5-1807-1.sng7
with Asterisk 13.22.0Have 2 nic’s
192.168.2.210 - inside lan and handsets
172.16.90.10, gw 172.16.90.1 (which is the interface to AT&T’s IP FLEX, for sip handoff.)Trunk
Outgoing - which works fine.
name = att_out_1
ty;e=peer
host=172.16.90.1
dtmfmode=rfc2833
qualify=2000Incoming, which doesn’t work
att_in_1
type=friend
host=172.16.90.1
dtmfmode=rfc2833
qualify=2000
insecure=port,inviteError I see in logs
[2018-11-14 14:10:00] NOTICE[860] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“KEL-TEK” sip:7146580061@172.16.90.1’ failed for ‘172.16.90.1:5060’ (callid: 7104279757036919@c1b07_2_1) - No matching endpoint found
[2018-11-14 14:10:03] NOTICE[23489] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“KEL-TEK” sip:7146580061@172.16.90.1’ failed for ‘172.16.90.1:5060’ (callid: 7330994338851397@c4b08_2_2) - No matching endpoint found
I have all inbound traffic routed to extension 107, which is working fine.
It is AT&T’s sip handoff product, it doesn’t require or want any usernames or passwords, just for sip to it, and receive sip from it.
I also get the following errors with wireshark;
from 172.16.90.10 (my pbx) to 172.16.90.1 status 401 Unauthorized
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