After many hours of fiddling around, Andres gave me the final piece. For those looking to implement SIP Trunks on a CS-1500 with Asterisk, here are the pieces:
Diagram: CS-1500 ------ customer PBX (172.16.10.40) (172.16.10.195) HOST: should be the DNS name assigned to the CS-1500's SIP interface. e.g. sip.acme.com NUSR: user name used for the CS 1500 to login into the customer PBX. Needs to match up FreePBX's "Trunk Name". For those who use the CLI, this section in sip.conf is encased in square brackets. i.e. [customername] NPSW: password used for the CS 1500 to login into the customer PBX. Needs to match up with the secret= line. i.e. secret=password IP: IP address of the customer PBX. i.e. 172.16.10.195 LUSR: user name used for the customer PBX to login into the CS 1500. Needs to match up with the username= line. i.e. username=customername LPSW: password used for the customer PBX to login into the CS 1500. Needs to match up with the secret= line. i.e. secret=password. For simplicity we made NUSR/LUSR the same and NPSW/LPSW the same. Since you need to define a trunk per customer, it makes the most sense and it easiest to support and implement. Here's what you need to add to Asterisk's sip.conf (yes, just those few lines!) [customername] host=sip.acme.com type=friend username=customername secret=password And the CS-1500 output: TYP TG NUM 1234 TGTP 2WAY TGNM SIP MG NO SIGT SIP STSI 0 HNPA 555 RC 0 RTP 0 TRNL PRFX PRFX 24 APFX NONE TRFC NONE 4XCD YES ACKA NO TYPC NOCO NXX UNKN LATA 000 CMCT NO TGID NONE SIT NO CNAR NO LRN NONE TNDM NO LDAT NO TRFC NONE EOAT NO ATIC NO CMCO NO TGMU NO HOST sip.acme.com NUSR customername NPSW password IP 172.16.10.195 PORT 5060 PROT UDP T38F NO AUTH YES LUSR customername LPSW password CLIM 7 CPBY 0 Frank -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Bulk - iName.com Sent: Monday, January 05, 2009 6:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very" 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 _______________________________________________ -- 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