I didn’t realize that I had to re-enter those commands once I restarted freepbx.
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c:
<--- SIP read from UDP:192.168.1.101:51543 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-6215b97cb10bde1a-1---d8754z-;rport
Max-Forwards: 70
Contact:
To:
From: "508";tag=1a6aeb04
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Content-Length: 408
v=0
o=3cxVCE 366547740 260319015 IN IP4 192.168.1.101
s=3cxVCE Audio Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 40030 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 192.168.1.101
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: — (13 headers 18 lines) —
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Sending to 192.168.1.101:51543 (NAT)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Using INVITE request as basis request - NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found peer ‘508’ for ‘508’ from 192.168.1.101:51543
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.101:51543 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-6215b97cb10bde1a-1—d8754z-;received=192.168.1.101;rport=51543
From: "508"sip:[email protected]:5060;tag=1a6aeb04
To: sip:[email protected]:5060;tag=as300100af
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 1 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0a030e25"
Content-Length: 0
<------------>
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Scheduling destruction of SIP dialog ‘NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.’ in 6720 ms (Method: INVITE)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c:
<— SIP read from UDP:192.168.1.101:51543 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-6215b97cb10bde1a-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected]:5060;tag=as300100af
From: "508"sip:[email protected]:5060;tag=1a6aeb04
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 1 ACK
Content-Length: 0
<------------->
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: — (8 headers 0 lines) —
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c:
<— SIP read from UDP:192.168.1.101:51543 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-507c532612351e01-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:51543;rinstance=145542dd98b73be4
To: sip:[email protected]:5060
From: "508"sip:[email protected]:5060;tag=1a6aeb04
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.20943.0
Authorization: Digest username=“508”,realm=“asterisk”,nonce=“0a030e25”,uri=“sip:[email protected]:5060”,response=“2bacb0a3be15f7d365435632185a7950”,algorithm=MD5
Content-Length: 408
v=0
o=3cxVCE 366547740 260319015 IN IP4 192.168.1.101
s=3cxVCE Audio Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 40030 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 192.168.1.101
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
<------------->
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: — (14 headers 18 lines) —
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Sending to 192.168.1.101:51543 (no NAT)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Using INVITE request as basis request - NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found peer ‘508’ for ‘508’ from 192.168.1.101:51543
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found RTP audio format 0
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found RTP audio format 8
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found RTP audio format 3
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found RTP audio format 101
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found audio description format PCMU for ID 0
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found audio description format PCMA for ID 8
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found audio description format GSM for ID 3
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found RTP video format 34
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Found video description format H263 for ID 34
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Peer audio RTP is at port 192.168.1.101:40030
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: Looking for 00967711532426 in from-internal (domain 192.168.1.103)
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c: list_route: hop: sip:[email protected]:51543;rinstance=145542dd98b73be4
[2012-07-17 13:42:29] VERBOSE[2884] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.101:51543 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-507c532612351e01-1—d8754z-;received=192.168.1.101;rport=51543
From: "508"sip:[email protected]:5060;tag=1a6aeb04
To: sip:[email protected]:5060
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Audio is at 16858
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-07-17 13:42:29] VERBOSE[3232] chan_sip.c: Reliably Transmitting (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:42:30] VERBOSE[2884] chan_sip.c: Retransmitting #1 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:42:31] VERBOSE[2884] chan_sip.c: Retransmitting #2 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:42:33] VERBOSE[2884] chan_sip.c: Retransmitting #3 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:42:34] VERBOSE[2884] chan_sip.c:
<— SIP read from UDP:192.168.1.101:51543 —>
<------------->
[2012-07-17 13:42:37] VERBOSE[2884] chan_sip.c: Retransmitting #4 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:42:42] VERBOSE[2884] chan_sip.c: Really destroying SIP dialog ‘OGZjNDcxOTFmNWQyYjk0OWQzNmUxZjNmYTJjYTM3ZDU.’ Method: REGISTER
[2012-07-17 13:42:45] VERBOSE[2884] chan_sip.c: Retransmitting #5 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:43:01] VERBOSE[2884] chan_sip.c: Retransmitting #6 (NAT) to 77.72.169.131:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 109.200.166.102:5060;branch=z9hG4bK73823a2c;rport
Max-Forwards: 70
From: “967712306000” sip:[email protected];tag=as7a9555c7
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 17 Jul 2012 10:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290
v=0
o=root 1802681376 1802681376 IN IP4 109.200.166.102
s=Asterisk PBX 1.8.13.0
c=IN IP4 109.200.166.102
t=0 0
m=audio 16858 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-07-17 13:43:01] VERBOSE[2884] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 32000 ms (Method: INVITE)
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Audio is at 19676
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-07-17 13:43:01] VERBOSE[3232] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.101:51543 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-507c532612351e01-1—d8754z-;received=192.168.1.101;rport=51543
From: "508"sip:[email protected]:5060;tag=1a6aeb04
To: sip:[email protected]:5060;tag=as4b4f3d70
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 308
v=0
o=root 1762514991 1762514991 IN IP4 192.168.1.103
s=Asterisk PBX 1.8.13.0
c=IN IP4 192.168.1.103
t=0 0
m=audio 19676 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
<------------>
[2012-07-17 13:43:01] WARNING[2884] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32003ms with no response
[2012-07-17 13:43:01] VERBOSE[2884] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
[2012-07-17 13:43:04] VERBOSE[2884] chan_sip.c:
<— SIP read from UDP:192.168.1.101:51543 —>
<------------->
[2012-07-17 13:43:05] VERBOSE[3232] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.101:51543 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-507c532612351e01-1—d8754z-;received=192.168.1.101;rport=51543
From: "508"sip:[email protected]:5060;tag=1a6aeb04
To: sip:[email protected]:5060;tag=as4b4f3d70
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2012-07-17 13:43:05] WARNING[3232] channel.c: Prodding channel ‘SIP/508-00000008’ failed
[2012-07-17 13:43:05] VERBOSE[2884] chan_sip.c:
<— SIP read from UDP:192.168.1.101:51543 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:51543;branch=z9hG4bK-d8754z-507c532612351e01-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected]:5060;tag=as4b4f3d70
From: "508"sip:[email protected]:5060;tag=1a6aeb04
Call-ID: NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.
CSeq: 2 ACK
Content-Length: 0
<------------->
[2012-07-17 13:43:05] VERBOSE[2884] chan_sip.c: — (8 headers 0 lines) —
[2012-07-17 13:43:05] VERBOSE[2884] chan_sip.c: Really destroying SIP dialog ‘NDYyNjgzYjFkODY2NTU1MGFjMjc0MGI1ODM2ZjMwNjM.’ Method: ACK