Ben WIlliams
2012-Apr-14 09:30 UTC
[asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE
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:1234568 at 172.16.0.2 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 To: <sip:1234568 at 172.16.0.2> Contact: <sip:test at 172.16.0.1:5066> Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 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:test at 172.16.0.1:5066>;tag=as1689b2b6 To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 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:1234568 at 172.16.0.2 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport Max-Forwards: 70 From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 Contact: <sip:test at 172.16.0.1:5066> Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 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:test at 172.16.0.1:5066>;tag=as1689b2b6'
Olivier
2012-Apr-14 10:03 UTC
[asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE
Le 14 avril 2012 11:30, Ben WIlliams <bwilliams+asterisk at jadeworld.com> a ?crit :> 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, AsteriskWhich instance are you talking about, here ?> 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:1234568 at 172.16.0.2 SIP/2.0 > Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport > Max-Forwards: 70 > From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2> > Contact: <sip:test at 172.16.0.1:5066> > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:1234568 at 172.16.0.2 SIP/2.0 > Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport > Max-Forwards: 70 > From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 > Contact: <sip:test at 172.16.0.1:5066> > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:test at 172.16.0.1:5066>;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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120414/b27e74af/attachment.htm>
Ben WIlliams
2012-Apr-15 00:59 UTC
[asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE
172.16.0.1 is not sending the authentication details to 172.16.0.2 when 172.16.0.2 responds with 401. On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams <bwilliams+asterisk at jadeworld.com> wrote:> 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:1234568 at 172.16.0.2 SIP/2.0 > Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport > Max-Forwards: 70 > From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2> > Contact: <sip:test at 172.16.0.1:5066> > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:1234568 at 172.16.0.2 SIP/2.0 > Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport > Max-Forwards: 70 > From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6 > To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364 > Contact: <sip:test at 172.16.0.1:5066> > Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1 > 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:test at 172.16.0.1:5066>;tag=as1689b2b6'