Hello, I am using Kamailio as SIP register with asterisk integration describe from hire: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060
everything is working well except some software VoIP clients (like Yate) and CISCO phone like Cisco-CP7940G/8.0 and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part: #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; If i commented out this part the call is passing, but I do not have auth anymore (everyone can register)
Here is ngrep: U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4. Call-ID: 931919626@1.2.3.4. CSeq: 13 INVITE. User-Agent: YATE/5.3.0. Contact: sip:10891@192.168.0.40:5060. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30.
U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626@1.2.3.4. CSeq: 13 INVITE. Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I". Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. .
U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626@1.2.3.4. CSeq: 13 ACK. Max-Forwards: 20. Contact: sip:10891@192.168.0.40:5060. User-Agent: YATE/5.3.0. Content-Length: 0. .
U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4. Call-ID: 931919626@1.2.3.4. User-Agent: YATE/5.3.0. Contact: sip:10891@192.168.0.40:5060. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30.
U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4. Call-ID: 931919626@1.2.3.4. CSeq: 14 INVITE. Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. .
U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Record-Route: sip:1.2.3.4;lr=on;ftag=838449717;nat=yes. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4. Call-ID: 931919626@1.2.3.4. User-Agent: YATE/5.3.0. Contact: sip:10891@192.168.0.40:5060. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Content-Type: application/sdp. Content-Length: 499. . v=0. o=yate 1406132227 1406132227 IN IP4 1.2.3.4. s=SIP Call. c=IN IP4 1.2.3.4. t=0 0. m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. a=nortpproxy:yes.
U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=as31a58bb2. Call-ID: 931919626@1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. .
U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=as31a58bb2. Call-ID: 931919626@1.2.3.4. CSeq: 14 ACK. Content-Length: 0. .
U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=as31a58bb2. Call-ID: 931919626@1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. .
U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: sip:10891@1.2.3.4;tag=838449717. To: sip:0896995837@1.2.3.4;tag=as31a58bb2. Call-ID: 931919626@1.2.3.4. CSeq: 14 ACK. Max-Forwards: 20. Contact: sip:10891@192.168.0.40:5060. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. User-Agent: YATE/5.3.0. Content-Length: 0. —
And kamailio.cfg attached: proLogika Sent with Airmail
Hello,
it is asterisk that asks second time - kamailio is verifying the auth ok, then forwards to asterisk which asks again for authentication. Read the notes from the Asterisk Database section in the tuorial:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb#...
You should not set the secret in asterisk table, but use another column to store the password.
Cheers, Daniel
On 23/07/14 21:52, proLogika wrote:
Hello, /I am using Kamailio as SIP register with asterisk integration describe from hire:/ http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060
everything is working well except some software VoIP clients (like Yate) and CISCO phone like/ Cisco-CP7940G/8.0 /and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part:
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sipusers", "1")) {#!else
if (!auth_check("$fd", "subscriber", "1")) {#!endif
auth_challenge("$fd", "0"); exit;If i commented out this part the call is passing, but I do not have auth anymore (everyone can register)
Here is ngrep:
U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060
INVITE sip:0896995837@1.2.3.4 SIP/2.0.
Max-Forwards: 20.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692.
From: sip:10891@1.2.3.4;tag=838449717.
Call-ID: 931919626@1.2.3.4.
CSeq: 13 INVITE.
User-Agent: YATE/5.3.0.
Contact: sip:10891@192.168.0.40:5060.
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO.
Content-Type: application/sdp.
Content-Length: 481.
.
v=0.
o=yate 1406132227 1406132227 IN IP4 192.168.0.40.
s=SIP Call.
c=IN IP4 192.168.0.40.
t=0 0.
m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:11 L16/8000.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:102 SPEEX/8000.
a=rtpmap:103 SPEEX/16000.
a=rtpmap:104 SPEEX/32000.
a=rtpmap:105 iSAC/16000.
a=rtpmap:106 iSAC/32000.
a=rtpmap:101 telephone-event/8000.
a=ptime:30.
U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=6be166fd53062bbc5b6dd79656b620cd.1950.
Call-ID: 931919626@1.2.3.4.
CSeq: 13 INVITE.
Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I".
Server: kamailio (4.0.6 (x86_64/linux)).
Content-Length: 0.
.
U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060
ACK sip:0896995837@1.2.3.4 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=6be166fd53062bbc5b6dd79656b620cd.1950.
Call-ID: 931919626@1.2.3.4.
CSeq: 13 ACK.
Max-Forwards: 20.
Contact: sip:10891@192.168.0.40:5060.
User-Agent: YATE/5.3.0.
Content-Length: 0.
.
U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060
INVITE sip:0896995837@1.2.3.4 SIP/2.0.
Max-Forwards: 20.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
Call-ID: 931919626@1.2.3.4.
User-Agent: YATE/5.3.0.
Contact: sip:10891@192.168.0.40:5060.
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO.
CSeq: 14 INVITE.
Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5.
Content-Type: application/sdp.
Content-Length: 481.
.
v=0.
o=yate 1406132227 1406132227 IN IP4 192.168.0.40.
s=SIP Call.
c=IN IP4 192.168.0.40.
t=0 0.
m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:11 L16/8000.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:102 SPEEX/8000.
a=rtpmap:103 SPEEX/16000.
a=rtpmap:104 SPEEX/32000.
a=rtpmap:105 iSAC/16000.
a=rtpmap:106 iSAC/32000.
a=rtpmap:101 telephone-event/8000.
a=ptime:30.
U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
Call-ID: 931919626@1.2.3.4.
CSeq: 14 INVITE.
Server: kamailio (4.0.6 (x86_64/linux)).
Content-Length: 0.
.
U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060
INVITE sip:0896995837@1.2.3.4 SIP/2.0.
Record-Route: sip:1.2.3.4;lr=on;ftag=838449717;nat=yes.
Max-Forwards: 16.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
Call-ID: 931919626@1.2.3.4.
User-Agent: YATE/5.3.0.
Contact: sip:10891@192.168.0.40:5060.
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO.
CSeq: 14 INVITE.
Content-Type: application/sdp.
Content-Length: 499.
.
v=0.
o=yate 1406132227 1406132227 IN IP4 1.2.3.4.
s=SIP Call.
c=IN IP4 1.2.3.4.
t=0 0.
m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:11 L16/8000.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:102 SPEEX/8000.
a=rtpmap:103 SPEEX/16000.
a=rtpmap:104 SPEEX/32000.
a=rtpmap:105 iSAC/16000.
a=rtpmap:106 iSAC/32000.
a=rtpmap:101 telephone-event/8000.
a=ptime:30.
a=nortpproxy:yes.
U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=as31a58bb2.
Call-ID: 931919626@1.2.3.4.
CSeq: 14 INVITE.
Server: Asterisk PBX 1.8.29.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c".
Content-Length: 0.
.
U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060
ACK sip:0896995837@1.2.3.4 SIP/2.0.
Max-Forwards: 16.
Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=as31a58bb2.
Call-ID: 931919626@1.2.3.4.
CSeq: 14 ACK.
Content-Length: 0.
.
U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=as31a58bb2.
Call-ID: 931919626@1.2.3.4.
CSeq: 14 INVITE.
Server: Asterisk PBX 1.8.29.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c".
Content-Length: 0.
.
U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060
ACK sip:0896995837@1.2.3.4 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344.
From: sip:10891@1.2.3.4;tag=838449717.
To: sip:0896995837@1.2.3.4;tag=as31a58bb2.
Call-ID: 931919626@1.2.3.4.
CSeq: 14 ACK.
Max-Forwards: 20.
Contact: sip:10891@192.168.0.40:5060.
Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5.
User-Agent: YATE/5.3.0.
Content-Length: 0.
—
And kamailio.cfg attached:
proLogika Sent with Airmail
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
proLogika