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ถามปัญหาการส่ง videocall ครับ

Asterisk Opensource IP Pbx
ตอบกระทู้

Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 01 พ.ค. 2011 16:08

ผมลองหาโหลด app ตัวอื่นมาไปเจอ Antisip เลยลองเอามาทดสอบ ปรากฏว่า ใช้ app ตัวนี้ มือถือสามารถรับภาพจาก x-lite ที่ส่งมาได้ แต่ sipdroid ผมพยายามแก้โคดยังไงก็ยังไม่สามารถ รับภาพที่ x-lite ส่งมาได้ เลยอยากให้ช่วยดู log ให้หน่อยครับว่ามีตรงไหนที่พอจะ รู้ถึงสาเหตุได้มั้งครับ :D
-----------------------------------------------------------------------------------------------
INVITE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1765233779
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>
Call-ID: 1165902499
CSeq: 20 INVITE
Contact: <sip:1004@192.168.1.5:5060>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, UPDATE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Subject: Talk
Session-Expires: 90
Supported: timer, 100rel, replaces
Content-Length: 241

v=0
o=amsip 0 0 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 18066 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=setup:passive

<------------->
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: --- (15 headers 12 lines) ---
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using SIP RTP TOS bits 184
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using SIP RTP CoS mark 5
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using SIP VRTP TOS bits 136
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using SIP VRTP CoS mark 6
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using UDPTL TOS bits 184
[May 1 15:56:04] VERBOSE[2898] netsock.c: == Using UDPTL CoS mark 5
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Using INVITE request as basis request - 1165902499
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found peer '1004' for '1004' from 192.168.1.5:5060
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1765233779;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as5fa01dae
Call-ID: 1165902499
CSeq: 20 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="745f1eb6"
Content-Length: 0


<------------>
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Scheduling destruction of SIP dialog '1165902499' in 6400 ms (Method: INVITE)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
jaK
<------------->
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1765233779
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as5fa01dae
Call-ID: 1165902499
CSeq: 20 ACK
Content-Length: 0


<------------->
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: --- (7 headers 0 lines) ---
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
INVITE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1964029822
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>
Call-ID: 1165902499
CSeq: 21 INVITE
Contact: <sip:1004@192.168.1.5:5060>
Authorization: Digest username="1004", realm="asterisk", nonce="745f1eb6", uri="sip:1003@192.168.1.150", response="8f18c9e1c49cefdd2410bbd161dfc73b", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, UPDATE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Subject: Talk
Session-Expires: 90
Supported: timer, 100rel, replaces
Content-Length: 241

v=0
o=amsip 0 0 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 18066 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=setup:passive

<------------->
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: --- (16 headers 12 lines) ---
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Using INVITE request as basis request - 1165902499
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found peer '1004' for '1004' from 192.168.1.5:5060
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found RTP audio format 3
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found RTP audio format 0
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found RTP audio format 8
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found RTP audio format 101
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found audio description format GSM for ID 3
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found audio description format PCMU for ID 0
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found audio description format PCMA for ID 8
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Found audio description format telephone-event for ID 101
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Peer audio RTP is at port 192.168.1.5:18066
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Peer doesn't provide video
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: Looking for 1003 in from-internal (domain 192.168.1.150)
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: list_route: hop: <sip:1004@192.168.1.5:5060>
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1964029822;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>
Call-ID: 1165902499
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 90;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Length: 0


<------------>
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [1003@from-internal:1] Macro("SIP/1004-00000014", "exten-vm,novm,1003") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/1004-00000014", "user-callerid,") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/1004-00000014", "AMPUSER=1004") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1004-00000014", "0?report") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1004-00000014", "1?Set(REALCALLERIDNUM=1004)") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/1004-00000014", "AMPUSER=1004") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/1004-00000014", "AMPUSERCIDNAME=test4") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1004-00000014", "0?report") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/1004-00000014", "AMPUSERCID=1004") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/1004-00000014", "CALLERID(all)="test4" <1004>") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1004-00000014", "0?Set(CHANNEL(language)=)") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1004-00000014", "0?continue") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/1004-00000014", "__TTL=64") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1004-00000014", "1?continue") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Goto (macro-user-callerid,s,19)
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/1004-00000014", "Using CallerID "test4" <1004>") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/1004-00000014", "RingGroupMethod=none") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/1004-00000014", "VMBOX=novm") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/1004-00000014", "EXTTOCALL=1003") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/1004-00000014", "CFUEXT=") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/1004-00000014", "CFBEXT=") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/1004-00000014", "RT=""") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/1004-00000014", "record-enable,1003,IN") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/1004-00000014", "1?check") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Goto (macro-record-enable,s,4)
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/1004-00000014", "0?MacroExit()") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/1004-00000014", "0?Group:OUT") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Goto (macro-record-enable,s,15)
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/1004-00000014", "1?IN") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Goto (macro-record-enable,s,20)
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/1004-00000014", "1?MacroExit()") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/1004-00000014", "dial,,tr,1003") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/1004-00000014", "1?dial") in new stack
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Goto (macro-dial,s,3)
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/1004-00000014", "dialparties.agi") in new stack
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[May 1 15:56:04] VERBOSE[9750] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[May 1 15:56:04] VERBOSE[9750] res_agi.c: dialparties.agi: Caller ID name is 'test4' number is '1004'
[May 1 15:56:04] VERBOSE[9750] res_agi.c: dialparties.agi: Methodology of ring is 'none'
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: Added extension 1003 to extension map
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: Extension 1003 cf is disabled
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: Extension 1003 do not disturb is disabled
[May 1 15:56:04] VERBOSE[9750] res_agi.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[May 1 15:56:04] VERBOSE[9750] res_agi.c: dialparties.agi: Extension 1003 has ExtensionState: 0
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: Checking CW and CFB status for extension 1003
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: dbset CALLTRACE/1003 to 1004
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- dialparties.agi: Filtered ARG3: 1003
[May 1 15:56:04] VERBOSE[9750] res_agi.c: -- <SIP/1004-00000014>AGI Script dialparties.agi completed, returning 0
[May 1 15:56:04] VERBOSE[9750] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/1004-00000014", "SIP/1003,,tr") in new stack
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using SIP RTP TOS bits 184
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using SIP RTP CoS mark 5
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using SIP VRTP TOS bits 136
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using SIP VRTP CoS mark 6
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using UDPTL TOS bits 184
[May 1 15:56:04] VERBOSE[9750] netsock.c: == Using UDPTL CoS mark 5
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Audio is at 192.168.1.150 port 17836
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Video is at 192.168.1.150 port 12422
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding video codec 0x40000 (h261) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 15:56:04] VERBOSE[9750] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
INVITE sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK071667fb;rport
Max-Forwards: 70
From: "test4" <sip:1004@192.168.1.150>;tag=as0e641658
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
Contact: <sip:1004@192.168.1.150>
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 444

v=0
o=root 1825576228 1825576228 IN IP4 192.168.1.150
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.150
b=CT:384
t=0 0
m=audio 17836 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12422 RTP/AVP 31 34 98 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
[May 1 15:56:04] VERBOSE[9750] app_dial.c: -- Called 1003
[May 1 15:56:04] VERBOSE[9750] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1964029822;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 90;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Length: 0


<------------>
[May 1 15:56:04] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK071667fb;rport=5060
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:56:04] VERBOSE[2898] chan_sip.c: --- (9 headers 0 lines) ---
[May 1 15:56:04] VERBOSE[9750] app_dial.c: -- SIP/1003-00000015 is ringing
[May 1 15:56:04] VERBOSE[9750] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1964029822;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 90;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Length: 0


<------------>
[May 1 15:56:05] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK170c3bba;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as7cb372c4
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 6b26130a7310171938d0d1066751a610@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:05] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
OPTIONS sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK57054701;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as18745b01
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 792ef13c504267d16c925ad6087300fd@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:05] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK57054701;rport=5060
Contact: <sip:192.168.1.3:60244>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=c9c0d419
From: "Unknown"<sip:Unknown@192.168.1.150>;tag=as18745b01
Call-ID: 792ef13c504267d16c925ad6087300fd@192.168.1.150
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:56:05] VERBOSE[2898] chan_sip.c: --- (13 headers 0 lines) ---
[May 1 15:56:05] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '792ef13c504267d16c925ad6087300fd@192.168.1.150' Method: OPTIONS
[May 1 15:56:06] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK170c3bba;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as7cb372c4
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 6b26130a7310171938d0d1066751a610@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:07] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK170c3bba;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as7cb372c4
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 6b26130a7310171938d0d1066751a610@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK170c3bba;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as7cb372c4
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 6b26130a7310171938d0d1066751a610@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:08] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK071667fb;rport=5060
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 649

v=0
o=- 12948713769652466 1 IN IP4 192.168.1.3
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.3
t=0 0
a=ice-ufrag:e00d01
a=ice-pwd:d24344356f8a34e81ce3d4c16010c065
m=audio 63598 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 63598 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 63599 typ host
m=video 61554 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=2
a=rtpmap:98 H263-1998/90000
a=fmtp:98 QCIF=1;CIF=1;VGA=2;I=1;J=1;T=1
a=inactive
a=candidate:1 1 UDP 659136 192.168.1.3 61554 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 61555 typ host

<------------->
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: --- (12 headers 21 lines) ---
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found RTP audio format 0
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found RTP audio format 8
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found RTP audio format 101
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found audio description format telephone-event for ID 101
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found RTP video format 34
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found RTP video format 98
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found video description format H263 for ID 34
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Found video description format H263-1998 for ID 98
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000c (ulaw|alaw|h263|h263p)
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Peer audio RTP is at port 192.168.1.3:63598
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Peer video RTP is at port 192.168.1.3:61554
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: list_route: hop: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67> for address/port to send to
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: set_destination: set destination to 192.168.1.3, port 60244
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: Transmitting (NAT) to 192.168.1.3:60244:
ACK sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK52d4705f;rport
Max-Forwards: 70
From: "test4" <sip:1004@192.168.1.150>;tag=as0e641658
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
Contact: <sip:1004@192.168.1.150>
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
[May 1 15:56:08] VERBOSE[9750] app_dial.c: -- SIP/1003-00000015 answered SIP/1004-00000014
[May 1 15:56:08] VERBOSE[9750] chan_sip.c: Audio is at 192.168.1.150 port 19304
[May 1 15:56:08] VERBOSE[9750] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[May 1 15:56:08] VERBOSE[9750] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 15:56:08] VERBOSE[9750] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 15:56:08] VERBOSE[9750] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 15:56:08] VERBOSE[9750] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1964029822;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 90;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1523946625 1523946625 IN IP4 192.168.1.150
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.150
t=0 0
m=audio 19304 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[May 1 15:56:08] VERBOSE[9750] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1004-00000014
[May 1 15:56:08] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1543346919
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 21 ACK
Contact: <sip:1004@192.168.1.5:5060>
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Content-Length: 0


<------------->
[May 1 15:56:08] VERBOSE[2898] chan_sip.c: --- (10 headers 0 lines) ---
[May 1 15:56:09] VERBOSE[9750] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1003-00000015
[May 1 15:56:09] NOTICE[9750] rtp.c: Unknown RTP codec 126 received from '192.168.1.3'
[May 1 15:56:09] NOTICE[9750] rtp.c: Unknown RTP codec 126 received from '192.168.1.3'
[May 1 15:56:09] NOTICE[9750] rtp.c: Unknown RTP codec 126 received from '192.168.1.3'
[May 1 15:56:09] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK170c3bba;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as7cb372c4
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 6b26130a7310171938d0d1066751a610@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:09] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '6b26130a7310171938d0d1066751a610@192.168.1.150' Method: OPTIONS
[May 1 15:56:10] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
seui
Silver Member
 
โพสต์: 20
ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
ข้างบน

Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 01 พ.ค. 2011 16:14

--------------------------------------------------------------------------------------------------------------
ตรงนี้สามารถบอกถึงอะไรได้บ้างครับ ลองหาข้อมูลแล้วแต่ไม่แน่ใจว่าเข้าใจถูกรึเปล่าครับ
v=0
o=amsip 0 0 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 0 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 0 RTP/AVP 116 115
b=AS:256
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000
-------------------------------------------------------------------------------------------

อันนี้เป็น log ที่ต่อจากด้านบนนะครับ
-------------------------------------------------------------------------------
<------------->
[May 1 15:56:14] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
INVITE sip:1004@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-41fe7ac57f0a5051-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
From: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 676

v=0
o=- 12948713769652466 2 IN IP4 192.168.1.3
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.3
t=0 0
a=ice-ufrag:e00d01
a=ice-pwd:d24344356f8a34e81ce3d4c16010c065
m=audio 63598 RTP/AVP 0 8 101 97
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:97 SPEEX/8000
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 63598 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 63599 typ host
m=video 61554 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=2
a=rtpmap:98 H263-1998/90000
a=fmtp:98 QCIF=1;CIF=1;VGA=2;I=1;J=1;T=1
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.3 61554 typ host
a=candidate:1 2 UDP 659134 192.168.1.3 61555 typ host

<------------->
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: --- (13 headers 22 lines) ---
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.3 : 60244 (NAT)
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP audio format 0
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP audio format 8
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP audio format 101
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP audio format 97
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found audio description format telephone-event for ID 101
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found audio description format SPEEX for ID 97
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP video format 34
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found RTP video format 98
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found video description format H263 for ID 34
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Found video description format H263-1998 for ID 98
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0x20c (ulaw|alaw|speex)/video=0x180000 (h263|h263p)/text=0x0 (nothing), combined - 0x18000c (ulaw|alaw|h263|h263p)
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Peer audio RTP is at port 192.168.1.3:63598
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Peer video RTP is at port 192.168.1.3:61554
[May 1 15:56:14] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:60244 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-41fe7ac57f0a5051-1---d8754z-;received=192.168.1.3;rport=60244
From: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
To: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:1004@192.168.1.150>
Content-Length: 0


<------------>
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Audio is at 192.168.1.150 port 17836
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Video is at 192.168.1.150 port 12422
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 15:56:14] VERBOSE[2898] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-41fe7ac57f0a5051-1---d8754z-;received=192.168.1.3;rport=60244
From: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
To: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:1004@192.168.1.150>
Content-Type: application/sdp
Content-Length: 367

v=0
o=root 1825576228 1825576229 IN IP4 192.168.1.150
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.150
b=CT:384
t=0 0
m=audio 17836 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12422 RTP/AVP 34 98
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=sendrecv

<------------>
[May 1 15:56:14] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
ACK sip:1004@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-6d9ab73a8cbf41cf-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
From: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 2 ACK
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:56:14] VERBOSE[2898] chan_sip.c: --- (10 headers 0 lines) ---
[May 1 15:56:18] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
INVITE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK870404142
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 22 INVITE
Contact: <sip:1004@192.168.1.5:5060>
Proxy-Authorization: Digest username="1004", realm="asterisk", nonce="745f1eb6", uri="sip:1003@192.168.1.150", response="8f18c9e1c49cefdd2410bbd161dfc73b", algorithm=MD5
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Supported: timer, 100rel, replaces
Content-Length: 340

v=0
o=amsip 0 1 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 18066 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=setup:passive
m=video 18192 RTP/AVP 116 115
b=AS:256
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000

<------------->
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: --- (13 headers 16 lines) ---
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP audio format 3
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP audio format 0
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP audio format 8
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP audio format 101
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found audio description format GSM for ID 3
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found audio description format PCMU for ID 0
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found audio description format PCMA for ID 8
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found audio description format telephone-event for ID 101
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP video format 116
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found RTP video format 115
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found video description format MP4V-ES for ID 116
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Found video description format H263-1998 for ID 115
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x500000 (h263p|mpeg4)/text=0x0 (nothing), combined - 0x10000e (gsm|ulaw|alaw|h263p)
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Peer audio RTP is at port 192.168.1.5:18066
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Peer video RTP is at port 192.168.1.5:18192
[May 1 15:56:18] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK870404142;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 22 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Length: 0


<------------>
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Audio is at 192.168.1.150 port 19304
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Video is at 192.168.1.150 port 19226
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 15:56:18] VERBOSE[2898] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK870404142;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 22 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1003@192.168.1.150>
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 1523946625 1523946626 IN IP4 192.168.1.150
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.150
b=CT:384
t=0 0
m=audio 19304 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 19226 RTP/AVP 115
a=rtpmap:115 h263-1998/90000
a=sendrecv

<------------>
[May 1 15:56:18] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK992805320
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 22 ACK
Contact: <sip:1004@192.168.1.5:5060>
Proxy-Authorization: Digest username="1004", realm="asterisk", nonce="745f1eb6", uri="sip:1003@192.168.1.150", response="8f18c9e1c49cefdd2410bbd161dfc73b", algorithm=MD5
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Content-Length: 0


<------------->
[May 1 15:56:18] VERBOSE[2898] chan_sip.c: --- (11 headers 0 lines) ---
[May 1 15:56:19] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
INFO sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK96242624
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 23 INFO
Contact: <sip:1004@192.168.1.5:5060>
Proxy-Authorization: Digest username="1004", realm="asterisk", nonce="745f1eb6", uri="sip:1003@192.168.1.150", response="ef1fefd3221f54b7e2254fa8f9be6d58", algorithm=MD5
Content-Type: application/media_control+xml
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Content-Length: 153

<?xml version="1.0" encoding="utf-8" ?> <media_control> <vc_primitive> <to_encoder> <picture_fast_update/> </to_encoder> </vc_primitive> </media_control>
<------------->
[May 1 15:56:19] VERBOSE[2898] chan_sip.c: --- (12 headers 1 lines) ---
[May 1 15:56:19] VERBOSE[2898] chan_sip.c: Receiving INFO!
[May 1 15:56:19] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK96242624;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 23 INFO
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[May 1 15:56:19] VERBOSE[9750] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67> for address/port to send to
[May 1 15:56:19] VERBOSE[9750] chan_sip.c: set_destination: set destination to 192.168.1.3, port 60244
[May 1 15:56:19] VERBOSE[9750] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
INFO sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK06392ebb;rport
Max-Forwards: 70
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
Contact: <sip:1004@192.168.1.150>
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 103 INFO
User-Agent: Asterisk PBX 1.6.2.13
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---
[May 1 15:56:19] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK06392ebb;rport=5060
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 103 INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:56:19] VERBOSE[2898] chan_sip.c: --- (9 headers 0 lines) ---
[May 1 15:56:19] VERBOSE[2898] chan_sip.c: SIP Response message for INCOMING dialog INFO arrived
[May 1 15:56:19] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK24cd7b4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as6cac85ea
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:20] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK24cd7b4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as6cac85ea
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:21] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK24cd7b4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as6cac85ea
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:22] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK24cd7b4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as6cac85ea
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:23] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK24cd7b4f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as6cac85ea
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:23] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '1dc6b12303ea351d4294c45f7fdc4303@192.168.1.150' Method: OPTIONS
[May 1 15:56:24] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
jaK
<------------->
[May 1 15:56:26] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.5:5060:
OPTIONS sip:1004@192.168.1.5:5060;line=2912a419c20457a SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK570348cb;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as58dccea6
To: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 415cf7fb5b947f5815aa3540750c261e@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:26] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK570348cb;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as58dccea6
To: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>;tag=528953192
Call-ID: 415cf7fb5b947f5815aa3540750c261e@192.168.1.150
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, UPDATE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Supported: timer, 100rel, replaces
Content-Length: 315

v=0
o=amsip 0 0 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 0 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 0 RTP/AVP 116 115
b=AS:256
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000

<------------->
[May 1 15:56:26] VERBOSE[2898] chan_sip.c: --- (12 headers 15 lines) ---
[May 1 15:56:26] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '415cf7fb5b947f5815aa3540750c261e@192.168.1.150' Method: OPTIONS
[May 1 15:56:33] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK71f6cdb3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as5e9ef61f
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 442c14847e9529c60b81dab8423f4085@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:34] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK71f6cdb3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as5e9ef61f
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 442c14847e9529c60b81dab8423f4085@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:35] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK71f6cdb3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as5e9ef61f
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 442c14847e9529c60b81dab8423f4085@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:36] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK71f6cdb3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as5e9ef61f
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 442c14847e9529c60b81dab8423f4085@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:37] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK71f6cdb3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as5e9ef61f
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 442c14847e9529c60b81dab8423f4085@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:37] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '442c14847e9529c60b81dab8423f4085@192.168.1.150' Method: OPTIONS
[May 1 15:56:40] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->



<------------->
[May 1 15:56:47] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
jaK
<------------->
[May 1 15:56:47] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK015cb77f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as1bc21ab1
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 235a07de29654f8b077f16ac3895da34@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:48] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK015cb77f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as1bc21ab1
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 235a07de29654f8b077f16ac3895da34@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:49] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK015cb77f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as1bc21ab1
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 235a07de29654f8b077f16ac3895da34@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:50] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK015cb77f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as1bc21ab1
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 235a07de29654f8b077f16ac3895da34@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:51] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK015cb77f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as1bc21ab1
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 235a07de29654f8b077f16ac3895da34@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:51] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '235a07de29654f8b077f16ac3895da34@192.168.1.150' Method: OPTIONS
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
REGISTER sip:192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1583391896
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>
Call-ID: 1806928584
CSeq: 53 REGISTER
Contact: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>
Authorization: Digest username="1004", realm="asterisk", nonce="055f6dae", uri="sip:192.168.1.150", response="19b95a574cdacbc09c936fa7f75ecd7d", algorithm=MD5
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Expires: 100
Supported: timer, 100rel, replaces
Content-Length: 0


<------------->
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: --- (13 headers 0 lines) ---
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1583391896;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>
Call-ID: 1806928584
CSeq: 53 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1583391896;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>;tag=as52d98b48
Call-ID: 1806928584
CSeq: 53 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a819f26"
Content-Length: 0


<------------>
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Scheduling destruction of SIP dialog '1806928584' in 32000 ms (Method: REGISTER)
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
REGISTER sip:192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK623465437
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>
Call-ID: 1806928584
CSeq: 54 REGISTER
Contact: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>
Authorization: Digest username="1004", realm="asterisk", nonce="6a819f26", uri="sip:192.168.1.150", response="4b752d82a6311ad73d9806a7c9d9fa28", algorithm=MD5
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Expires: 100
Supported: timer, 100rel, replaces
Content-Length: 0


<------------->
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: --- (13 headers 0 lines) ---
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK623465437;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>
Call-ID: 1806928584
CSeq: 54 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.5:5060:
OPTIONS sip:1004@192.168.1.5:5060;line=2912a419c20457a SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK67a6cb77;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as157f0c33
To: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 60bca6874d49e31e41236c562e51a917@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:56:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK623465437;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=12445667
To: <sip:1004@192.168.1.150>;tag=as52d98b48
Call-ID: 1806928584
CSeq: 54 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 100
Contact: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>;expires=100
Date: Sun, 01 May 2011 08:56:57 GMT
Content-Length: 0


<------------>
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Scheduling destruction of SIP dialog '1806928584' in 32000 ms (Method: REGISTER)
[May 1 15:56:57] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK67a6cb77;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as157f0c33
To: <sip:1004@192.168.1.5:5060;line=2912a419c20457a>;tag=514871784
Call-ID: 60bca6874d49e31e41236c562e51a917@192.168.1.150
CSeq: 102 OPTIONS
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, UPDATE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Supported: timer, 100rel, replaces
Content-Length: 315

v=0
o=amsip 0 0 IN IP4 192.168.1.5
s=talk
c=IN IP4 192.168.1.5
t=0 0
m=audio 0 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 0 RTP/AVP 116 115
b=AS:256
a=rtpmap:116 MP4V-ES/90000
a=rtpmap:115 H263-1998/90000

<------------->
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: --- (12 headers 15 lines) ---
[May 1 15:56:57] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '60bca6874d49e31e41236c562e51a917@192.168.1.150' Method: OPTIONS
[May 1 15:57:01] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7e6428ac;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b3f4a73
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 65de8cef309bafbf725e333c116190a1@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:02] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7e6428ac;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b3f4a73
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 65de8cef309bafbf725e333c116190a1@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:03] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7e6428ac;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b3f4a73
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 65de8cef309bafbf725e333c116190a1@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:04] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7e6428ac;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b3f4a73
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 65de8cef309bafbf725e333c116190a1@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:05] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7e6428ac;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b3f4a73
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 65de8cef309bafbf725e333c116190a1@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:05] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '65de8cef309bafbf725e333c116190a1@192.168.1.150' Method: OPTIONS
[May 1 15:57:05] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
OPTIONS sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK434e10c4;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as64497c3a
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 716fb76633a3edb9433b5c5e34eaf2e6@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:05] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK434e10c4;rport=5060
Contact: <sip:192.168.1.3:60244>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=e37edeb9
From: "Unknown"<sip:Unknown@192.168.1.150>;tag=as64497c3a
Call-ID: 716fb76633a3edb9433b5c5e34eaf2e6@192.168.1.150
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:57:05] VERBOSE[2898] chan_sip.c: --- (13 headers 0 lines) ---
[May 1 15:57:05] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '716fb76633a3edb9433b5c5e34eaf2e6@192.168.1.150' Method: OPTIONS
[May 1 15:57:10] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->



<------------->
[May 1 15:57:10] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
BYE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bK1071850157
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 24 BYE
Contact: <sip:1004@192.168.1.5:5060>
Proxy-Authorization: Digest username="1004", realm="asterisk", nonce="745f1eb6", uri="sip:1003@192.168.1.150", response="2d81ab52ccf6601a26791d0b7dd41019", algorithm=MD5
Max-Forwards: 70
User-Agent: antisip/4.6.0-Apr-27-2011 amdroid/1.2.15 WellcoM-A99/2.2.2
Content-Length: 0


<------------->
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: --- (11 headers 0 lines) ---
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: Sending to 192.168.1.5 : 5060 (NAT)
[May 1 15:57:10] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK1071850157;received=192.168.1.5;rport=5060
From: <sip:1004@192.168.1.150>;tag=1645903406
To: <sip:1003@192.168.1.150>;tag=as72e2dc6a
Call-ID: 1165902499
CSeq: 24 BYE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[May 1 15:57:10] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.5:5060 --->
jaK
<------------->
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [h@macro-dial:1] Macro("SIP/1004-00000014", "hangupcall") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1004-00000014", "1?noautomon") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Goto (macro-hangupcall,s,3)
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/1004-00000014", "TOUCH_MONITOR_OUTPUT=") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1004-00000014", "1?skiprg") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Goto (macro-hangupcall,s,7)
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1004-00000014", "1?skipblkvm") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Goto (macro-hangupcall,s,10)
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1004-00000014", "1?theend") in new stack
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Goto (macro-hangupcall,s,12)
[May 1 15:57:10] VERBOSE[9750] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/1004-00000014", "") in new stack
[May 1 15:57:10] VERBOSE[9750] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1004-00000014' in macro 'hangupcall'
[May 1 15:57:10] VERBOSE[9750] chan_sip.c: Scheduling destruction of SIP dialog '0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150' in 6400 ms (Method: ACK)
[May 1 15:57:10] VERBOSE[9750] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67> for address/port to send to
[May 1 15:57:10] VERBOSE[9750] chan_sip.c: set_destination: set destination to 192.168.1.3, port 60244
[May 1 15:57:10] VERBOSE[9750] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
BYE sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK38a171e8;rport
Max-Forwards: 70
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[May 1 15:57:10] VERBOSE[9750] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1003-00000015
[May 1 15:57:10] VERBOSE[9750] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1004-00000014' in macro 'dial'
[May 1 15:57:10] VERBOSE[9750] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/1004-00000014' in macro 'exten-vm'
[May 1 15:57:10] VERBOSE[9750] pbx.c: == Spawn extension (from-internal, 1003, 1) exited non-zero on 'SIP/1004-00000014'
[May 1 15:57:10] VERBOSE[9750] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1004-00000014
[May 1 15:57:10] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK38a171e8;rport=5060
Contact: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>
To: <sip:1003@192.168.1.3:60244;rinstance=a428abf43b509b67>;tag=5d2e483e
From: "test4"<sip:1004@192.168.1.150>;tag=as0e641658
Call-ID: 0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150
CSeq: 104 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: --- (9 headers 0 lines) ---
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '1165902499' Method: BYE
[May 1 15:57:10] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '0fa11e8b31bc380137e21fd466ad9eae@192.168.1.150' Method: ACK
[May 1 15:57:12] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SUBSCRIBE sip:Unknown@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-ab45f4c104daaefb-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.3:60244>
To: <sip:1003@192.168.1.150>;tag=as668aca3a
From: <sip:1003@192.168.1.150>;tag=e65f8a82
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 61 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1003",realm="asterisk",nonce="77f650ed",uri="sip:Unknown@192.168.1.150",response="b933351619b18218b6379b4f19e6a46b",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: --- (14 headers 0 lines) ---
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.3:60244
[May 1 15:57:12] NOTICE[2898] chan_sip.c: Correct auth, but based on stale nonce received from '<sip:1003@192.168.1.150>;tag=e65f8a82'
[May 1 15:57:12] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:60244 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-ab45f4c104daaefb-1---d8754z-;received=192.168.1.3;rport=60244
From: <sip:1003@192.168.1.150>;tag=e65f8a82
To: <sip:1003@192.168.1.150>;tag=as668aca3a
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 61 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45d2b0ce", stale=true
Content-Length: 0


<------------>
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: Scheduling destruction of SIP dialog 'MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.' in 6400 ms (Method: SUBSCRIBE)
[May 1 15:57:12] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SUBSCRIBE sip:Unknown@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-60d782e0a7582aa6-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.3:60244>
To: <sip:1003@192.168.1.150>;tag=as668aca3a
From: <sip:1003@192.168.1.150>;tag=e65f8a82
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 62 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1003",realm="asterisk",nonce="45d2b0ce",uri="sip:Unknown@192.168.1.150",response="323decda4e95e0b353f8e2ece319ecb5",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: --- (14 headers 0 lines) ---
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.3:60244
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: Scheduling destruction of SIP dialog 'MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.' in 310000 ms (Method: SUBSCRIBE)
[May 1 15:57:12] VERBOSE[2898] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:60244;branch=z9hG4bK-d8754z-60d782e0a7582aa6-1---d8754z-;received=192.168.1.3;rport=60244
From: <sip:1003@192.168.1.150>;tag=e65f8a82
To: <sip:1003@192.168.1.150>;tag=as668aca3a
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 62 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:Unknown@192.168.1.150>;expires=300
Content-Length: 0


<------------>
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:60244:
NOTIFY sip:1003@192.168.1.3:60244 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3658b4ec;rport
Max-Forwards: 70
Route: <sip:1003@192.168.1.3:60244>
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as668aca3a
To: <sip:1003@192.168.1.3:60244>;tag=e65f8a82
Contact: <sip:Unknown@192.168.1.150>
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 132 NOTIFY
User-Agent: Asterisk PBX 1.6.2.13
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.150
Voice-Message: 0/0 (0/0)

---
[May 1 15:57:12] VERBOSE[2898] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:60244 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3658b4ec;rport=5060
Contact: <sip:1003@192.168.1.3:60244>
To: <sip:1003@192.168.1.3:60244>;tag=e65f8a82
From: "Unknown"<sip:Unknown@192.168.1.150>;tag=as668aca3a
Call-ID: MWRkNTM0NDgzNDBmYjBhMzEyY2E1YzJlZjNlZDk2NDA.
CSeq: 132 NOTIFY
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[May 1 15:57:12] VERBOSE[2898] chan_sip.c: --- (9 headers 0 lines) ---
[May 1 15:57:15] VERBOSE[2898] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK35dfdfa8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b664590
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:16] VERBOSE[2898] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK35dfdfa8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b664590
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:17] VERBOSE[2898] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK35dfdfa8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b664590
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:18] VERBOSE[2898] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK35dfdfa8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b664590
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:19] VERBOSE[2898] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.4:56698:
OPTIONS sip:1002@192.168.1.4:56698;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK35dfdfa8;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3b664590
To: <sip:1002@192.168.1.4:56698;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Sun, 01 May 2011 08:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[May 1 15:57:19] VERBOSE[2898] chan_sip.c: Really destroying SIP dialog '3b3b7dc05e0c60610fff94525ee64c4d@192.168.1.150' Method: OPTIONS
[May 1 15:57:21] VERBOSE[9752] manager.c: == Manager 'admin' logged on from 127.0.0.1
seui
Silver Member
 
โพสต์: 20
ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
ข้างบน

Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย nuiz » 01 พ.ค. 2011 16:41

ใน FAQ ของ SIPdroid มีบอกไว้ครับว่า video จะเวอร์ค (send ออกจาก SIPdroid) เมื่อเชื่อมต่อกับ PBXes แสดงว่าถ้าปลอมตัวเป็น PBXes ได้ ก็คงจะเวอร์คหน่ะครับ แต่ปลอมยังไงไม่รู้
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
Diamond Member
 
โพสต์: 7069
ลงทะเบียนเมื่อ: 24 มี.ค. 2010 09:33
ข้างบน
ย้อนกลับ
ตอบกระทู้

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กำลังดูบอร์ดนี้: ไม่มีสมาชิกใหม่ และ บุคคลทั่วไป 2 ท่าน


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