rendered paste body[Jun 2 16:14:42]
<--- SIP read from UDP:192.168.5.237:5060 --->
INVITE sip:158@192.168.5.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>
Remote-Party-ID: 103 <sip:103@192.168.5.249>;screen=yes;party=calling
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 103 <sip:103@192.168.5.237:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.12
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 381093 381093 IN IP4 192.168.5.237
s=-
c=IN IP4 192.168.5.237
t=0 0
m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jun 2 16:14:42] --- (15 headers 20 lines) ---
[Jun 2 16:14:42] == Using UDPTL TOS bits 10
[Jun 2 16:14:42] == Using UDPTL CoS mark 5
[Jun 2 16:14:42] Sending to 192.168.5.237:5060 (no NAT)
[Jun 2 16:14:42] Using INVITE request as basis request - 70fb8f98-668ef86c@192.168.5.237
[Jun 2 16:14:42] Found peer '103' for '103' from 192.168.5.237:5060
[Jun 2 16:14:42]
<--- Reliably Transmitting (NAT) to 192.168.5.237:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c;received=192.168.5.237;rport=5060
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>;tag=as2cd4fb5f
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="mail", nonce="5bfd6462"
Content-Length: 0
<------------>
[Jun 2 16:14:42] Scheduling destruction of SIP dialog '70fb8f98-668ef86c@192.168.5.237' in 6400 ms (Method: INVITE)
[Jun 2 16:14:42]
<--- SIP read from UDP:192.168.5.237:5060 --->
ACK sip:158@192.168.5.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-3e243e7c
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>;tag=as2cd4fb5f
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 101 ACK
Max-Forwards: 70
Contact: 103 <sip:103@192.168.5.237:5060>
User-Agent: Linksys/SPA2102-5.2.12
Content-Length: 0
<------------->
[Jun 2 16:14:42] --- (10 headers 0 lines) ---
[Jun 2 16:14:42]
<--- SIP read from UDP:192.168.5.237:5060 --->
INVITE sip:158@192.168.5.249 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>
Remote-Party-ID: 103 <sip:103@192.168.5.249>;screen=yes;party=calling
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="103",realm="mail",nonce="5bfd6462",uri="sip:158@192.168.5.249",algorithm=MD5,response="66046d0057cc44a3f9878f502d641469"
Contact: 103 <sip:103@192.168.5.237:5060>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.12
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 381093 381093 IN IP4 192.168.5.237
s=-
c=IN IP4 192.168.5.237
t=0 0
m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jun 2 16:14:42] --- (16 headers 20 lines) ---
[Jun 2 16:14:42] Sending to 192.168.5.237:5060 (NAT)
[Jun 2 16:14:42] Using INVITE request as basis request - 70fb8f98-668ef86c@192.168.5.237
[Jun 2 16:14:42] Found peer '103' for '103' from 192.168.5.237:5060
[Jun 2 16:14:42] == Using SIP RTP TOS bits 10
[Jun 2 16:14:42] == Using SIP RTP CoS mark 5
[Jun 2 16:14:42] Found RTP audio format 8
[Jun 2 16:14:42] Found RTP audio format 0
[Jun 2 16:14:42] Found RTP audio format 18
[Jun 2 16:14:42] Found RTP audio format 2
[Jun 2 16:14:42] Found RTP audio format 4
[Jun 2 16:14:42] Found RTP audio format 96
[Jun 2 16:14:42] Found RTP audio format 97
[Jun 2 16:14:42] Found RTP audio format 98
[Jun 2 16:14:42] Found RTP audio format 100
[Jun 2 16:14:42] Found RTP audio format 101
[Jun 2 16:14:42] Found audio description format PCMA for ID 8
[Jun 2 16:14:42] Found audio description format PCMU for ID 0
[Jun 2 16:14:42] Found audio description format G729a for ID 18
[Jun 2 16:14:42] Found audio description format G726-32 for ID 2
[Jun 2 16:14:42] Found audio description format G723 for ID 4
[Jun 2 16:14:42] Found audio description format G726-40 for ID 96
[Jun 2 16:14:42] Found audio description format G726-24 for ID 97
[Jun 2 16:14:42] Found audio description format G726-16 for ID 98
[Jun 2 16:14:42] Found audio description format NSE for ID 100
[Jun 2 16:14:42] Found audio description format telephone-event for ID 101
[Jun 2 16:14:42] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90d (g723|ulaw|alaw|g726|g729)
[Jun 2 16:14:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 2 16:14:42] Peer audio RTP is at port 192.168.5.237:16480
[Jun 2 16:14:42] Looking for 158 in office1 (domain 192.168.5.249)
[Jun 2 16:14:42] list_route: hop: <sip:103@192.168.5.237:5060>
[Jun 2 16:14:42]
<--- Transmitting (NAT) to 192.168.5.237:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220;received=192.168.5.237;rport=5060
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:158@192.168.5.249:5060>
Content-Length: 0
<------------>
[Jun 2 16:14:42] -- Executing [158@office1:1] Goto("SIP/103-00000063", "fax,s,1") in new stack
[Jun 2 16:14:42] -- Goto (fax,s,1)
[Jun 2 16:14:42] -- Executing [s@fax:1] Answer("SIP/103-00000063", "") in new stack
[Jun 2 16:14:42] Audio is at 5060
[Jun 2 16:14:42] Adding codec 0x8 (alaw) to SDP
[Jun 2 16:14:42] Adding codec 0x4 (ulaw) to SDP
[Jun 2 16:14:42] Adding codec 0x100 (g729) to SDP
[Jun 2 16:14:42] Adding codec 0x800 (g726) to SDP
[Jun 2 16:14:42] Adding codec 0x1 (g723) to SDP
[Jun 2 16:14:42] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 2 16:14:42]
<--- Reliably Transmitting (NAT) to 192.168.5.237:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-2adbd220;received=192.168.5.237;rport=5060
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>;tag=as334cec94
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:158@192.168.5.249:5060>
Content-Type: application/sdp
Content-Length: 380
v=0
o=root 1240040461 1240040461 IN IP4 192.168.5.249
s=Asterisk PBX 1.8.4.1
c=IN IP4 192.168.5.249
t=0 0
m=audio 19098 RTP/AVP 8 0 18 2 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[Jun 2 16:14:42]
<--- SIP read from UDP:192.168.5.237:5060 --->
ACK sip:158@192.168.5.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.237:5060;branch=z9hG4bK-26845788
From: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
To: <sip:158@192.168.5.249>;tag=as334cec94
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="103",realm="mail",nonce="5bfd6462",uri="sip:158@192.168.5.249",algorithm=MD5,response="66046d0057cc44a3f9878f502d641469"
Contact: 103 <sip:103@192.168.5.237:5060>
User-Agent: Linksys/SPA2102-5.2.12
Content-Length: 0
<------------->
[Jun 2 16:14:42] --- (11 headers 0 lines) ---
[Jun 2 16:14:42] -- Executing [s@fax:2] Set("SIP/103-00000063", "FAXFILE=/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack
[Jun 2 16:14:42] -- Executing [s@fax:3] ReceiveFAX("SIP/103-00000063", "/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack
[Jun 2 16:14:42] -- Channel 'SIP/103-00000063' receiving FAX '/mnt/fax_kw12/fax_2011-06-02_16-14-42.tif'
[Jun 2 16:14:45] set_destination: Parsing <sip:103@192.168.5.237:5060> for address/port to send to
[Jun 2 16:14:45] set_destination: set destination to 192.168.5.237:5060
[Jun 2 16:14:45] Reliably Transmitting (NAT) to 192.168.5.237:5060:
INVITE sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK2b03b0bb;rport
Max-Forwards: 70
From: <sip:158@192.168.5.249>;tag=as334cec94
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
Contact: <sip:158@192.168.5.249:5060>
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 1240040461 1240040462 IN IP4 192.168.5.249
s=Asterisk PBX 1.8.4.1
c=IN IP4 192.168.5.249
t=0 0
m=image 4155 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
---
[Jun 2 16:14:45]
<--- SIP read from UDP:192.168.5.237:5060 --->
SIP/2.0 200 OK
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
From: <sip:158@192.168.5.249>;tag=as334cec94
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK2b03b0bb
Contact: 103 <sip:103@192.168.5.237:5060>
Server: Linksys/SPA2102-5.2.12
Content-Length: 267
Content-Type: application/sdp
v=0
o=- 381405 381405 IN IP4 192.168.5.237
s=-
c=IN IP4 192.168.5.237
t=0 0
m=image 16480 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
[Jun 2 16:14:45] --- (10 headers 12 lines) ---
[Jun 2 16:14:45] Got T.38 offer in SDP in dialog 70fb8f98-668ef86c@192.168.5.237
[Jun 2 16:14:46] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
[Jun 2 16:14:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jun 2 16:14:46] Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Jun 2 16:14:46] set_destination: Parsing <sip:103@192.168.5.237:5060> for address/port to send to
[Jun 2 16:14:46] set_destination: set destination to 192.168.5.237:5060
[Jun 2 16:14:46] Transmitting (NAT) to 192.168.5.237:5060:
ACK sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK059e67fa;rport
Max-Forwards: 70
From: <sip:158@192.168.5.249>;tag=as334cec94
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
Contact: <sip:158@192.168.5.249:5060>
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.4.1
Content-Length: 0
---
[Jun 2 16:14:53] Reliably Transmitting (NAT) to 192.168.5.237:5060:
OPTIONS sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK7c9c6d99;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.249>;tag=as1c9981c6
To: <sip:103@192.168.5.237:5060>
Contact: <sip:asterisk@192.168.5.249:5060>
Call-ID: 0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.4.1
Date: Thu, 02 Jun 2011 09:14:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Jun 2 16:14:53]
<--- SIP read from UDP:192.168.5.237:5060 --->
SIP/2.0 200 OK
To: <sip:103@192.168.5.237:5060>;tag=9304557d408e15a1i0
From: "asterisk" <sip:asterisk@192.168.5.249>;tag=as1c9981c6
Call-ID: 0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK7c9c6d99
Server: Linksys/SPA2102-5.2.12
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
<------------->
[Jun 2 16:14:53] --- (10 headers 0 lines) ---
[Jun 2 16:14:53] Really destroying SIP dialog '0e4b62b847063ceb60ecc2452e3c3ecf@192.168.5.249:5060' Method: OPTIONS
[Jun 2 16:14:53] NOTICE[19931]: chan_sip.c:12419 sip_reregister: -- Re-registration for 3832091931@195.189.239.32
[Jun 2 16:14:53] > doing dnsmgr_lookup for '195.189.239.32'
[Jun 2 16:14:53] > doing dnsmgr_lookup for '195.189.239.32'
[Jun 2 16:14:53] NOTICE[19931]: chan_sip.c:19815 handle_response_register: Outbound Registration: Expiry for 195.189.239.32 is 120 sec (Scheduling reregistration in 105 s)
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
pbx*CLI>
[Jun 2 16:15:25] set_destination: Parsing <sip:103@192.168.5.237:5060> for address/port to send to
[Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060
[Jun 2 16:15:25] Audio is at 5060
[Jun 2 16:15:25] Adding codec 0x8 (alaw) to SDP
[Jun 2 16:15:25] Adding codec 0x4 (ulaw) to SDP
[Jun 2 16:15:25] Adding codec 0x100 (g729) to SDP
[Jun 2 16:15:25] Adding codec 0x800 (g726) to SDP
[Jun 2 16:15:25] Adding codec 0x1 (g723) to SDP
[Jun 2 16:15:25] Adding non-codec 0x1 (telephone-event) to SDP
[Jun 2 16:15:25] Reliably Transmitting (NAT) to 192.168.5.237:5060:
INVITE sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK0221c7da;rport
Max-Forwards: 70
From: <sip:158@192.168.5.249>;tag=as334cec94
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
Contact: <sip:158@192.168.5.249:5060>
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 380
v=0
o=root 1240040461 1240040463 IN IP4 192.168.5.249
s=Asterisk PBX 1.8.4.1
c=IN IP4 192.168.5.249
t=0 0
m=audio 19098 RTP/AVP 8 0 18 2 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Jun 2 16:15:25]
<--- SIP read from UDP:192.168.5.237:5060 --->
SIP/2.0 200 OK
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
From: <sip:158@192.168.5.249>;tag=as334cec94
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK0221c7da
Contact: 103 <sip:103@192.168.5.237:5060>
Server: Linksys/SPA2102-5.2.12
Content-Length: 255
Content-Type: application/sdp
v=0
o=- 385325 385325 IN IP4 192.168.5.237
s=-
c=IN IP4 192.168.5.237
t=0 0
m=audio 16480 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jun 2 16:15:25] --- (10 headers 13 lines) ---
[Jun 2 16:15:25] Found RTP audio format 8
[Jun 2 16:15:25] Found RTP audio format 100
[Jun 2 16:15:25] Found RTP audio format 101
[Jun 2 16:15:25] Found audio description format PCMA for ID 8
[Jun 2 16:15:25] Found audio description format NSE for ID 100
[Jun 2 16:15:25] Found audio description format telephone-event for ID 101
[Jun 2 16:15:25] Capabilities: us - 0x90d (g723|ulaw|alaw|g726|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Jun 2 16:15:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jun 2 16:15:25] Peer audio RTP is at port 192.168.5.237:16480
[Jun 2 16:15:25] set_destination: Parsing <sip:103@192.168.5.237:5060> for address/port to send to
[Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060
[Jun 2 16:15:25] Transmitting (NAT) to 192.168.5.237:5060:
ACK sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK072adc8b;rport
Max-Forwards: 70
From: <sip:158@192.168.5.249>;tag=as334cec94
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
Contact: <sip:158@192.168.5.249:5060>
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.4.1
Content-Length: 0
---
[Jun 2 16:15:25] -- Auto fallthrough, channel 'SIP/103-00000063' status is 'UNKNOWN'
[Jun 2 16:15:25] -- Executing [h@fax:1] System("SIP/103-00000063", "chmod 777 /mnt/fax_kw12/fax_2011-06-02_16-14-42.tif") in new stack
[Jun 2 16:15:25] -- Executing [h@fax:2] NoOp("SIP/103-00000063", "SUCCESS") in new stack
[Jun 2 16:15:25] -- Executing [h@fax:3] NoOp("SIP/103-00000063", "") in new stack
[Jun 2 16:15:25] Scheduling destruction of SIP dialog '70fb8f98-668ef86c@192.168.5.237' in 6400 ms (Method: ACK)
[Jun 2 16:15:25] set_destination: Parsing <sip:103@192.168.5.237:5060> for address/port to send to
[Jun 2 16:15:25] set_destination: set destination to 192.168.5.237:5060
[Jun 2 16:15:25] Reliably Transmitting (NAT) to 192.168.5.237:5060:
BYE sip:103@192.168.5.237:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK267a0702;rport
Max-Forwards: 70
From: <sip:158@192.168.5.249>;tag=as334cec94
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.4.1
Proxy-Authorization: Digest username="103", realm="mail", algorithm=MD5, uri="192.168.5.249", nonce="", response="0a86260e8deb20fa8b193b050fcf6aaf"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Jun 2 16:15:25]
<--- SIP read from UDP:192.168.5.237:5060 --->
SIP/2.0 200 OK
To: 103 <sip:103@192.168.5.249>;tag=f5ec85a8451bf99co0
From: <sip:158@192.168.5.249>;tag=as334cec94
Call-ID: 70fb8f98-668ef86c@192.168.5.237
CSeq: 104 BYE
Via: SIP/2.0/UDP 192.168.5.249:5060;branch=z9hG4bK267a0702
Server: Linksys/SPA2102-5.2.12
Content-Length: 0
<------------->
[Jun 2 16:15:25] --- (8 headers 0 lines) ---
[Jun 2 16:15:25] Really destroying SIP dialog '70fb8f98-668ef86c@192.168.5.237' Method: ACK