rendered paste bodyasterisk = 10.0.0.1
firewall = 10.0.0.2
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
asterisk*CLI> sip debug peer 358968910308
SIP Debugging Enabled for IP: 10.0.0.1:24311
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.1:24311:
OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK04bbca72
From: "asterisk" <sip:asterisk@10.0.0.2>;tag=as4903b7e3
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:asterisk@10.0.0.2>
Call-ID: 084ae87e0037879369444e5f7b412db1@10.0.0.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
---
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
We're at 10.0.0.2 port 12794
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x100 (g729)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x2 (gsm)
12 headers, 12 lines
Reliably Transmitting (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
Retransmitting #1 (no NAT) to 10.0.0.1:24311:
OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK04bbca72
From: "asterisk" <sip:asterisk@10.0.0.2>;tag=as4903b7e3
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:asterisk@10.0.0.2>
Call-ID: 084ae87e0037879369444e5f7b412db1@10.0.0.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
---
Retransmitting #1 (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
Destroying call '084ae87e0037879369444e5f7b412db1@10.0.0.2'
Reliably Transmitting (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Scheduling destruction of call '63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2' in 15000 ms
Retransmitting #2 (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
We're at 10.0.0.2 port 12934
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x100 (g729)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x2 (gsm)
12 headers, 12 lines
Reliably Transmitting (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12934 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
To: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 102 INVITE
Server: X-Lite release 1105x
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 10.0.0.1:24311:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
To: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 102 INVITE
Server: X-Lite release 1105x
Content-Length: 0
--- (9 headers 0 lines)---
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
Retransmitting #3 (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
Retransmitting #1 (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Retransmitting #4 (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
Retransmitting #2 (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
BYE sip:097537565@10.0.0.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:24311;rport;branch=z9hG4bK761E4AF4F79F48CA854C3F6E96078EE9
From: <sip:358968910308@10.0.0.1:24311>;tag=1990678324
To: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 30485 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 10.0.0.1 : 24311 (non-NAT)
Transmitting (no NAT) to 10.0.0.1:24311:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:24311;branch=z9hG4bK761E4AF4F79F48CA854C3F6E96078EE9
From: <sip:358968910308@10.0.0.1:24311>;tag=1990678324
To: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 30485 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:097537565@10.0.0.2>
Content-Length: 0
---
Retransmitting #5 (no NAT) to 10.0.0.1:24311:
INVITE sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 15913 15913 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 12794 RTP/AVP 0 8 18 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
---
Retransmitting #3 (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Oct 5 12:35:37 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call
63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 for seqno 102 (Critical Request)
Retransmitting #4 (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Retransmitting #5 (no NAT) to 10.0.0.1:24311:
CANCEL sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK11efe807;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as61e25645
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:097537565@10.0.0.2>
Call-ID: 63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
---
Oct 5 12:35:39 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call
63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2 for seqno 102 (Non-critical Request)
Destroying call '63da4e2d0435d3ab4c1cfe57673b1667@10.0.0.2'
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK447719b4;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
To: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105x
Content-Length: 316
v=0
o=358968910308 269908784 269915575 IN IP4 10.0.0.1
s=X-Lite
c=IN IP4 10.0.0.1
t=0 0
m=audio 26432 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.1:26432
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing),
combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
list_route: hop: <sip:358968910308@10.0.0.1:24311>
set_destination: Parsing <sip:358968910308@10.0.0.1:24311> for address/port to send to
set_destination: set destination to 10.0.0.1, port 24311
Transmitting (no NAT) to 10.0.0.1:24311:
ACK sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK5449bb8b;rport
From: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
To: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
Contact: <sip:097537565@10.0.0.2>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.1:24311:
OPTIONS sip:358968910308@10.0.0.1:24311 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK06992209
From: "asterisk" <sip:asterisk@10.0.0.2>;tag=as4949058b
To: <sip:358968910308@10.0.0.1:24311>
Contact: <sip:asterisk@10.0.0.2>
Call-ID: 29f507a30c0a46993e7c49b44b0223a0@10.0.0.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Thu, 05 Oct 2006 09:35:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
---
<-- SIP read from 10.0.0.1:24311:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK06992209
From: "asterisk" <sip:asterisk@10.0.0.2>;tag=as4949058b
To: <sip:358968910308@10.0.0.1:24311>;tag=2987254751
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 29f507a30c0a46993e7c49b44b0223a0@10.0.0.2
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY
CSeq: 102 OPTIONS
Server: X-Lite release 1105x
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '29f507a30c0a46993e7c49b44b0223a0@10.0.0.2'
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
Oct 5 12:35:46 WARNING[15913]: chan_sip.c:860 retrans_pkt: Maximum retries exceeded on call
002B-007F-4FD55E00-99@81.29.134.169 for seqno 229155361 (Non-critical Response)
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
BYE sip:097537565@10.0.0.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:24311;rport;branch=z9hG4bKC9BDCBA5F2C64D409CDF1D0E43A91C4F
From: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
To: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
Contact: <sip:358968910308@10.0.0.1:24311>
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 21636 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 10.0.0.1 : 24311 (non-NAT)
Transmitting (no NAT) to 10.0.0.1:24311:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:24311;branch=z9hG4bKC9BDCBA5F2C64D409CDF1D0E43A91C4F
From: <sip:358968910308@10.0.0.1:24311>;tag=3018104732
To: "asreceived" <sip:097537565@10.0.0.2>;tag=as52a969b3
Call-ID: 2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2
CSeq: 21636 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:097537565@10.0.0.2>
Content-Length: 0
---
Destroying call '2f15a05b0d561e835a7adfc8786fbe54@10.0.0.2'
asterisk*CLI>
<-- SIP read from 10.0.0.1:24311:
--- (0 headers 0 lines) Nat keepalive ---
asterisk*CLI> sip no debug
SIP Debugging Disabled
asterisk*CLI> exit
[root@asterisk asterisk]#