I've got PBXNSIP running on a windows box and it is trying to register with my Asterisk box. I can set up one trunk and it works fine, however if I try to setup a second trunk from the same box, there is some sort of authentication issue where Asterisk appears to be confusing which trunk is which. Here is the chunk from my sip.conf:
[TEST1] context=STUFF-LD type=friend callerid="TEST1" <> username=TEST1 secret=xxxx host=dynamic nat=yes canreinvite=no qualify=yes [TEST2] context=STUFF-LD type=friend callerid="TEST2" <> username=TEST2 secret=xxxx host=dynamic nat=yes canreinvite=no qualify=yes 'sip show peers' shows both registered on Asterisk ok. If I try and call out test2, it works. However, if I try and call out test1, it fails with this: [Sep 30 12:01:10] WARNING[16678]: chan_sip.c:8272 check_auth: username mismatch, have <TEST2>, digest has <TEST1> [Sep 30 12:01:10] NOTICE[16678]: chan_sip.c:13587 handle_request_invite: Failed to authenticate user "3210" <sip:[email protected];user=phone>; tag=9055 What is happening is that since both regs come from the same remote IP, Asterisk thinks the call is coming from test2, even though it is really coming from test1 per the sip debug below. Any idea how to make Asterisk realize that the call is on test1? <--- SIP read from 192.168.100.98:5060 ---> INVITE sip:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;rport From: "3210" <sip:[email protected];user=phone>;tag=63019 To: <sip:[email protected];user=phone> Call-ID: 26f91...@pbx CSeq: 23974 INVITE Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 P-Asserted-Identity: "TEST1" <sip:[email protected]> Content-Type: application/sdp Content-Length: 196 v=0 o=- 30939 30939 IN IP4 192.168.100.98 s=- c=IN IP4 192.168.100.98 t=0 0 m=audio 63088 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (16 headers 10 lines) --- Sending to 192.168.100.98 : 5060 (NAT) Using INVITE request as basis request - 26f91...@pbx Found peer 'TEST2' <--- Reliably Transmitting (NAT) to 192.168.100.98:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;received =192.168.100.98;rport=5060 From: "3210" <sip:[email protected];user=phone>;tag=63019 To: <sip:[email protected];user=phone>;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7edf0cb1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '26f91...@pbx' in 1344 ms (Method: INVITE) Retransmitting #1 (NAT) to 192.168.100.98:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;received =192.168.100.98;rport=5060 From: "3210" <sip:[email protected];user=phone>;tag=63019 To: <sip:[email protected];user=phone>;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7edf0cb1" Content-Length: 0 --- dell860*CLI> <--- SIP read from 192.168.100.98:5060 ---> ACK sip:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;rport From: "3210" <sip:[email protected];user=phone>;tag=63019 To: <sip:[email protected];user=phone>;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- dell860*CLI> <--- SIP read from 192.168.100.98:5060 ---> INVITE sip:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-757d8077f9a2b9108ec158f2fc07d30a;rport From: "3210" <sip:[email protected];user=phone>;tag=63019 To: <sip:[email protected];user=phone> Call-ID: 26f91...@pbx CSeq: 23975 INVITE Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 P-Asserted-Identity: "TEST1" <sip:[email protected]> Proxy-Authorization: Digest realm="asterisk",nonce="7edf0cb1",response="5ececd40c28f0378503e2dd6ee5cef14 ",username="TEST1",uri="sip:[email protected];user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 196 v=0 o=- 30939 30939 IN IP4 192.168.100.98 s=- c=IN IP4 192.168.100.98 t=0 0 m=audio 63088 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <-------------> --- (17 headers 10 lines) --- Sending to 192.168.100.98 : 5060 (NAT) Using INVITE request as basis request - 26f91...@pbx Found peer 'TEST2' [Sep 30 12:11:16] WARNING[16678]: chan_sip.c:8272 check_auth: username mismatch, have <TEST2>, digest has <TEST1> [Sep 30 12:11:16] NOTICE[16678]: chan_sip.c:13587 handle_request_invite: Failed to authenticate user "3210" <sip:[email protected];user=phone>;tag=63019 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
