SIP authentication with 7960 phones


#1

I have AAH set up with 7960 phones. Everything is ok, except that I realized recently, that I always get tuck in “calling (out inv)” and “reorder” status.

The sip debug showes that my Asterisk keeps trying to Challenge the Invite as follows,

Retransmitting #4 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK39194026
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780652042ec82d-6a1bf24f
To: sip:*97@192.168.1.99;user=phone;tag=as5cd9c7ec
Call-ID: 00082166-ff7800c6-390916ff-252ac3b8@192.168.1.101
CSeq: 102 INVITE

If I try several times, I will get it through. But, this is really annoying.

I even tried to use “insecure=very” for my extentions to disable furthur authentication, but it seems that the Authentication for that Invite is still sending out by AAH.

I really don’t know whether this is a bug of Asterisk or AAH.

Please help.

tks

Peng


#2

Yes, you’re right.

But, when I read again about the definition of Peer/User/Friend, I finally think they have no much difference, except that friend=peer+user, I mean attributes.

So, my understanding is that , it’s really doesn’t matter much, in this case, as both User and Peer or Friend(user+peer) have the Insecure attribute.

Thanks again for your input, this will finally leads to more clear conclusion.

What I don’t understand is that if you try several times, you get it working.

not sure whether this is something wrong with Asterisk or Phone.

Need more help.

tks

By: savvid - lipeng
RE: SIP Authentication problem
2005-07-08 10:28
Furthur debug from the phone, it seems that Phone is sending the Hash with Invite 102, but, for some reason, at that time, the asterisk still just finished it’s 488 message.

It just missed that Invite 102 with credentials and it started to see invite 102 with no credentials and it Chanllenged again.

oiceStream> [06:18:16] SIPTaskProcessListEvent: pSm->Cmd= 0x161700
[06:18:16] sip_cc_event LINE 0/1: --0x0004b279-- : SIP_STATE_IDLE <- E_CC_SETUP
[06:18:16] idle_ev_cc_setup: All digits collected. Placing the call
[06:18:16] SIPSM 0/1/1: idle_ev_cc_setup : Local RTP port: 16386
[06:18:16] SIPSPISendInvite: Sending INVITE…
[06:18:16] sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK27f3ac1b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
Date: Fri, 08 Jul 2005 10:18:16 GMT
CSeq: 101 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Accept: application/sdp

v=0
o=Cisco-SIPUA 6302 22039 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16386 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

, length=<739>
[06:18:16] LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
[06:18:16] LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Starting reTx timer (500 msec)
[06:18:16] CHANGE STATE: LINE 0/1: : State change: SIP_STATE_IDLE -> SIP_STATE_SENT_INVITE
[06:18:17] SIPTaskProcessListEvent: pSm->Cmd= 0x161c09
[06:18:17] sip_sm_Procees_event LINE 0/1: --0x00050585-- : SIP_STATE_SENT_INVITE <- E_SIP_TIMER
[06:18:17] LINE 0/1: ccsip_handle_default_sip_timer : Resending message: #1
[06:18:17] sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK27f3ac1b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
Date: Fri, 08 Jul 2005 10:18:16 GMT
CSeq: 101 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 248
Accept: application/sdp

v=0
o=Cisco-SIPUA 6302 22039 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16386 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

, length=<739>
[06:18:17] LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
[06:18:17] LINE 0/1: ccsip_restart_reTx_timer : Restarting timer (1000 msec) (msg is INVITE)
[06:18:17] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:17] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK27f3ac1b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as35d4f4b4
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="379db67a"
Content-Length: 0

, length=503
[06:18:17] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:17] SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
[06:18:17] SIPTaskProcessSIPMessage: Received SIP response.
[06:18:17] sipSPICheckResponse: Response match: callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=101, cseq_method=INVITE
[06:18:17] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:17] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=101, cseq_method=INVITE)
[06:18:17] SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
[06:18:17] sip_sm_Procees_event LINE 0/1: --0x0004c901-- : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
[06:18:17] LINE 0/1: SIP 407 Proxy Authentication required
[06:18:17] SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
[06:18:17] sipRelDevCoupledMessageStore: Storing for reTx (cseq=101, method=INVITE, to_tag=)
[06:18:17] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <192.168.1.99>:<5060>, handle = 8
[06:18:17] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <192.168.1.99>:<5060>, handle=<8>:
message=
<ACK sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK27f3ac1b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as35d4f4b4
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
Date: Fri, 08 Jul 2005 10:18:17 GMT
CSeq: 101 ACK
Content-Length: 0

, length=<368>
[06:18:17] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8
[06:18:17] Proxy-Authenticate= Digest realm=“asterisk”, nonce=“379db67a”
[06:18:17] sipSPISendInviteMidCall: Sending INVITE…
[06:18:17] sipSPIGenRequestURI: Forming Req-URI (Caller): using original Req-URI
[06:18:17] SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
[06:18:17] sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
Date: Fri, 08 Jul 2005 10:18:17 GMT
CSeq: 102 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“7c00c7ba5308b822f5ed47f403696520”,nonce=“379db67a”,algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 248

v=0
o=Cisco-SIPUA 6302 22039 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16386 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

, length=<874>
[06:18:17] LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
[06:18:17] LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Starting reTx timer (500 msec)
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:18] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK27f3ac1b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as35d4f4b4
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Content-Length: 0

, length=430
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
[06:18:18] SIPTaskProcessSIPMessage: Received SIP response.
[06:18:18] sipSPICheckResponse: Response mismatch:
(Response:callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=101, cseq_method=INVITE),
(Request: callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:18] sipSPICheckResponse: Stray Response: Response branch: z9hG4bK27f3ac1bRequest branch: z9hG4bK62eaa502
[06:18:18] SIPTaskProcessSIPMessage: Stale response detected. Discarding w/o sending 400…
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:18] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
[06:18:18] SIPTaskProcessSIPMessage: Received SIP response.
[06:18:18] sipSPICheckResponse: Response match: callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE
[06:18:18] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:18] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:18] SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
[06:18:18] sip_sm_Procees_event LINE 0/1: --0x0004c901-- : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
[06:18:18] LINE 0/1: SIP 407 Proxy Authentication required
[06:18:18] LINE 0/1: retries exceeded: 2/2
[06:18:18] CHANGE STATE: LINE 0/1: : State change: SIP_STATE_SENT_INVITE -> SIP_STATE_RELEASE
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x161700
[06:18:18] sip_cc_event LINE 0/1: --0x0004bd01-- : SIP_STATE_RELEASE <- E_CC_RELEASE_COMPLETE
[06:18:18] LINE 0/1: sip_sm_call_cleanup : Cleaning up the call…
[06:18:18] CHANGE STATE: LINE 0/1: : State change: SIP_STATE_RELEASE -> SIP_STATE_IDLE
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:18] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:18] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:19] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:19] SIPTaskProcessSIPMessage: Not forwarding response to SIP SM.
[06:18:19] SIPTaskProcessSIPPreviousCallInviteResponse:Error: Last Bye CSeq=0, Failure Code = 407, CSeq:102
[06:18:19] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:19] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:20] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:20] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:20] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:20] SIPTaskProcessSIPMessage: Not forwarding response to SIP SM.
[06:18:20] SIPTaskProcessSIPPreviousCallInviteResponse:Error: Last Bye CSeq=0, Failure Code = 407, CSeq:102
[06:18:20] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:20] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:21] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:21] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:21] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:21] SIPTaskProcessSIPMessage: Not forwarding response to SIP SM.
[06:18:21] SIPTaskProcessSIPPreviousCallInviteResponse:Error: Last Bye CSeq=0, Failure Code = 407, CSeq:102
[06:18:21] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:21] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:22] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:22] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:22] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:22] SIPTaskProcessSIPMessage: Not forwarding response to SIP SM.
[06:18:22] SIPTaskProcessSIPPreviousCallInviteResponse:Error: Last Bye CSeq=0, Failure Code = 407, CSeq:102
[06:18:22] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:22] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:23] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:23] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:23] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:23] SIPTaskProcessSIPMessage: Not forwarding response to SIP SM.
[06:18:23] SIPTaskProcessSIPPreviousCallInviteResponse:Error: Last Bye CSeq=0, Failure Code = 407, CSeq:102
[06:18:25] SIPTaskProcessListEvent: pSm->Cmd= 0x161700
[06:18:25] sip_sm_process_cc_event: No ccb with matching gsm_id = <1>
VoiceStream>
VoiceStream>
VoiceStream> undebug all
debugs: NONE
VoiceStream> undebug sip-task sip-state sip-messages sip-reg-state
debugs: NONE

By: savvid - lipeng
RE: SIP Authentication problem
2005-07-08 10:34
it seems that Asterisk gets ACK from phone to its challenge for invite 101, so, Asterisk sends a 488.

at this time, the invite 102 comes, and the Asterisk is still processing that 488 , so it doesn’t have time in processing this 102.

When it’s ready for this 102, phone never sends that Cretential again.

REtry 2/2 exceeded?

By: savvid - lipeng
RE: SIP Authentication problem
2005-07-08 10:47
Response:callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=101, cseq_method=INVITE),
(Request: callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:18] sipSPICheckResponse: Stray Response: Response branch: z9hG4bK27f3ac1bRequest branch: z9hG4bK62eaa502
[06:18:18] SIPTaskProcessSIPMessage: Stale response detected. Discarding w/o sending 400…
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x160200
[06:18:18] SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK62eaa502
From: “2001” sip:2001@192.168.1.99;tag=00082166ff7800021d5ae4c8-0672e7ff
To: sip:*97@192.168.1.99;user=phone;tag=as37283c88
Call-ID: 00082166-ff780003-5153d840-2f7247b9@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63cced98"
Content-Length: 0

, length=503
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Determining destination line…
[06:18:18] SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
[06:18:18] SIPTaskProcessSIPMessage: Received SIP response.
[06:18:18] sipSPICheckResponse: Response match: callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE
[06:18:18] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
[06:18:18] LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780003-5153d840-2f7247b9@192.168.1.101, cseq=102, cseq_method=INVITE)
[06:18:18] SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
[06:18:18] sip_sm_Procees_event LINE 0/1: --0x0004c901-- : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
[06:18:18] LINE 0/1: SIP 407 Proxy Authentication required
[06:18:18] LINE 0/1: retries exceeded: 2/2
[06:18:18] CHANGE STATE: LINE 0/1: : State change: SIP_STATE_SENT_INVITE -> SIP_STATE_RELEASE
[06:18:18] SIPTaskProcessListEvent: pSm->Cmd= 0x161700
[06:18:18] sip_cc_event LINE 0/1: --0x0004bd01-- : SIP_STATE_RELEASE <- E_CC_RELEASE_COMPLETE
[06:18:18] LINE 0/1: sip_sm_call_cleanup : Cleaning up the call…

This part of message proves that it indeed, the asterisk get ACK for it’s 407 challenge for 101. The Asterisk didn’t get what it expectes,so it sends a 488,
but, at this time, phone is expecting a respons for 102, but got 488,

So, either the asterisk should not reply this 488 when it receives the ACK, wait a second, then it will see 102 with cretentials.

Or , the phone doesn’t send that ACK for 407, but a 101 with credentials,

or it starts with 101 invite with Credials.

confused.

By: usatracy - usatracy
RE: SIP Authentication problem
2005-07-08 11:06
I had a somewhat similar problem with a hacked lingo ATA

It would register and work but if the registration timedout only a reboot of thr ATA would get it up again.

I set the
defaultip= to the ip of the ata and that let asterisk find it to call it and that fixed it.

By: savvid - lipeng
RE: SIP Authentication problem
2005-07-08 11:27
This is the debug from the phone when it sometimes works.

really wired.

???

oiceStream> SIPTaskProcessListEvent: pSm->Cmd= 0x161700
sip_cc_event LINE 0/1: --0x0004b279-- : SIP_STATE_IDLE <- E_CC_SETUP
idle_ev_cc_setup: All digits collected. Placing the call
SIPSM 0/1/3: idle_ev_cc_setup : Local RTP port: 16386
SIPSPISendInvite: Sending INVITE…
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK50765343
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:15 GMT
CSeq: 101 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 249
Accept: application/sdp

v=0
o=Cisco-SIPUA 16982 25318 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16386 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

, length=<740>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Starting reTx timer (500 msec)
CHANGE STATE: LINE 0/1: : State change: SIP_STATE_IDLE -> SIP_STATE_SENT_INVITE
SIPTaskProcessListEvent: pSm->Cmd= 0x160200
SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK50765343
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as681a1b64
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="63284c2a"
Content-Length: 0

, length=503
SIPTaskProcessSIPMessage: Line filter: Determining destination line…
SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
SIPTaskProcessSIPMessage: Received SIP response.
sipSPICheckResponse: Response match: callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=101, cseq_method=INVITE
SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=101, cseq_method=INVITE)
SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
sip_sm_Procees_event LINE 0/1: --0x0004c901-- : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
LINE 0/1: SIP 407 Proxy Authentication required
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
sipRelDevCoupledMessageStore: Storing for reTx (cseq=101, method=INVITE, to_tag=)
sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <192.168.1.99>:<5060>, handle = 8
sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <192.168.1.99>:<5060>, handle=<8>:
message=
<ACK sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK50765343
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as681a1b64
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:15 GMT
CSeq: 101 ACK
Content-Length: 0

, length=<368>
sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8
Proxy-Authenticate= Digest realm=“asterisk”, nonce="63284c2a"
sipSPISendInviteMidCall: Sending INVITE…
sipSPIGenRequestURI: Forming Req-URI (Caller): using original Req-URI
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK3c6b714b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:15 GMT
CSeq: 102 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“8cbfd0c4dabb523fdda10ccf463456b8”,nonce=“63284c2a”,algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 249

v=0
o=Cisco-SIPUA 16982 25318 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16386 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

, length=<875>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Starting reTx timer (500 msec)
SIPTaskProcessListEvent: pSm->Cmd= 0x160200
SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK3c6b714b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Content-Length: 0

, length=402
SIPTaskProcessSIPMessage: Line filter: Determining destination line…
SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
SIPTaskProcessSIPMessage: Received SIP response.
sipSPICheckResponse: Response match: callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=102, cseq_method=INVITE
SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=102, cseq_method=INVITE)
SIPTaskProcessSIPMessage: Recv 1xx message.
sip_sm_Procees_event LINE 0/1: --0x0004b799-- : SIP_STATE_SENT_INVITE <- E_SIP_1xx
LINE 0/1: sip_sm_200and300_update : To header doesn’t contain “;tag=” param!
LINE 0/1: sip_sm_200and300_update : Recorded to_tag=<>
LINE 0/1: sentinvite_ev_sip_1xx : SIP_STATE_SENT_INVITE <- SIP 100 TRYING
SIPTaskProcessListEvent: pSm->Cmd= 0x160200
SIPProcessUDPMessage: recv UDP message from <192.168.1.99>:<50195>:
<SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK3c6b714b
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 2011 2011 IN IP4 192.168.1.99
s=session
c=IN IP4 192.168.1.99
t=0 0
m=audio 10088 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

, length=684
SIPTaskProcessSIPMessage: Line filter: Determining destination line…
SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line = <0/1>.
SIPTaskProcessSIPMessage: Received SIP response.
sipSPICheckResponse: Response match: callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=102, cseq_method=INVITE
SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers…
LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101, cseq=102, cseq_method=INVITE)
SIPTaskProcessSIPMessage: Recv 2xx message.
sip_sm_Procees_event LINE 0/1: --0x0004ba59-- : SIP_STATE_SENT_INVITE <- E_SIP_2xx
LINE 0/1: sip_sm_200and300_update : Recorded to_tag=
LINE 0/1: sip_util_extract_sdp : Process SDP: Dest=<192.168.1.99>:<10088>
CHANGE STATE: LINE 0/1: : State change: SIP_STATE_SENT_INVITE -> SIP_STATE_SENT_INVITE_CONNECTED
SIPTaskProcessListEvent: pSm->Cmd= 0x161700
sip_cc_event LINE 0/1: --0x0004cf03-- : SIP_STATE_SENT_INVITE_CONNECTED <- E_CC_CONNECTED_ACK
sipSPISendAck: Sending ACK…
sipSPIGenRequestURI: Forming Req-URI: using Contact
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=)
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<ACK sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK553a0458
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:16 GMT
CSeq: 102 ACK
User-Agent: CSCO/7
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“8cbfd0c4dabb523fdda10ccf463456b8”,nonce=“63284c2a”,algorithm=md5
Content-Length: 0

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
CHANGE STATE: LINE 0/1: : State change: SIP_STATE_SENT_INVITE_CONNECTED -> SIP_STATE_ACTIVE
SIPTaskProcessListEvent: pSm->Cmd= 0x161700
sip_cc_event LINE 0/1: --0x0004d359-- : SIP_STATE_ACTIVE <- E_CC_RELEASE
sipSPISendBye: Sending BYE…
sipSPIGenRequestURI: Forming Req-URI: using Contact
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<BYE sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK71a3a270
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:23 GMT
CSeq: 103 BYE
User-Agent: CSCO/7
Content-Length: 0
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6793db3fbb5609367175b5aa37d77977”,nonce=“63284c2a”,algorithm=md5

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Starting reTx timer (500 msec)
CHANGE STATE: LINE 0/1: : State change: SIP_STATE_ACTIVE -> SIP_STATE_RELEASE
SIPTaskProcessListEvent: pSm->Cmd= 0x161c09
sip_sm_Procees_event LINE 0/1: --0x00050585-- : SIP_STATE_RELEASE <- E_SIP_TIMER
LINE 0/1: ccsip_handle_default_sip_timer : Resending message: #1
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<BYE sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK71a3a270
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:23 GMT
CSeq: 103 BYE
User-Agent: CSCO/7
Content-Length: 0
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6793db3fbb5609367175b5aa37d77977”,nonce=“63284c2a”,algorithm=md5

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: ccsip_restart_reTx_timer : Restarting timer (1000 msec) (msg is BYE)
SIPTaskProcessListEvent: pSm->Cmd= 0x161c09
sip_sm_Procees_event LINE 0/1: --0x00050585-- : SIP_STATE_RELEASE <- E_SIP_TIMER
LINE 0/1: ccsip_handle_default_sip_timer : Resending message: #2
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<BYE sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK71a3a270
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:23 GMT
CSeq: 103 BYE
User-Agent: CSCO/7
Content-Length: 0
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6793db3fbb5609367175b5aa37d77977”,nonce=“63284c2a”,algorithm=md5

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: ccsip_restart_reTx_timer : Restarting timer (2000 msec) (msg is BYE)

VoiceStream>
VoiceStream> undebugSIPTaskProcessListEvent: pSm->Cmd= 0x161c09
sip_sm_Procees_event LINE 0/1: --0x00050585-- : SIP_STATE_RELEASE <- E_SIP_TIMER
LINE 0/1: ccsip_handle_default_sip_timer : Resending message: #3
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<BYE sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK71a3a270
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:23 GMT
CSeq: 103 BYE
User-Agent: CSCO/7
Content-Length: 0
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6793db3fbb5609367175b5aa37d77977”,nonce=“63284c2a”,algorithm=md5

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: ccsip_restart_reTx_timer : Restarting timer (4000 msec) (msg is BYE)
sip-task sip-state sip-messages sip-reg-stateSIPTaskProcessListEvent: pSm->Cmd= 0x161c09
sip_sm_Procees_event LINE 0/1: --0x00050585-- : SIP_STATE_RELEASE <- E_SIP_TIMER
LINE 0/1: ccsip_handle_default_sip_timer : Resending message: #4
sip_sm_cc_channel_send_buf_fcn: Sent SIP message: handle=<2>:
message=
<BYE sip:*97@192.168.1.99:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK71a3a270
From: “2001” sip:2001@192.168.1.99;tag=00082166ff780004043ed566-1297d95b
To: sip:*97@192.168.1.99;user=phone;tag=as434d96ff
Call-ID: 00082166-ff780005-7c0e7366-1e060f5d@192.168.1.101
Date: Fri, 08 Jul 2005 11:24:23 GMT
CSeq: 103 BYE
User-Agent: CSCO/7
Content-Length: 0
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6793db3fbb5609367175b5aa37d77977”,nonce=“63284c2a”,algorithm=md5

, length=<542>
LINE 0/1: sip_sm_cc_channel_send_buf_fcn : Stopping reTx timer
LINE 0/1: ccsip_restart_reTx_timer : Restarting timer (4000 msec) (msg is BYE)

debugs: NONE

By: savvid - lipeng
RE: SIP Authentication problem
2005-07-08 11:31
So, it seems that phone is stable in its behaviour, which send invite 101 and get 407 and send back ACK with 101 and generates new invite 102.

the problem is, at this time, sometimes, the asterisk takes it, some times, it doesn’t, instead, keeping sedning out another 407 for invite 102?

is this a pace (timing issue)? or a bug?

tks
peng


#3

This is the debug in the similiar problem situation.

The asterisk receiveds ACK for 101 and invite 102 with Credentials, but still request 407 for 102.

Really don’t know what it thinks. it seems for the successful one, it takes from here right away and sends back Tring and 200 ok.
Sip read:
ACK sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK75065310
From: “2001” sip:2001@192.168.1.99;tag=00082166ff78000a381bc4fb-000c10db
To: sip:*97@192.168.1.99;user=phone;tag=as7bc34092
Call-ID: 00082166-ff78000b-70fa3674-13f44249@192.168.1.101
Date: Fri, 08 Jul 2005 13:11:27 GMT
CSeq: 101 ACK
Content-Length: 0

8 headers, 0 lines
Destroying call '00082166-ff78000b-70fa3674-13f44249@192.168.1.101’
asterisk1*CLI>

Sip read:
INVITE sip:*97@192.168.1.99;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5a26f0be
From: “2001” sip:2001@192.168.1.99;tag=00082166ff78000a381bc4fb-000c10db
To: sip:*97@192.168.1.99;user=phone
Call-ID: 00082166-ff78000b-70fa3674-13f44249@192.168.1.101
Date: Fri, 08 Jul 2005 13:11:27 GMT
CSeq: 102 INVITE
User-Agent: CSCO/7
Contact: sip:2001@192.168.1.101:5060
Proxy-Authorization: Digest username=“2001”,realm=“asterisk”,uri=“sip:192.168.1.99”,response=“6e3f5ef7a1d59aae309e29c5c6ab710d”,nonce=“189c04a5”,algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 249

v=0
o=Cisco-SIPUA 19617 11006 IN IP4 192.168.1.101
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 16388 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 11 lines
Using latest request as basis request
Sending to 192.168.1.101 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5a26f0be
From: “2001” sip:2001@192.168.1.99;tag=00082166ff78000a381bc4fb-000c10db
To: sip:*97@192.168.1.99;user=phone;tag=as3e3c6c42
Call-ID: 00082166-ff78000b-70fa3674-13f44249@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="603218c8"
Content-Length: 0

to 192.168.1.101:5060
Scheduling destruction of call ‘00082166-ff78000b-70fa3674-13f44249@192.168.1.101’ in 15000 ms
Found user '2001’
Retransmitting #1 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5a26f0be
From: “2001” sip:2001@192.168.1.99;tag=00082166ff78000a381bc4fb-000c10db
To: sip:*97@192.168.1.99;user=phone;tag=as3e3c6c42
Call-ID: 00082166-ff78000b-70fa3674-13f44249@192.168.1.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*97@192.168.1.99
Proxy-Authenticate: Digest realm=“asterisk”, nonce="603218c8"
ontent-Length: 0

Gurus from Asterisk@digium and Asterisk community are really welcome for the solutions.

Many thanks

Peng