This is a really simple problem that I just can't get to work. There are two Asterisk servers with the following sip user and peer. When a call is attempted, Asterisk is not sending authentication details in response to the 401. Note, if the secret is blank on 172.16.0.2 test, the INVITE succeeds.
on 172.16.0.2: [test] type=friend secret=abcde host=dynamic context=demo on 172.16.0.1 : [natty] type=peer host=172.16.0.2 fromuser=test secret=abcde originate SIP/natty/1234568 extension 200 == Using SIP RTP CoS mark 5 Audio is at 172.16.0.1 port 19486 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.0.2:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]:5066>;tag=as1689b2b6 To: <sip:[email protected]> Contact: <sip:[email protected]:5066> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1 Date: Sat, 14 Apr 2012 09:10:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 290 v=0 o=root 1594270426 1594270426 IN IP4 172.16.0.1 s=Asterisk PBX 1.6.2.9-2ubuntu2.1 c=IN IP4 172.16.0.1 t=0 0 m=audio 19486 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:172.16.0.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066 From: "asterisk" <sip:[email protected]:5066>;tag=as1689b2b6 To: <sip:[email protected]>;tag=as1a6c2364 Call-ID: [email protected] CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.9-2ubuntu2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a03a1d3" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 172.16.0.2:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]:5066>;tag=as1689b2b6 To: <sip:[email protected]>;tag=as1a6c2364 Contact: <sip:[email protected]:5066> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1 Content-Length: 0 --- [Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:[email protected]:5066>;tag=as1689b2b6' -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
