[Asterisk-Dev] Re: 482 Loop Detected problem

snacktime snacktime at gmail.com
Tue Dec 6 00:44:10 MST 2005 

I ran into this same "482 loop detected" message this evening when
reconfiguring my asterisk box at home.

Asterisk is behind a nat and my phone is on a sipura 3000.  Placing a
call from the sipura 3000 to my forward number (630362) should result
in fwd sending the call back to the 's' extension in asterisk. 
However asterisk gives a 482 error and fwd plays me the unavailable
message.  I could be missing something but I don't see where there
could be a loop here.

The sipura is registered as user homeline1 with callerid 10 and is on
ip 192.168.1.47.  The asterisk server's ip is 192.168.1.25 and is
behind a static nat.  The outside ip is 71.112.7.19.  Following is the
full debug log.

Connected to Asterisk 1.2.0 currently running on debian (pid = 20861)
Verbosity is at least 8
debian*CLI> sip debug
SIP Debugging enabled
debian*CLI>
<-- SIP read from 192.168.1.47:5060:
INVITE sip:393630362 at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-cd13c83b
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>
Remote-Party-ID: <sip:homeline1 at 192.168.1.25>;screen=yes;party=calling
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:homeline1 at 192.168.1.47:5060>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 544577 544577 IN IP4 192.168.1.47
s=-
c=IN IP4 192.168.1.47
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 19 lines)---
Using INVITE request as basis request - 51993ea1-247feb93 at 192.168.1.47
Sending to 192.168.1.47 : 5060 (NAT)
Reliably Transmitting (no NAT) to 192.168.1.47:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-cd13c83b;received=192.168.1.47
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as09c916ea
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:393630362 at 192.168.1.25>
Proxy-Authenticate: Digest realm="asterisk", nonce="04789a28"
Content-Length: 0


---
Scheduling destruction of call '51993ea1-247feb93 at 192.168.1.47' in 15000 ms
Found user 'homeline1'
debian*CLI>
<-- SIP read from 192.168.1.47:5060:
ACK sip:393630362 at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-cd13c83b
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as09c916ea
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:homeline1 at 192.168.1.47:5060>
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (10 headers 0 lines)---
debian*CLI>
<-- SIP read from 192.168.1.47:5060:
INVITE sip:393630362 at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-1e1579db
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>
Remote-Party-ID: <sip:homeline1 at 192.168.1.25>;screen=yes;party=calling
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest

username="homeline1",realm="asterisk",nonce="04789a28",uri="sip:393630362 at 192.168.1.25",algorithm=MD5,response="a4ba941dfd49e

7396035fc5424e46fe3"
Contact: <sip:homeline1 at 192.168.1.47:5060>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 544577 544577 IN IP4 192.168.1.47
s=-
c=IN IP4 192.168.1.47
t=0 0
m=audio 16388 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (16 headers 19 lines)---
Using INVITE request as basis request - 51993ea1-247feb93 at 192.168.1.47
Sending to 192.168.1.47 : 5060 (non-NAT)
Found user 'homeline1'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.47:16388
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 393630362 in __home__from-local__ (domain 192.168.1.25)
list_route: hop: <sip:homeline1 at 192.168.1.47:5060>
Transmitting (no NAT) to 192.168.1.47:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-1e1579db;received=192.168.1.47
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:393630362 at 192.168.1.25>
Content-Length: 0


---
    -- Executing Dial("SIP/homeline1-3669", "SIP/fwd/630362") in new stack
We're at 71.112.7.19 port 15248
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (NAT) to 69.90.155.70:5060:
INVITE sip:630362 at fwd.pulver.com SIP/2.0
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK0a5a0c11;rport
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>
Contact: <sip:10 at 71.112.7.19>
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 06 Dec 2005 07:24:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 21200 21200 IN IP4 71.112.7.19
s=session
c=IN IP4 71.112.7.19
t=0 0
m=audio 15248 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called fwd/630362
debian*CLI>
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK0a5a0c11;rport=5060
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0


--- (8 headers 0 lines)---
debian*CLI>
<-- SIP read from 69.90.155.70:5060:
INVITE sip:s at 71.112.7.19 SIP/2.0
Record-Route: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bK9f64.3fe71114.1
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK0a5a0c11;rport=5060
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>
Contact: <sip:10 at 71.112.7.19>
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Tue, 06 Dec 2005 07:24:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 21200 21200 IN IP4 71.112.7.19
s=session
c=IN IP4 71.112.7.19
t=0 0
m=audio 15248 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (15 headers 10 lines)---
Transmitting (NAT) to 69.90.155.70:5060:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP
69.90.155.70;branch=z9hG4bK9f64.3fe71114.1;received=69.90.155.70
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK0a5a0c11;rport=5060
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>;tag=as68fab5d0
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:10 at 71.112.7.19>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
debian*CLI>
<-- SIP read from 69.90.155.70:5060:
ACK sip:s at 71.112.7.19 SIP/2.0
Via: SIP/2.0/UDP 69.90.155.70;branch=z9hG4bK9f64.3fe71114.1
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
To: <sip:630362 at fwd.pulver.com>;tag=as68fab5d0
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14-6 (i386/linux))
Content-Length: 0


--- (8 headers 0 lines)---
debian*CLI>
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK0a5a0c11
Record-Route: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>;tag=as444c6dce
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:630362 at 69.90.168.15:5028>
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 9749 9749 IN IP4 69.90.168.15
s=session
c=IN IP4 69.90.168.15
t=0 0
m=audio 17730 RTP/AVP 0 8 3 4 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 69.90.168.15:17730
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G723
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x40f
(g723|gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
set_destination: Parsing
<sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on> for address/port to
send to
set_destination: set destination to 69.90.155.70, port 5060
Transmitting (NAT) to 69.90.155.70:5060:
ACK sip:630362 at 69.90.168.15:5028 SIP/2.0
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK42fb3ae3;rport
Route: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>;tag=as444c6dce
Contact: <sip:10 at 71.112.7.19>
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/fwd-3847 answered SIP/homeline1-3669
We're at 192.168.1.25 port 17650
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.47:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-1e1579db;received=192.168.1.47
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as1a0a3397
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:393630362 at 192.168.1.25>
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 21200 21200 IN IP4 192.168.1.25
s=session
c=IN IP4 192.168.1.25
t=0 0
m=audio 17650 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/homeline1-3669 and SIP/fwd-3847
debian*CLI>
<-- SIP read from 192.168.1.47:5060:
ACK sip:393630362 at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-c31959c9
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as1a0a3397
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest

username="homeline1",realm="asterisk",nonce="04789a28",uri="sip:393630362 at 192.168.1.25",algorithm=MD5,response="3e2258c6f36b5

b476de2aadfe3e56abd"
Contact: <sip:homeline1 at 192.168.1.47:5060>
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0



---
Destroying call '1793054c-99f70e72 at 10.139.10.180'
debian*CLI> sip no debug
<-- SIP read from 192.168.1.47:5060:
NOTIFY sip:192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-cd64e1e
From: <sip:homeline1 at 192.168.1.25>;tag=9b8326991ffa15fao0
To: <sip:192.168.1.25>
Call-ID: df859929-560fc4aa at 192.168.1.47
CSeq: 363 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (10 headers 0 lines)---
Transmitting (NAT) to 192.168.1.47:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-cd64e1e;received=192.168.1.47
From: <sip:homeline1 at 192.168.1.25>;tag=9b8326991ffa15fao0
To: <sip:192.168.1.25>;tag=as4381e296
Call-ID: df859929-560fc4aa at 192.168.1.47
CSeq: 363 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
Destroying call 'df859929-560fc4aa at 192.168.1.47'
debian*CLI> sip no debug
<-- SIP read from 192.168.1.47:5061:
NOTIFY sip:192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5061;branch=z9hG4bK-33175dcd
From: <sip:homepstn at 192.168.1.25>;tag=fbb5c9a14c8c8e92o1
To: <sip:192.168.1.25>
Call-ID: f4371d31-f8c8d042 at 192.168.1.47
CSeq: 363 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (10 headers 0 lines)---
Transmitting (NAT) to 192.168.1.47:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.47:5061;branch=z9hG4bK-33175dcd;received=192.168.1.47
From: <sip:homepstn at 192.168.1.25>;tag=fbb5c9a14c8c8e92o1
To: <sip:192.168.1.25>;tag=as013f78c6
Call-ID: f4371d31-f8c8d042 at 192.168.1.47
CSeq: 363 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
Destroying call 'f4371d31-f8c8d042 at 192.168.1.47'
debian*CLI> sip no debug
<-- SIP read from 192.168.1.47:5060:
BYE sip:393630362 at 192.168.1.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-e9fe7aba
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as1a0a3397
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest

username="homeline1",realm="asterisk",nonce="04789a28",uri="sip:393630362 at 192.168.1.25",algorithm=MD5,response="b680e0712f07f

16d4c672f8cd7a32720"
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 192.168.1.47 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.47:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK-e9fe7aba;received=192.168.1.47
From: <sip:homeline1 at 192.168.1.25>;tag=c11ed349f331f0bo0
To: <sip:393630362 at 192.168.1.25>;tag=as1a0a3397
Call-ID: 51993ea1-247feb93 at 192.168.1.47
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:393630362 at 192.168.1.25>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing
<sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on> for address/port to
send to
set_destination: set destination to 69.90.155.70, port 5060
Reliably Transmitting (NAT) to 69.90.155.70:5060:
BYE sip:630362 at 69.90.168.15:5028 SIP/2.0
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK4d6f9dcf;rport
Route: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>;tag=as444c6dce
Contact: <sip:10 at 71.112.7.19>
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (__home__from-local__, 393630362, 1) exited
non-zero on 'SIP/homeline1-3669'
debian*CLI> sip no debug
<-- SIP read from 69.90.155.70:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 71.112.7.19:5060;branch=z9hG4bK4d6f9dcf
Record-Route: <sip:630362 at 69.90.155.70;ftag=as68fab5d0;lr=on>
From: "10" <sip:10 at 71.112.7.19>;tag=as68fab5d0
To: <sip:630362 at fwd.pulver.com>;tag=as444c6dce
Call-ID: 1442573035f0a29800adf46b263ea889 at 71.112.7.19
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:630362 at 69.90.168.15:5028>
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '1442573035f0a29800adf46b263ea889 at 71.112.7.19'
debian*CLI> sip no debug

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