All pastes #232334 Raw Edit

Unnamed

public text v1 · immutable
·published 2006-11-01 09:16 UTC
rendered paste body
SIP Debugging enabled
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:
INVITE sip:12016180234@bmy.gotdns.org SIP/2.0
To: <sip:12016180234@bmy.gotdns.org>
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;rport
Call-ID: 5615f77f064f0558
CSeq: 1 INVITE
Contact: <sip:RemoteBrian@10.0.0.100:9168>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 3004w stamp 16863
Content-Length: 265

v=0
o=- 10621336 10621370 IN IP4 10.0.0.100
s=eyeBeam
c=IN IP4 10.0.0.100
t=0 0
m=audio 9170 RTP/AVP 100 6 0 8 3 18 5 101
a=alt:1 1 : 97678153 DB683B18 10.0.0.100 9170
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (12 headers 11 lines)---
Sending to 67.84.215.54 : 9168 (NAT)
Using INVITE request as basis request - 5615f77f064f0558
Reliably Transmitting (NAT) to 67.84.215.54:9168:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as3c692f9f
Call-ID: 5615f77f064f0558
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f9ec4a9"
Content-Length: 0


---
Scheduling destruction of SIP dialog '5615f77f064f0558' in 32000 ms (Method: INVITE)
Found user 'RemoteBrian'
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:
ACK sip:12016180234@bmy.gotdns.org SIP/2.0
To: <sip:12016180234@bmy.gotdns.org>;tag=as3c692f9f
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-130614794-1--d87543-;rport
Call-ID: 5615f77f064f0558
CSeq: 1 ACK
Content-Length: 0


--- (7 headers 0 lines)---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:
INVITE sip:12016180234@bmy.gotdns.org SIP/2.0
To: <sip:12016180234@bmy.gotdns.org>
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;rport
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
Contact: <sip:RemoteBrian@10.0.0.100:9168>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="RemoteBrian",realm="asterisk",nonce="1f9ec4a9",uri="sip:12016180234@bmy.gotdns.org",response="56a8d5af038526da41fb51b4108811eb",algorithm=MD5
User-Agent: eyeBeam release 3004w stamp 16863
Content-Length: 265

v=0
o=- 10621336 10621370 IN IP4 10.0.0.100
s=eyeBeam
c=IN IP4 10.0.0.100
t=0 0
m=audio 9170 RTP/AVP 100 6 0 8 3 18 5 101
a=alt:1 1 : 97678153 DB683B18 10.0.0.100 9170
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (13 headers 11 lines)---
Sending to 67.84.215.54 : 9168 (NAT)
Using INVITE request as basis request - 5615f77f064f0558
Found user 'RemoteBrian'
Found RTP audio format 100
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.100:9170
Found description format speex for ID 100
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.100:9170
Peer video RTP is at port 10.0.0.100:54710
Looking for 12016180234 in superuser (domain bmy.gotdns.org)
list_route: hop: <sip:RemoteBrian@10.0.0.100:9168>
Transmitting (NAT) to 67.84.215.54:9168:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Length: 0


---
    -- Executing [12016180234@superuser:1] SetCallerID("SIP/RemoteBrian-0820e840", "<2016180234>") in new stack
    -- Executing [12016180234@superuser:2] Dial("SIP/RemoteBrian-0820e840", "iax2/espike@nufone/12016180234") in new stack
    -- Called espike@nufone/12016180234
    -- Call accepted by 66.225.202.80 (format ulaw)
    -- Format for call is ulaw
Audio is at 127.0.0.1 port 13202
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 67.84.215.54:9168:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8032 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
    -- IAX2/nufone-5 is making progress passing it to SIP/RemoteBrian-0820e840
    -- IAX2/nufone-5 is making progress passing it to SIP/RemoteBrian-0820e840
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
    -- IAX2/nufone-5 answered SIP/RemoteBrian-0820e840
Audio is at 127.0.0.1 port 13202
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
Retransmitting #1 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
Retransmitting #2 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
Retransmitting #3 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
    -- Hungup 'IAX2/nufone-5'
  == Spawn extension (superuser, 12016180234, 2) exited non-zero on 'SIP/RemoteBrian-0820e840'
Scheduling destruction of SIP dialog '5615f77f064f0558' in 32000 ms (Method: INVITE)
Retransmitting #4 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
Retransmitting #5 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
Retransmitting #6 (NAT) to 67.84.215.54:9168:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.100:9168;branch=z9hG4bK-d87543-377315927-1--d87543-;received=67.84.215.54;rport=9168
From: <sip:RemoteBrian@bmy.gotdns.org>;tag=d33b6915
To: <sip:12016180234@bmy.gotdns.org>;tag=as06d5e6bb
Call-ID: 5615f77f064f0558
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12016180234@127.0.0.1>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 8032 8033 IN IP4 127.0.0.1
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 13202 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
[Nov  1 04:10:41] WARNING[8157]: chan_sip.c:1847 retrans_pkt: Maximum retries exceeded on transmission 5615f77f064f0558 for seqno 2 (Critical Response)
Really destroying SIP dialog '5615f77f064f0558' Method: INVITE
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
[Nov  1 04:11:11] NOTICE[8157]: chan_sip.c:6853 sip_reregister:    -- Re-registration for  17476016765@proxy01.sipphone.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 198.65.166.131:5060:
REGISTER sip:proxy01.sipphone.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2d2dabd5;rport
From: <sip:17476016765@proxy01.sipphone.com>;tag=as0a08979f
To: <sip:17476016765@proxy01.sipphone.com>
Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="17476016765", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4548651e9701dae1e90c3d21edfdf7284d7e4744", response="b615ce6c17abc0f4347c040c8d97120e", opaque=""
Expires: 120
Contact: <sip:s@127.0.0.1>
Event: registration
Content-Length: 0


---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2d2dabd5;rport=5060;received=67.83.110.168
From: <sip:17476016765@proxy01.sipphone.com>;tag=as0a08979f
To: <sip:17476016765@proxy01.sipphone.com>;tag=21a483426c2cd5d9b85bffe6bba40a2e.c0df
Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com
CSeq: 104 REGISTER
P-Behind-NAT: Yes
Contact: <sip:s@67.83.110.168:5060>;expires=120
Content-Length: 0


--- (9 headers 0 lines)---
Scheduling destruction of SIP dialog '2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com' in 32000 ms (Method: REGISTER)
[Nov  1 04:11:11] NOTICE[8157]: chan_sip.c:11675 handle_response_register: Outbound Registration: Expiry for proxy01.sipphone.com is 120 sec (Scheduling reregistration in 105 s)
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
[Nov  1 04:11:25] WARNING[8144]: chan_iax2.c:3788 iax2_send: No private structure for packet?
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
Really destroying SIP dialog '2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com' Method: REGISTER
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
[Nov  1 04:12:05] WARNING[8148]: chan_iax2.c:3788 iax2_send: No private structure for packet?
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
[Nov  1 04:12:25] WARNING[8141]: chan_iax2.c:3788 iax2_send: No private structure for packet?
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 67.84.215.54:9168:


--- (0 headers 0 lines) Nat keepalive ---
bmy*CLI>
<-- SIP read from 198.65.166.131:5060:

--- (0 headers 0 lines) Nat keepalive ---
[Nov  1 04:12:56] NOTICE[8157]: chan_sip.c:6853 sip_reregister:    -- Re-registration for  17476016765@proxy01.sipphone.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 198.65.166.131:5060:
REGISTER sip:proxy01.sipphone.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0d5c90a4;rport
From: <sip:17476016765@proxy01.sipphone.com>;tag=as14d3782b
To: <sip:17476016765@proxy01.sipphone.com>
Call-ID: 2f3118f0525176d848a1a5a1250aaa13@proxy01.sipphone.com
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="17476016765", realm="proxy01.sipphone.com", algorithm=MD5, uri="sip:proxy01.sipphone.com", nonce="4548651e9701dae1e90c3d21edfdf7284d7e4744", response="b615ce6c17abc0f4347c040c8d97120e", opaque=""
Expires: 120
Contact: <sip:s@127.0.0.1>
Event: registration
Content-Length: 0