I tried to get it all this time.
[code]<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘030915bb03e1454f072073d305ab342d@127.0.1.1’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
INVITE sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Expires: 20LI>
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 289 0 IN IP4 192.168.1.75
s=SIP Call
t=0 0
m=audio 23236 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.75
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (18 headers 13 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)
Using INVITE request as basis request - 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
<— Reliably Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as06d9944f
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3d5d199d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75’ in 32000 ms (Method: INVITE)
Found user '400’
Server-01*CLI>
<— SIP read from 192.168.1.75:51809 —>
ACK sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK7851866e
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as06d9944f
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 101 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
INVITE sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:46 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“db4dbe0ce8a5a774a16a860f984e08ea”,nonce=“3d5d199d”,algorithm=MD5
Expires: 20
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 289 0 IN IP4 192.168.1.75
s=SIP Call
t=0 0
m=audio 23236 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.75
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (19 headers 13 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)
Using INVITE request as basis request - 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Found user '400’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.75:23236
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.75:23236
Looking for 7125726 in from-internal (domain 192.168.1.50)
list_route: hop: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7125726@192.168.1.50
Content-Length: 0
<------------>
– Executing [7125726@from-internal:1] Macro(“SIP/400-90028250”, “user-callerid|SKIPTTL|”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/400-90028250”, “AMPUSER=400”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/400-90028250”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/400-90028250”, “1|Set|REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/400-90028250”, “AMPUSER=400”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/400-90028250”, “AMPUSERCIDNAME=Scott Dickens”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/400-90028250”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/400-90028250”, “AMPUSERCID=400”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/400-90028250”, “CALLERID(all)=“Scott Dickens” <400>”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/400-90028250”, “REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-user-callerid:10] GotoIf(“SIP/400-90028250”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] NoOp(“SIP/400-90028250”, “Using CallerID “Scott Dickens” <400>”) in new stack
– Executing [7125726@from-internal:2] Set(“SIP/400-90028250”, “_NODEST=”) in new stack
– Executing [7125726@from-internal:3] Macro(“SIP/400-90028250”, “record-enable|400|OUT|”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/400-90028250”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/400-90028250”, “recordingcheck|20090506-094455|1241621095.6”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090506-094455|1241621095.6: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/400-90028250”, “”) in new stack
– Executing [7125726@from-internal:4] Macro(“SIP/400-90028250”, “dialout-trunk|2|7125726||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/400-90028250”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/400-90028250”, “0?sub-pincheck|s|1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/400-90028250”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/400-90028250”, “DIAL_NUMBER=7125726”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/400-90028250”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/400-90028250”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/400-90028250”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/400-90028250”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/400-90028250”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/400-90028250”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/400-90028250”, “outbound-callerid|2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/400-90028250”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/400-90028250”, “0|Set|REALCALLERIDNUM=400”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/400-90028250”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/400-90028250”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/400-90028250”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/400-90028250”, “TRUNKOUTCID=OrderCounter.com <8504177821>”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/400-90028250”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/400-90028250”, “1|Set|CALLERID(all)=OrderCounter.com <8504177821>”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/400-90028250”, “0|Set|CALLERID(all)=”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/400-90028250”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/400-90028250”, “1|AGI|fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 1850|NXXXXXX
– AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/400-90028250”, “OUTNUM=7125726”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/400-90028250”, “custom=SIP/Broadvoice”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/400-90028250”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/400-90028250”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/400-90028250”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/400-90028250”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/400-90028250”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/400-90028250”, “SIP/Broadvoice/7125726|300|”) in new stack
Audio is at 192.168.1.50 port 19056
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.32.221:5060:
INVITE sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;rport
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 18549 18549 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 19056 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called Broadvoice/7125726
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 100 Trying
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;received=70.167.229.87;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 INVITE
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=ghij
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;received=70.167.229.87;rport=5060
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXfue552qrTz7eynoBW"
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 147.135.32.221:5060:
ACK sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4034aae8;rport
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=ghij
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Audio is at 192.168.1.50 port 19056
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.32.221:5060:
INVITE sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri="sip:7125726@sip.broadvoice.com", nonce=“BroadWorksXfue552qrTz7eynoBW”, response=“deca2fc65d0cc8ed176d5f6910ae14af”, qop=auth, cnonce=“6b695575”, nc=00000001
Date: Wed, 06 May 2009 14:44:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 18549 18550 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 19056 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 100 Trying
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 183 Session Progress
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Contact: sip:7125726@147.135.32.221
Content-Length: 170
Content-Type: application/sdp
v=0
o=2475106551 10 10 IN IP4 147.135.32.247
s=-
c=IN IP4 147.135.32.247
t=0 0
m=audio 36276 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (9 headers 8 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 147.135.32.247:36276
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 147.135.32.247:36276
– SIP/Broadvoice-00b70c60 is making progress passing it to SIP/400-90028250
Audio is at 192.168.1.50 port 14632
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7125726@192.168.1.50
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 18549 18549 IN IP4 192.168.1.50
s=session
c=IN IP4 192.168.1.50
t=0 0
m=audio 14632 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 - - - -
a=ptime:20
a=sendrecv
<------------>
Really destroying SIP dialog ‘030915bb03e1454f072073d305ab342d@127.0.1.1’ Method: REGISTER
Server-01*CLI>
<— SIP read from 192.168.1.76:49160 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:54 GMT
CSeq: 5483 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79cb2f9”;+u.sip!model.ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30
<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.76 : 5060 (no NAT)
<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5483 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK62f2716f;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50;tag=as79139fa8
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5483 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="54499ea5"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.76:49160 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:54 GMT
CSeq: 5484 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79cb2f9”;+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest username=“337”,realm=“asterisk”,uri=“sip:192.168.1.50”,response=“0929bfac132e5c01d39861527ff6cf69”,nonce=“54499ea5”,algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30
<------------->
— (14 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.76 : 5060 (no NAT)
<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5484 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.76:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK64187b42;received=192.168.1.76
From: sip:337@192.168.1.50;tag=0016c79cb2f90b07bc97f578-f7d0c421
To: sip:337@192.168.1.50;tag=as79139fa8
Call-ID: 0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76
CSeq: 5484 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp;expires=60
Date: Wed, 06 May 2009 14:45:03 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0016c79c-b2f90002-6d6c8468-072ffd98@192.168.1.76’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
CSeq: 4886 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model.ccm.cisco.com="30006"
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30
<------------->
— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4886 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bK0e5c2e3e;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4886 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0f86f375"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75’ in 32000 ms (Method: REGISTER)
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:44:55 GMT
CSeq: 4887 REGISTER
User-Agent: Cisco-CP7970G/8.3.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0016c79d6ce2”;+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest username=“400”,realm=“asterisk”,uri=“sip:192.168.1.50”,response=“5f5a7bc5af19f61c85eaa8ac89f74569”,nonce=“0f86f375”,algorithm=MD5
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 30
<------------->
— (14 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.1.75 : 5060 (no NAT)
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4887 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Server-01*CLI>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKea117db4;received=192.168.1.75
From: sip:400@192.168.1.50;tag=0016c79d6ce209a3a05388cc-f27163a6
To: sip:400@192.168.1.50;tag=as04670f5c
Call-ID: 0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75
CSeq: 4887 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp;expires=60
Date: Wed, 06 May 2009 14:45:04 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0016c79d-6ce20002-da6f5440-e325a7c6@192.168.1.75’ in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog ‘792c1f223157143e197b1e795a56920f@192.168.1.50’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.75:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK374d8bc9;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as2b89eccd
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Contact: sip:Unknown@192.168.1.50
Call-ID: 792c1f223157143e197b1e795a56920f@192.168.1.50
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 81
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.50
Voice-Message: 0/0
Server-01*CLI>
<— SIP read from 192.168.1.75:51810 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK374d8bc9;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as2b89eccd
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.75:5060;transport=udp
Call-ID: 792c1f223157143e197b1e795a56920f@192.168.1.50
Date: Wed, 06 May 2009 14:44:59 GMT
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘792c1f223157143e197b1e795a56920f@192.168.1.50’ Method: NOTIFY
Scheduling destruction of SIP dialog ‘4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.76:5060:
NOTIFY sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2b3a005e;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5eb7ca8a
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp
Contact: sip:Unknown@192.168.1.50
Call-ID: 4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 81
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.50
Voice-Message: 0/0
Server-01*CLI>
<— SIP read from 192.168.1.76:51561 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2b3a005e;rport
From: “Unknown” sip:Unknown@192.168.1.50;tag=as5eb7ca8a
To: sip:7b452e87-4496-4762-e11f-b26751a1884b@192.168.1.76:5060;transport=udp
Call-ID: 4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50
Date: Wed, 06 May 2009 14:45:02 GMT
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘4ef03e4f28371cc4486efcaa1b2f2e1c@192.168.1.50’ Method: NOTIFY
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
CANCEL sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:45:06 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP7970G/8.3.0
Content-Length: 0
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“d3e7327ba70cb5ea39ac0aecddb44184”,nonce=“3d5d199d”,algorithm=MD5
<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.75 : 5060 (no NAT)
<— Reliably Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<— Transmitting (no NAT) to 192.168.1.75:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKaae44f24;received=192.168.1.75
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50;tag=as5b0fd446
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 147.135.32.221:5060:
CANCEL sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Scheduling destruction of SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ in 32000 ms (Method: INVITE)
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/400-90028250’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 7125726, 4) exited non-zero on ‘SIP/400-90028250’
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/400-90028250”, “hangupcall|”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/400-90028250”, “vw”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/400-90028250”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/400-90028250”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/400-90028250”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/400-90028250”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/400-90028250”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/400-90028250’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/400-90028250’
Server-01*CLI>
<— SIP read from 192.168.1.75:49399 —>
ACK sip:7125726@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKb516ab71
From: “Scott Dickens” sip:400@192.168.1.50;tag=0016c79d6ce209a2c40430bc-752e8516
To: sip:7125726@192.168.1.50
Call-ID: 0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75
Max-Forwards: 70
Date: Wed, 06 May 2009 14:45:06 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7970G/8.3.0
Proxy-Authorization: Digest username=“400”,realm=“asterisk”,uri="sip:7125726@192.168.1.50",response=“d3e7327ba70cb5ea39ac0aecddb44184”,nonce=“3d5d199d”,algorithm=MD5
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘0016c79d-6ce20011-b02e0494-ed5ea8de@192.168.1.75’ Method: ACK
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 200 OK
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 CANCEL
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Server-01*CLI>
<— SIP read from 147.135.32.221:5060 —>
SIP/2.0 487 Request Terminated
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 INVITE
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;received=70.167.229.87;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to 147.135.32.221:5060:
ACK sip:7125726@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK21775af7;rport
From: “OrderCounter.com” sip:8504177821@sip.broadvoice.com;tag=as08350477
To: sip:7125726@sip.broadvoice.com;tag=1234
Contact: sip:8504177821@192.168.1.50
Call-ID: 3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Really destroying SIP dialog ‘3fd46d76694ee9d071a04b154e7ec602@sip.broadvoice.com’ Method: INVITE
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.32.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK20184940;rport
From: sip:8504177821@sip.broadvoice.com;tag=as05667d6a
To: sip:8504177821@sip.broadvoice.com
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 195 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfue3wjwlT1mcs6mBW”, response=“eb001b625a8e8d608c2f0d691c88ac22”, qop=auth, cnonce=“1830fa28”, nc=00000058
Expires: 120
Contact: sip:400@192.168.1.50
Event: registration
Content-Length: 0
Retransmitting #1 (no NAT) to 147.135.32.221:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK20184940;rport
From: sip:8504177821@sip.broadvoice.com;tag=as05667d6a
To: sip:8504177821@sip.broadvoice.com
Call-ID: 030915bb03e1454f072073d305ab342d@127.0.1.1
CSeq: 195 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“8504177821”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfue3wjwlT1mcs6mBW”, response=“eb001b625a8e8d608c2f0d691c88ac22”, qop=auth, cnonce=“1830fa28”, nc=00000058
Expires: 120
Contact: sip:400@192.168.1.50
Event: registration
Content-Length: 0
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