The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work.
The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's sending out. But the INVITE is responded by the Asterisk with "SIP/2.0 403 Forbidden" I've changed the INVITE message to mask the real telephone numbers, SIP server, passwords, and IP addresses, but I did that using search and replace so the structure is intact. What do I need to configure in the "Incoming Settings" panel for the CS 1500's INVITE to my Asterisk server to work? I've tried all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027...@sip.acme.com SIP/2.0 From: <sip:5552022...@172.16.10.40>;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: <sip:+15552027...@sip.acme.com> Call-ID: f379f62-29173-3895-b14271f5-40802-45...@172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Asserted-Identity: <sip:5552022...@172.16.10.40;user=phone> Privacy: none Remote-Party-ID: <sip:5552022...@172.16.10.40;user=phone>; party=calling; privacy=off Max-Forwards: 70 Supported: 100rel,replaces Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK Contact: <sip:5552022...@172.16.10.40> Authorization: Digest username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020@ sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5 Content-Type: application/SDP Content-Length: 167 v=0 o=- 2973921782 2973921782 IN IP4 172.16.10.65 s=SIP Call c=IN IP4 172.16.10.65 t=0 0 m=audio 36224 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users