Jonas Kellens
2011-Nov-22 14:40 UTC
[asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call to the number 3221112233... This is all the sip debug trace gathered with asterisk : <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD To: <sip:3221112233 at A1.A1.A1.A1:5060> Call-ID: 0201FFFFCEFEA742 CSeq: 1 INVITE Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="67105ac4", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="60be856773 f86450fc9ddbaf7a568505", algorithm=MD5 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE Max-Forwards: 70 Privacy: none P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Type: application/sdp Content-Length:195 v=0 o=- 0 0 IN IP4 sip.domain.tld s=- i=(o=IN IP4 10.1.2.35) c=IN IP4 AS.AS.AS.AS t=0 0 m=audio 62654 RTP/AVP 8 0 a=rtcp:65115 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCEFEA742 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490 From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as68f71fe5 Call-ID: 0201FFFFCEFEA742 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9" Content-Length: 0 <------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCEFEA742' in 32000 ms (Method: INVITE) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as68f71fe5 Call-ID: 0201FFFFCEFEA742 CSeq: 1 ACK Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Max-Forwards: 70 User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Length: 0 <-------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 0 lines) --- [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD To: <sip:3221112233 at A1.A1.A1.A1:5060> Call-ID: 0201FFFFCCFEA242 CSeq: 1 INVITE Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="46ef24d9", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="14ecbfc7df24b49926151284c123ea11", algorithm=MD5 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE Max-Forwards: 70 Privacy: none P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Type: application/sdp Content-Length:195 v=0 o=- 0 0 IN IP4 sip.domain.tld s=- i=(o=IN IP4 10.1.2.35) c=IN IP4 AS.AS.AS.AS t=0 0 m=audio 62654 RTP/AVP 8 0 a=rtcp:65115 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCCFEA242 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;received=AS.AS.AS.AS;rport=61490 From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as1ba6ed56 Call-ID: 0201FFFFCCFEA242 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="3df09f45" Content-Length: 0 <------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCCFEA242' in 32000 ms (Method: INVITE) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as1ba6ed56 Call-ID: 0201FFFFCCFEA242 CSeq: 1 ACK Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Max-Forwards: 70 User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Length: 0 <-------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 0 lines) --- [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD To: <sip:3221112233 at A1.A1.A1.A1:5060> Call-ID: 0201FFFFCBFE9C42 CSeq: 1 INVITE Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="3df09f45", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="80683cd640815b362f74afcfcb68809a", algorithm=MD5 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE Max-Forwards: 70 Privacy: none P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Type: application/sdp Content-Length:195 v=0 o=- 0 0 IN IP4 sip.domain.tld s=- i=(o=IN IP4 10.1.2.35) c=IN IP4 AS.AS.AS.AS t=0 0 m=audio 62654 RTP/AVP 8 0 a=rtcp:65115 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCBFE9C42 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;received=AS.AS.AS.AS;rport=61490 From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as26c6d395 Call-ID: 0201FFFFCBFE9C42 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.20 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="6a7cfd54" Content-Length: 0 <------------> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCBFE9C42' in 32000 ms (Method: INVITE) [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as26c6d395 Call-ID: 0201FFFFCBFE9C42 CSeq: 1 ACK Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> Max-Forwards: 70 User-Agent: A5000 R52-H2C0205 P-Behind-Gsi: 192.168.6.1 Content-Length: 0 Thanks. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111122/28201220/attachment-0001.htm>
Bruce Ferrell
2011-Nov-22 15:25 UTC
[asterisk-users] Asterisk refuses INVITE (401) and I don't know why
Jonas, May I suggest that you present us your sip.conf entry for this peer, properly redacted, of course. That might help more. What I do for "gateways" at known addresses is to put an entry like this into the sip.conf entry: [peer] type=peer defaultip=192.168.40.123 insecure=invite,port context=some_context On 11/22/2011 06:40 AM, Jonas Kellens wrote:> Hello list, > > this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't > know why. > > Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. > > > A1.A1.A1.A1 = IP-address Asterisk PBX > AS.AS.AS.AS = IP-address Aastra PBX > > Aastra PBX makes a call to the number 3221112233... > > This is all the sip debug trace gathered with asterisk : > > > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD > To: <sip:3221112233 at A1.A1.A1.A1:5060> > Call-ID: 0201FFFFCEFEA742 > CSeq: 1 INVITE > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="67105ac4", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="60be856773 > f86450fc9ddbaf7a568505", algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE > Max-Forwards: 70 > Privacy: none > P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Type: application/sdp > Content-Length:195 > > v=0 > o=- 0 0 IN IP4 sip.domain.tld > s=- > i=(o=IN IP4 10.1.2.35) > c=IN IP4 AS.AS.AS.AS > t=0 0 > m=audio 62654 RTP/AVP 8 0 > a=rtcp:65115 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > <-------------> > > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCEFEA742 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490 > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as68f71fe5 > Call-ID: 0201FFFFCEFEA742 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9" > Content-Length: 0 > > <------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCEFEA742' in 32000 ms (Method: INVITE) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=310158BD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as68f71fe5 > Call-ID: 0201FFFFCEFEA742 > CSeq: 1 ACK > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Max-Forwards: 70 > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Length: 0 > > > <-------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 0 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD > To: <sip:3221112233 at A1.A1.A1.A1:5060> > Call-ID: 0201FFFFCCFEA242 > CSeq: 1 INVITE > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="46ef24d9", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="14ecbfc7df24b49926151284c123ea11", > algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE > Max-Forwards: 70 > Privacy: none > P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Type: application/sdp > Content-Length:195 > > v=0 > o=- 0 0 IN IP4 sip.domain.tld > s=- > i=(o=IN IP4 10.1.2.35) > c=IN IP4 AS.AS.AS.AS > t=0 0 > m=audio 62654 RTP/AVP 8 0 > a=rtcp:65115 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > <-------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCCFEA242 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;received=AS.AS.AS.AS;rport=61490 > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as1ba6ed56 > Call-ID: 0201FFFFCCFEA242 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="3df09f45" > Content-Length: 0 > > > <------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCCFEA242' in 32000 ms (Method: INVITE) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=33015DBD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as1ba6ed56 > Call-ID: 0201FFFFCCFEA242 > CSeq: 1 ACK > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Max-Forwards: 70 > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Length: 0 > > > <-------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 0 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > INVITE sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD > To: <sip:3221112233 at A1.A1.A1.A1:5060> > Call-ID: 0201FFFFCBFE9C42 > CSeq: 1 INVITE > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", nonce="3df09f45", uri="sip:3221112233 at A1.A1.A1.A1:5060", response="80683cd640815b362f74afcfcb68809a", > algorithm=MD5 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE > Max-Forwards: 70 > Privacy: none > P-Asserted-Identity: <sip:3224445566 at sip.domain.tld> > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Type: application/sdp > Content-Length:195 > > v=0 > o=- 0 0 IN IP4 sip.domain.tld > s=- > i=(o=IN IP4 10.1.2.35) > c=IN IP4 AS.AS.AS.AS > t=0 0 > m=audio 62654 RTP/AVP 8 0 > a=rtcp:65115 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > > > <-------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 11 lines) --- > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP TOS bits 184 > [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35] == Using SIP RTP CoS mark 5 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to AS.AS.AS.AS : 61490 (no NAT) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE request as basis request - 0201FFFFCBFE9C42 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490 > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;received=AS.AS.AS.AS;rport=61490 > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as26c6d395 > Call-ID: 0201FFFFCBFE9C42 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="6a7cfd54" > Content-Length: 0 > > > <------------> > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling destruction of SIP dialog '0201FFFFCBFE9C42' in 32000 ms (Method: INVITE) > [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] > <--- SIP read from UDP:AS.AS.AS.AS:61490 ---> > ACK sip:3221112233 at A1.A1.A1.A1:5060 SIP/2.0 > Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK847851481531358325;rport > From: <sip:SIPPEERusername at sip.domain.tld:5060>;tag=340163BD > To: <sip:3221112233 at A1.A1.A1.A1:5060>;tag=as26c6d395 > Call-ID: 0201FFFFCBFE9C42 > CSeq: 1 ACK > Contact: <sip:SIPPEERusername at AS.AS.AS.AS:61490> > Max-Forwards: 70 > User-Agent: A5000 R52-H2C0205 > P-Behind-Gsi: 192.168.6.1 > Content-Length: 0 > > > > Thanks. > > Kind regards, > Jonas. > > > -- > _____________________________________________________________________ > -- 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/20111122/cd164b58/attachment.htm>
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