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ถามปัญหาการส่ง videocall ครับ

Asterisk Opensource IP Pbx
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 28 มี.ค. 2011 12:45

nuiz เขียน:เซ็ตด้วย trixbox หรือ freepbx หรือ elastix แต่ละเบอร์ extension ให้ใส่ตรงช่อง disallow ด้วยนะครับ
disallow = all

แล้วช่อง allow ให้เพิ่ม codec ของ voice ด้วย เช่น gsm, ulaw, alaw

สรุปว่าแต่ละ extension ให้ทำแบบนี้

disallow=all
allow=gsm,ulaw,alaw,h264,h263p,h263,h261

แล้วดูบทความนี้ครับ เผื่อจะช่วยได้ เพราะใน TrixBox ก็มี FreePBX เหมือนกันกับ Elastix

การเซ็ต Video บน Elastix 2.x


ขอบคุณมากครับ :D
seui
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 18 เม.ย. 2011 12:45

nuiz เขียน:เซ็ตด้วย trixbox หรือ freepbx หรือ elastix แต่ละเบอร์ extension ให้ใส่ตรงช่อง disallow ด้วยนะครับ
disallow = all

แล้วช่อง allow ให้เพิ่ม codec ของ voice ด้วย เช่น gsm, ulaw, alaw

สรุปว่าแต่ละ extension ให้ทำแบบนี้

disallow=all
allow=gsm,ulaw,alaw,h264,h263p,h263,h261

แล้วดูบทความนี้ครับ เผื่อจะช่วยได้ เพราะใน TrixBox ก็มี FreePBX เหมือนกันกับ Elastix

การเซ็ต Video บน Elastix 2.x


ลองทำ x-lite กับ x-lite โดยผ่าน server ตัวเองได้ครับ แต่พอ x-lite กับ โทรศัพท์ที่ลง sipdroid ไว้ x-lite รับภาพได้ แต่ sipdroid รับภาพที่ x-lite ส่งมาไม่ได้ครับ ถ้าพอทราบรบกวนแนะนำด้วยนะครับ :?:
seui
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ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย nuiz » 18 เม.ย. 2011 17:20

asterisk รองรับ video codec h261, h263, h263p, h264
แล้ว sipdroid รองรับ video codec อะไรบ้าง?

ใน Asterisk Console ผมว่ามันต้องมี Messages เกี่ยวกับ Error ของ Video ตอนที่คุยกับ sipdroid ลองสังเกตุดูครับ

ได้แต่แนะนำ เพราะไม่เคยใช้อ่ะครับ
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
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ลงทะเบียนเมื่อ: 24 มี.ค. 2010 09:33
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 20 เม.ย. 2011 11:34

nuiz เขียน:asterisk รองรับ video codec h261, h263, h263p, h264
แล้ว sipdroid รองรับ video codec อะไรบ้าง?

ใน Asterisk Console ผมว่ามันต้องมี Messages เกี่ยวกับ Error ของ Video ตอนที่คุยกับ sipdroid ลองสังเกตุดูครับ

ได้แต่แนะนำ เพราะไม่เคยใช้อ่ะครับ


ถ้าเป็น h263-1998 อันนี้เป็นอันเดียวกับ h263 หรือเปล่าครับ asterisk สามารถเพิ่ม codec เข้าไปได้ไหมครับ :)

[Apr 20 11:36:49] NOTICE[3291] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 20 11:36:49] NOTICE[3291] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 20 11:36:49] NOTICE[3291] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'

messages นี้น่าจะเกี่ยวข้องไหมครับผมไม่แน่ใจครับ :)
seui
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โพสต์: 20
ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย nuiz » 22 เม.ย. 2011 08:26

คงมีส่วนหน่ะครับ เกิดจากทางฝั่ง Client และ Server ตกลงกันแล้วว่าจะเลือกใช้ Codec นี้ แต่ปรากฏว่าไม่ Compatible กัน ไอพี 192.168.1.4 เป็นไอพีของอะไรครับ X-Lite หรือว่า SIPdroid

ผมลองค้นดูใน google ก็เจอโพสต์ที่สอบถามว่าจะแก้ปัญหา video ระหว่าง sipdroid กับ asterisk ได้ยังไง

FAQ ในเว็บ http://code.google.com/p/sipdroid/wiki/FAQ มีพูดถึง Video ด้วยครับ ส่งได้ รับได้ แต่รับได้เฉพาะตอนที่รีจิสเตอร์กับ PBXes เท่านั้น
Sending
By pressing the MENU button and choosing "Send Video" you can start video transmission to a SIP phone with video.
Receiving
This is not supported natively. If you are registered to PBXes and the other party starts sending video it will show up on the Android phone.

คงกั๊กไว้หน่ะครับ

ลองให้ x-lite คุยตรงกับ sipdroid ดูครับ โดยเซ็ต canreinvite=yes บน extensions ของทั้ง sipdroid และ x-lite คือไม่ให้ rtp ผ่าน asterisk ดูว่าจะเป็นยังไง
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย nuiz » 22 เม.ย. 2011 08:41

รบกวนลองแบบนี้ให้ผมหน่อยครับ เซ็ตแบบปกติที่เคยเซ็ตหน่ะครับ ผมอยากเห็น SIP Messages ที่ SIPdroid คุยกับ Asterisk

1. เข้า asterisk console
2. พิมพ์ sip set debug on
ถ้าพิมพ์คำสั่งนี้แล้ว error ก็ลองพิมพ์ help sip ดูครับ เผื่อคำสั่งจะเปลี่ยน (ผมใช้ Asterisk 1.4 อยู่)
3. พิมพ์ sip set debug
4. ลองโทรจาก sipdroid ไปหาปลายทาง
จะเห็นข้อความแปลกๆ ซึ่งเป็น SIP message
5. ก๊อบทั้งหมดแล้วโพสต์มาครับ ตั้งแต่เริ่มโทร ปลายทางรับสาย คุยกันสักสองสามวินาที แล้วกดวางสาย
6. พิมพ์ sip set debug off
เพื่อปิด debug

แล้วทำอีกรอบนึงครับ คราวนี้ให้ x-lite เป็นคนโทร แล้วโพสต์ แยกกันกับชุดแรกนะครับ จะได้ดูง่ายๆหน่อย

ขอบคุณครับ
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
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โพสต์: 6995
ลงทะเบียนเมื่อ: 24 มี.ค. 2010 09:33
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Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 22 เม.ย. 2011 11:59

nuiz เขียน:รบกวนลองแบบนี้ให้ผมหน่อยครับ เซ็ตแบบปกติที่เคยเซ็ตหน่ะครับ ผมอยากเห็น SIP Messages ที่ SIPdroid คุยกับ Asterisk

1. เข้า asterisk console
2. พิมพ์ sip set debug on
ถ้าพิมพ์คำสั่งนี้แล้ว error ก็ลองพิมพ์ help sip ดูครับ เผื่อคำสั่งจะเปลี่ยน (ผมใช้ Asterisk 1.4 อยู่)
3. พิมพ์ sip set debug
4. ลองโทรจาก sipdroid ไปหาปลายทาง
จะเห็นข้อความแปลกๆ ซึ่งเป็น SIP message
5. ก๊อบทั้งหมดแล้วโพสต์มาครับ ตั้งแต่เริ่มโทร ปลายทางรับสาย คุยกันสักสองสามวินาที แล้วกดวางสาย
6. พิมพ์ sip set debug off
เพื่อปิด debug

แล้วทำอีกรอบนึงครับ คราวนี้ให้ x-lite เป็นคนโทร แล้วโพสต์ แยกกันกับชุดแรกนะครับ จะได้ดูง่ายๆหน่อย

ขอบคุณครับ


Messages Sipdroid :D

-------------------------------------------------------------------------------

CSeq: 1 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="0e3f3e81"
Content-Length: 0


<------------>
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog '598278927302@192.168.1.3' in 11648 ms (Method: INVITE)
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
REGISTER sip:192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:41967;rport;branch=z9hG4bK12642
Max-Forwards: 70
To: <sip:1002@192.168.1.150>
From: <sip:1002@192.168.1.150>;tag=z9hG4bK33294780
Call-ID: 424690801129@192.168.1.3
CSeq: 2 REGISTER
Contact: <sip:1002@192.168.1.3:41967;transport=udp>
Expires: 3600
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Authorization: Digest username="1002", realm="asterisk", nonce="75fc0a18", uri="sip:192.168.1.150", algorithm=MD5, response="6d18a3df0607a45ba8a82eb81fbbc4c4"
Content-Length: 0


<------------->
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: --- (12 headers 0 lines) ---
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.3 : 41967 (NAT)
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK12642;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK33294780
To: <sip:1002@192.168.1.150>
Call-ID: 424690801129@192.168.1.3
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:41967:
OPTIONS sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3530b858;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3bb05c02
To: <sip:1002@192.168.1.3:41967;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3bbd7a6d0a22777f568d3fb8681c223d@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 22 Apr 2011 04:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK12642;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK33294780
To: <sip:1002@192.168.1.150>;tag=as2e2d3a47
Call-ID: 424690801129@192.168.1.3
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 3600
Contact: <sip:1002@192.168.1.3:41967;transport=udp>;expires=3600
Date: Fri, 22 Apr 2011 04:38:32 GMT
Content-Length: 0


<------------>
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog '424690801129@192.168.1.3' in 32000 ms (Method: REGISTER)
[Apr 22 11:38:32] VERBOSE[2907] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.3:41967:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK53126;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
To: <sip:1003@192.168.1.150>;tag=as5b7fbaf9
Call-ID: 598278927302@192.168.1.3
CSeq: 1 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="0e3f3e81"
Content-Length: 0


---
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:41967;rport;branch=z9hG4bK53126
Max-Forwards: 70
To: <sip:1003@192.168.1.150>;tag=as5b7fbaf9
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 1 ACK
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
INVITE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:41967;rport;branch=z9hG4bK63651
Max-Forwards: 70
To: <sip:1003@192.168.1.150>
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 2 INVITE
Contact: <sip:1002@192.168.1.3:41967;transport=udp>
Expires: 3600
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Authorization: Digest username="1002", realm="asterisk", nonce="0e3f3e81", uri="sip:1003@192.168.1.150", algorithm=MD5, response="78d2463713fad56b2534ef862d181765"
Content-Length: 282
Content-Type: application/sdp

v=0
o=1002@192.168.1.150 0 0 IN IP4 192.168.1.3
s=Session SIP/SDP
c=IN IP4 192.168.1.3
t=0 0
m=audio 21000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 21070 RTP/AVP 103
a=rtpmap:103 h263-1998/90000

<------------->
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: --- (13 headers 12 lines) ---
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.3 : 41967 (NAT)
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Using INVITE request as basis request - 598278927302@192.168.1.3
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found peer '1002' for '1002' from 192.168.1.3:41967
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found RTP audio format 8
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found audio description format PCMU for ID 0
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found RTP video format 103
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Found video description format h263-1998 for ID 103
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x10000c (ulaw|alaw|h263p)
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.3:21000
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Peer video RTP is at port 192.168.1.3:21070
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Looking for 1003 in from-internal (domain 192.168.1.150)
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: list_route: hop: <sip:1002@192.168.1.3:41967;transport=udp>
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK63651;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
To: <sip:1003@192.168.1.150>
Call-ID: 598278927302@192.168.1.3
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:1003@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [1003@from-internal:1] Macro("SIP/1002-0000000c", "exten-vm,novm,1003") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/1002-0000000c", "user-callerid,") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/1002-0000000c", "AMPUSER=1002") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-0000000c", "0?report") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-0000000c", "1?Set(REALCALLERIDNUM=1002)") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/1002-0000000c", "AMPUSER=1002") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/1002-0000000c", "AMPUSERCIDNAME=test2") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-0000000c", "0?report") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/1002-0000000c", "AMPUSERCID=1002") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/1002-0000000c", "CALLERID(all)="test2" <1002>") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-0000000c", "0?Set(CHANNEL(language)=)") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-0000000c", "0?continue") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/1002-0000000c", "__TTL=64") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1002-0000000c", "1?continue") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Goto (macro-user-callerid,s,19)
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/1002-0000000c", "Using CallerID "test2" <1002>") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/1002-0000000c", "RingGroupMethod=none") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/1002-0000000c", "VMBOX=novm") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/1002-0000000c", "EXTTOCALL=1003") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/1002-0000000c", "CFUEXT=") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/1002-0000000c", "CFBEXT=") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/1002-0000000c", "RT=""") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/1002-0000000c", "record-enable,1003,IN") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/1002-0000000c", "1?check") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Goto (macro-record-enable,s,4)
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/1002-0000000c", "0?MacroExit()") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/1002-0000000c", "0?Group:OUT") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Goto (macro-record-enable,s,15)
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/1002-0000000c", "1?IN") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Goto (macro-record-enable,s,20)
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/1002-0000000c", "1?MacroExit()") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/1002-0000000c", "dial,,tr,1003") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/1002-0000000c", "1?dial") in new stack
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Goto (macro-dial,s,3)
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/1002-0000000c", "dialparties.agi") in new stack
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: dialparties.agi: Caller ID name is 'test2' number is '1002'
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: dialparties.agi: Methodology of ring is 'none'
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: Added extension 1003 to extension map
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: Extension 1003 cf is disabled
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: Extension 1003 do not disturb is disabled
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: dialparties.agi: Extension 1003 has ExtensionState: 0
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: Checking CW and CFB status for extension 1003
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: dbset CALLTRACE/1003 to 1002
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- dialparties.agi: Filtered ARG3: 1003
[Apr 22 11:38:33] VERBOSE[3418] res_agi.c: -- <SIP/1002-0000000c>AGI Script dialparties.agi completed, returning 0
[Apr 22 11:38:33] VERBOSE[3418] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/1002-0000000c", "SIP/1003,,tr") in new stack
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using SIP RTP TOS bits 184
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using SIP RTP CoS mark 5
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using SIP VRTP TOS bits 136
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using SIP VRTP CoS mark 6
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using UDPTL TOS bits 184
[Apr 22 11:38:33] VERBOSE[3418] netsock.c: == Using UDPTL CoS mark 5
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Audio is at 192.168.1.150 port 11164
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Video is at 192.168.1.150 port 12768
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:53412:
INVITE sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK260bfe8b;rport
Max-Forwards: 70
From: "test2" <sip:1002@192.168.1.150>;tag=as54f98291
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
Contact: <sip:1002@192.168.1.150>
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 22 Apr 2011 04:38:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 1002878268 1002878268 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 11164 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 12768 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
a=sendrecv

---
[Apr 22 11:38:33] VERBOSE[3418] app_dial.c: -- Called 1003
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK63651;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
To: <sip:1003@192.168.1.150>;tag=as64eeb04a
Call-ID: 598278927302@192.168.1.3
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:1003@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK260bfe8b;rport=5060
Contact: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=a0979a4f
From: "test2"<sip:1002@192.168.1.150>;tag=as54f98291
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:38:33] VERBOSE[3418] app_dial.c: -- SIP/1003-0000000d is ringing
[Apr 22 11:38:33] VERBOSE[3418] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK63651;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
To: <sip:1003@192.168.1.150>;tag=as64eeb04a
Call-ID: 598278927302@192.168.1.3
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:1003@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK3530b858;rport=5060
To: <sip:1002@192.168.1.3:41967;transport=udp>
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as3bb05c02
Call-ID: 3bbd7a6d0a22777f568d3fb8681c223d@192.168.1.150
CSeq: 102 OPTIONS
Contact: <sip:1002@192.168.1.3:41967;transport=udp>
Content-Length: 0


<------------->
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '3bbd7a6d0a22777f568d3fb8681c223d@192.168.1.150' Method: OPTIONS
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:41967;rport;branch=z9hG4bK53126
Max-Forwards: 70
To: <sip:1003@192.168.1.150>;tag=as5b7fbaf9
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 1 ACK
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:38:33] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK260bfe8b;rport=5060
Contact: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=a0979a4f
From: "test2"<sip:1002@192.168.1.150>;tag=as54f98291
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@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: 595

v=0
o=- 12947920706125108 1 IN IP4 192.168.1.4
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.4
t=0 0
a=ice-ufrag:14588b
a=ice-pwd:4caf05d472bca8b2b6fc9836b6efb79a
m=audio 64562 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 64562 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 64563 typ host
m=video 62010 RTP/AVP 103
a=rtpmap:103 H263-1998/90000
a=fmtp:103 QCIF=1;CIF=1;VGA=2;I=1;J=1;T=1
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 62010 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 62011 typ host

<------------->
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: --- (12 headers 19 lines) ---
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found RTP audio format 8
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found RTP video format 103
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Found video description format H263-1998 for ID 103
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x10000c (ulaw|alaw|h263p)
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.4:64562
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Peer video RTP is at port 192.168.1.4:62010
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: list_route: hop: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279> for address/port to send to
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: set_destination: set destination to 192.168.1.4, port 53412
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Transmitting (NAT) to 192.168.1.4:53412:
ACK sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK30d8c3a6;rport
Max-Forwards: 70
From: "test2" <sip:1002@192.168.1.150>;tag=as54f98291
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=a0979a4f
Contact: <sip:1002@192.168.1.150>
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
[Apr 22 11:38:35] VERBOSE[3418] app_dial.c: -- SIP/1003-0000000d answered SIP/1002-0000000c
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Audio is at 192.168.1.150 port 16378
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Video is at 192.168.1.150 port 13988
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:38:35] VERBOSE[3418] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.3:41967 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:41967;branch=z9hG4bK63651;received=192.168.1.3;rport=41967
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
To: <sip:1003@192.168.1.150>;tag=as64eeb04a
Call-ID: 598278927302@192.168.1.3
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 1678456766 1678456766 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 16378 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 13988 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
a=sendrecv

<------------>
[Apr 22 11:38:35] VERBOSE[3418] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1002-0000000c
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->



<------------->
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:41967 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:41967;rport;branch=z9hG4bK64869
Max-Forwards: 70
To: <sip:1003@192.168.1.150>;tag=as64eeb04a
From: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 2 ACK
Contact: <sip:1002@192.168.1.3:41967;transport=udp>
Expires: 3600
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:53412:
OPTIONS sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK178b5233;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as2fb7b411
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 3b31ccd64055a6e23dbe8a6c20733e91@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 22 Apr 2011 04:38:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK178b5233;rport=5060
Contact: <sip:192.168.1.4:53412>
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=8d491b55
From: "Unknown"<sip:Unknown@192.168.1.150>;tag=as2fb7b411
Call-ID: 3b31ccd64055a6e23dbe8a6c20733e91@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


<------------->
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: --- (13 headers 0 lines) ---
[Apr 22 11:38:35] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '3b31ccd64055a6e23dbe8a6c20733e91@192.168.1.150' Method: OPTIONS
[Apr 22 11:38:35] NOTICE[3418] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:38:35] NOTICE[3418] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:38:35] NOTICE[3418] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:38:35] VERBOSE[3418] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1003-0000000d
[Apr 22 11:38:52] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
BYE sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-f32f2ee2ea9b2080-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
To: "test2"<sip:1002@192.168.1.150>;tag=as54f98291
From: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=a0979a4f
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150
CSeq: 2 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[Apr 22 11:38:52] VERBOSE[2907] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 22 11:38:52] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.4 : 53412 (NAT)
[Apr 22 11:38:52] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-f32f2ee2ea9b2080-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=a0979a4f
To: "test2"<sip:1002@192.168.1.150>;tag=as54f98291
Call-ID: 3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [h@macro-dial:1] Macro("SIP/1002-0000000c", "hangupcall") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1002-0000000c", "1?noautomon") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,3)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/1002-0000000c", "TOUCH_MONITOR_OUTPUT=") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1002-0000000c", "1?skiprg") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,7)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1002-0000000c", "1?skipblkvm") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,10)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1002-0000000c", "1?theend") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,12)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/1002-0000000c", "") in new stack
[Apr 22 11:38:52] VERBOSE[3418] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1002-0000000c' in macro 'hangupcall'
[Apr 22 11:38:52] VERBOSE[3418] features.c: == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/1002-0000000c'
[Apr 22 11:38:52] VERBOSE[3418] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1003-0000000d
[Apr 22 11:38:52] VERBOSE[3418] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1002-0000000c' in macro 'dial'
[Apr 22 11:38:52] VERBOSE[3418] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/1002-0000000c' in macro 'exten-vm'
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: == Spawn extension (from-internal, 1003, 1) exited non-zero on 'SIP/1002-0000000c'
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/1002-0000000c", "hangupcall") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1002-0000000c", "1?noautomon") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,3)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/1002-0000000c", "TOUCH_MONITOR_OUTPUT=") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1002-0000000c", "1?skiprg") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,7)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1002-0000000c", "1?skipblkvm") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,10)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1002-0000000c", "1?theend") in new stack
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Goto (macro-hangupcall,s,12)
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/1002-0000000c", "") in new stack
[Apr 22 11:38:52] VERBOSE[3418] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1002-0000000c' in macro 'hangupcall'
[Apr 22 11:38:52] VERBOSE[3418] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1002-0000000c'
[Apr 22 11:38:52] VERBOSE[3418] chan_sip.c: Scheduling destruction of SIP dialog '598278927302@192.168.1.3' in 11648 ms (Method: ACK)
[Apr 22 11:38:52] VERBOSE[3418] chan_sip.c: set_destination: Parsing <sip:1002@192.168.1.3:41967;transport=udp> for address/port to send to
[Apr 22 11:38:52] VERBOSE[3418] chan_sip.c: set_destination: set destination to 192.168.1.3, port 41967
[Apr 22 11:38:52] VERBOSE[3418] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:41967:
BYE sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK198ab028;rport
Max-Forwards: 70
From: <sip:1003@192.168.1.150>;tag=as64eeb04a
To: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:38:52] VERBOSE[3418] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1002-0000000c
[Apr 22 11:38:53] VERBOSE[2907] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.3:41967:
BYE sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK198ab028;rport
Max-Forwards: 70
From: <sip:1003@192.168.1.150>;tag=as64eeb04a
To: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:38:53] VERBOSE[2907] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.3:41967:
BYE sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK198ab028;rport
Max-Forwards: 70
From: <sip:1003@192.168.1.150>;tag=as64eeb04a
To: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:38:53] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '3aa851e4468336bb2755d8552ea4f3b7@192.168.1.150' Method: BYE
[Apr 22 11:38:54] VERBOSE[2907] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.3:41967:
BYE sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK198ab028;rport
Max-Forwards: 70
From: <sip:1003@192.168.1.150>;tag=as64eeb04a
To: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:59708 --->
REGISTER sip:192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:59708;rport;branch=z9hG4bK59139
Max-Forwards: 70
To: <sip:1002@192.168.1.150>
From: <sip:1002@192.168.1.150>;tag=z9hG4bK42336985
Call-ID: 149713939278@192.168.1.3
CSeq: 1 REGISTER
Contact: <sip:1002@192.168.1.3:59708;transport=udp>
Expires: 3600
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.3 : 59708 (NAT)
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:59708 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.3:59708;branch=z9hG4bK59139;received=192.168.1.3;rport=59708
From: <sip:1002@192.168.1.150>;tag=z9hG4bK42336985
To: <sip:1002@192.168.1.150>
Call-ID: 149713939278@192.168.1.3
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:59708 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:59708;branch=z9hG4bK59139;received=192.168.1.3;rport=59708
From: <sip:1002@192.168.1.150>;tag=z9hG4bK42336985
To: <sip:1002@192.168.1.150>;tag=as11f43f7d
Call-ID: 149713939278@192.168.1.3
CSeq: 1 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="48cad3a1"
Content-Length: 0


<------------>
[Apr 22 11:38:55] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog '149713939278@192.168.1.3' in 32000 ms (Method: REGISTER)
[Apr 22 11:38:56] VERBOSE[2907] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.3:41967:
BYE sip:1002@192.168.1.3:41967;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK198ab028;rport
Max-Forwards: 70
From: <sip:1003@192.168.1.150>;tag=as64eeb04a
To: <sip:1002@192.168.1.150>;tag=z9hG4bK90936038
Call-ID: 598278927302@192.168.1.3
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:38:57] VERBOSE[3420] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:38:57] VERBOSE[3420] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:39:02] VERBOSE[2907] chan_sip.c: -- Registered SIP '1002' at 192.168.1.3 port 33597
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using SIP RTP TOS bits 184
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using SIP RTP CoS mark 5
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using SIP VRTP TOS bits 136
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using SIP VRTP CoS mark 6
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using UDPTL TOS bits 184
[Apr 22 11:39:06] VERBOSE[2907] netsock.c: == Using UDPTL CoS mark 5
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [1003@from-internal:1] Macro("SIP/1002-0000000e", "exten-vm,novm,1003") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/1002-0000000e", "user-callerid,") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/1002-0000000e", "AMPUSER=1002") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-0000000e", "0?report") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-0000000e", "1?Set(REALCALLERIDNUM=1002)") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/1002-0000000e", "AMPUSER=1002") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/1002-0000000e", "AMPUSERCIDNAME=test2") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-0000000e", "0?report") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/1002-0000000e", "AMPUSERCID=1002") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/1002-0000000e", "CALLERID(all)="test2" <1002>") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-0000000e", "0?Set(CHANNEL(language)=)") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-0000000e", "0?continue") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/1002-0000000e", "__TTL=64") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1002-0000000e", "1?continue") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Goto (macro-user-callerid,s,19)
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/1002-0000000e", "Using CallerID "test2" <1002>") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/1002-0000000e", "RingGroupMethod=none") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/1002-0000000e", "VMBOX=novm") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/1002-0000000e", "EXTTOCALL=1003") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/1002-0000000e", "CFUEXT=") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/1002-0000000e", "CFBEXT=") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/1002-0000000e", "RT=""") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/1002-0000000e", "record-enable,1003,IN") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/1002-0000000e", "1?check") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Goto (macro-record-enable,s,4)
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/1002-0000000e", "0?MacroExit()") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/1002-0000000e", "0?Group:OUT") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Goto (macro-record-enable,s,15)
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/1002-0000000e", "1?IN") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Goto (macro-record-enable,s,20)
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/1002-0000000e", "1?MacroExit()") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/1002-0000000e", "dial,,tr,1003") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/1002-0000000e", "1?dial") in new stack
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Goto (macro-dial,s,3)
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/1002-0000000e", "dialparties.agi") in new stack
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: dialparties.agi: Caller ID name is 'test2' number is '1002'
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: dialparties.agi: Methodology of ring is 'none'
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: Added extension 1003 to extension map
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: Extension 1003 cf is disabled
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: Extension 1003 do not disturb is disabled
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: dialparties.agi: Extension 1003 has ExtensionState: 0
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: Checking CW and CFB status for extension 1003
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: dbset CALLTRACE/1003 to 1002
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- dialparties.agi: Filtered ARG3: 1003
[Apr 22 11:39:07] VERBOSE[3422] res_agi.c: -- <SIP/1002-0000000e>AGI Script dialparties.agi completed, returning 0
[Apr 22 11:39:07] VERBOSE[3422] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/1002-0000000e", "SIP/1003,,tr") in new stack
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using SIP RTP TOS bits 184
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using SIP RTP CoS mark 5
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using SIP VRTP TOS bits 136
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using SIP VRTP CoS mark 6
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using UDPTL TOS bits 184
[Apr 22 11:39:07] VERBOSE[3422] netsock.c: == Using UDPTL CoS mark 5
[Apr 22 11:39:07] VERBOSE[3422] app_dial.c: -- Called 1003
[Apr 22 11:39:07] VERBOSE[3422] app_dial.c: -- SIP/1003-0000000f is ringing
[Apr 22 11:39:14] VERBOSE[3424] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:39:14] VERBOSE[3424] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK0af70523;rport=5060
Contact: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=e0825ebf
From: "test2"<sip:1002@192.168.1.150>;tag=as78a5d302
Call-ID: 1d6e8c363e05218a47c5ce146a1f9e12@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: 595

v=0
o=- 12947920748628539 1 IN IP4 192.168.1.4
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.4
t=0 0
a=ice-ufrag:bcad12
a=ice-pwd:a5625d40ce6a2e574455cff20f4b801f
m=audio 49336 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 49336 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 49337 typ host
m=video 51836 RTP/AVP 103
a=rtpmap:103 H263-1998/90000
a=fmtp:103 QCIF=1;CIF=1;VGA=2;I=1;J=1;T=1
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 51836 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 51837 typ host

<------------->
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: --- (12 headers 19 lines) ---
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found RTP audio format 8
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found RTP video format 103
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Found video description format H263-1998 for ID 103
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0xc (ulaw|alaw)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x10000c (ulaw|alaw|h263p)
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.4:49336
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Peer video RTP is at port 192.168.1.4:51836
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: list_route: hop: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279> for address/port to send to
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: set_destination: set destination to 192.168.1.4, port 53412
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: Transmitting (NAT) to 192.168.1.4:53412:
ACK sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK100d6b41;rport
Max-Forwards: 70
From: "test2" <sip:1002@192.168.1.150>;tag=as78a5d302
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=e0825ebf
Contact: <sip:1002@192.168.1.150>
Call-ID: 1d6e8c363e05218a47c5ce146a1f9e12@192.168.1.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
[Apr 22 11:39:17] VERBOSE[3422] app_dial.c: -- SIP/1003-0000000f answered SIP/1002-0000000e
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Audio is at 192.168.1.150 port 14396
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Video is at 192.168.1.150 port 15730
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:39:17] VERBOSE[3422] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.3:33597 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:33597;branch=z9hG4bK13792;received=192.168.1.3;rport=33597
From: <sip:1002@192.168.1.150>;tag=z9hG4bK48666914
To: <sip:1003@192.168.1.150>;tag=as1beb2564
Call-ID: 388580858050@192.168.1.3
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:1003@192.168.1.150>
Content-Type: application/sdp
Content-Length: 340

v=0
o=root 749557174 749557174 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 14396 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 15730 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
a=sendrecv

<------------>
[Apr 22 11:39:17] VERBOSE[3422] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1002-0000000e
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
ACK sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:33597;rport;branch=z9hG4bK39053
Max-Forwards: 70
To: <sip:1003@192.168.1.150>;tag=as1beb2564
From: <sip:1002@192.168.1.150>;tag=z9hG4bK48666914
Call-ID: 388580858050@192.168.1.3
CSeq: 2 ACK
Contact: <sip:1002@192.168.1.3:33597;transport=udp>
Expires: 3600
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:39:17] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:39:18] VERBOSE[3422] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1003-0000000f
[Apr 22 11:39:18] NOTICE[3422] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:39:18] NOTICE[3422] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:39:18] NOTICE[3422] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:39:27] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '149713939278@192.168.1.3' Method: REGISTER
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
BYE sip:1003@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:33597;rport;branch=z9hG4bK46442
Max-Forwards: 70
To: <sip:1003@192.168.1.150>;tag=as1beb2564
From: <sip:1002@192.168.1.150>;tag=z9hG4bK48666914
Call-ID: 388580858050@192.168.1.3
CSeq: 3 BYE
User-Agent: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.3 : 33597 (NAT)
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.3:33597 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:33597;branch=z9hG4bK46442;received=192.168.1.3;rport=33597
From: <sip:1002@192.168.1.150>;tag=z9hG4bK48666914
To: <sip:1003@192.168.1.150>;tag=as1beb2564
Call-ID: 388580858050@192.168.1.3
CSeq: 3 BYE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [h@macro-dial:1] Macro("SIP/1002-0000000e", "hangupcall") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1002-0000000e", "1?noautomon") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Goto (macro-hangupcall,s,3)
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/1002-0000000e", "TOUCH_MONITOR_OUTPUT=") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1002-0000000e", "1?skiprg") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Goto (macro-hangupcall,s,7)
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1002-0000000e", "1?skipblkvm") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Goto (macro-hangupcall,s,10)
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1002-0000000e", "1?theend") in new stack
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Goto (macro-hangupcall,s,12)
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/1002-0000000e", "") in new stack
[Apr 22 11:39:32] VERBOSE[3422] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1002-0000000e' in macro 'hangupcall'
[Apr 22 11:39:32] VERBOSE[3422] chan_sip.c: Scheduling destruction of SIP dialog '1d6e8c363e05218a47c5ce146a1f9e12@192.168.1.150' in 6400 ms (Method: INVITE)
[Apr 22 11:39:32] VERBOSE[3422] chan_sip.c: set_destination: Parsing <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279> for address/port to send to
[Apr 22 11:39:32] VERBOSE[3422] chan_sip.c: set_destination: set destination to 192.168.1.4, port 53412
[Apr 22 11:39:32] VERBOSE[3422] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:53412:
BYE sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK31776137;rport
Max-Forwards: 70
From: "test2" <sip:1002@192.168.1.150>;tag=as78a5d302
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=e0825ebf
Call-ID: 1d6e8c363e05218a47c5ce146a1f9e12@192.168.1.150
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:39:32] VERBOSE[3422] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1003-0000000f
[Apr 22 11:39:32] VERBOSE[3422] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1002-0000000e' in macro 'dial'
[Apr 22 11:39:32] VERBOSE[3422] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/1002-0000000e' in macro 'exten-vm'
[Apr 22 11:39:32] VERBOSE[3422] pbx.c: == Spawn extension (from-internal, 1003, 1) exited non-zero on 'SIP/1002-0000000e'
[Apr 22 11:39:32] VERBOSE[3422] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1002-0000000e
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK31776137;rport=5060
Contact: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>
To: <sip:1003@192.168.1.4:53412;rinstance=7ac1e5d5099fa279>;tag=e0825ebf
From: "test2"<sip:1002@192.168.1.150>;tag=as78a5d302
Call-ID: 1d6e8c363e05218a47c5ce146a1f9e12@192.168.1.150
CSeq: 103 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '1d6e8c363e05218a47c5ce146a1f9e12@192.168.1.150' Method: INVITE
[Apr 22 11:39:32] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '388580858050@192.168.1.3' Method: BYE
[Apr 22 11:39:34] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '000916818627@192.168.1.3' Method: REGISTER
[Apr 22 11:39:35] VERBOSE[3426] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:39:35] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->



<------------->
[Apr 22 11:39:35] VERBOSE[3426] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:39:42] VERBOSE[3428] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:39:42] VERBOSE[3428] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:39:47] VERBOSE[3430] 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 seui » 22 เม.ย. 2011 12:01

Messages X-lite :D

-----------------------------------------------------------------------------------------------------------

[Apr 22 11:44:34] VERBOSE[3442] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:44:34] VERBOSE[3442] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:44:48] VERBOSE[3444] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:44:48] VERBOSE[3444] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
INVITE sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-583d1026b6c87f74-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 1 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: 429

v=0
o=- 12947921085741821 1 IN IP4 192.168.1.4
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.4
t=0 0
a=ice-ufrag:271d07
a=ice-pwd:429434487c60060ed3cd922d2cf8728a
m=audio 56798 RTP/AVP 107 0 97 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 56798 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 56799 typ host

<------------->
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: --- (13 headers 15 lines) ---
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using SIP RTP TOS bits 184
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using SIP RTP CoS mark 5
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using SIP VRTP TOS bits 136
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using SIP VRTP CoS mark 6
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using UDPTL TOS bits 184
[Apr 22 11:44:54] VERBOSE[2907] netsock.c: == Using UDPTL CoS mark 5
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.4 : 53412 (NAT)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Using INVITE request as basis request - YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.4:53412
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-583d1026b6c87f74-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as4e884d61
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 1 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="256ac9b8"
Content-Length: 0


<------------>
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog 'YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.' in 6400 ms (Method: INVITE)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
ACK sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-583d1026b6c87f74-1---d8754z-;rport
Max-Forwards: 70
To: <sip:1002@192.168.1.150>;tag=as4e884d61
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 1 ACK
Content-Length: 0


<------------->
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
INVITE sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-3f4b18c555833143-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
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
Authorization: Digest username="1003",realm="asterisk",nonce="256ac9b8",uri="sip:1002@192.168.1.150",response="baa9dd4669767cad9356cecf438a9520",algorithm=MD5
Content-Length: 429

v=0
o=- 12947921085741821 1 IN IP4 192.168.1.4
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.4
t=0 0
a=ice-ufrag:271d07
a=ice-pwd:429434487c60060ed3cd922d2cf8728a
m=audio 56798 RTP/AVP 107 0 97 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 56798 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 56799 typ host

<------------->
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: --- (14 headers 15 lines) ---
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.4 : 53412 (NAT)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Using INVITE request as basis request - YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.4:53412
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found RTP audio format 107
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found RTP audio format 97
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found RTP audio format 8
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found audio description format BV32 for ID 107
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found audio description format SPEEX for ID 97
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0x20c (ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.4:56798
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Peer doesn't provide video
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: Looking for 1002 in from-internal (domain 192.168.1.150)
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c: list_route: hop: <sip:1003@192.168.1.4:53412>
[Apr 22 11:44:54] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-3f4b18c555833143-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
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:1002@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [1002@from-internal:1] Macro("SIP/1003-00000010", "exten-vm,novm,1002") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/1003-00000010", "user-callerid,") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/1003-00000010", "AMPUSER=1003") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1003-00000010", "0?report") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1003-00000010", "1?Set(REALCALLERIDNUM=1003)") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/1003-00000010", "AMPUSER=1003") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/1003-00000010", "AMPUSERCIDNAME=test3") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1003-00000010", "0?report") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/1003-00000010", "AMPUSERCID=1003") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/1003-00000010", "CALLERID(all)="test3" <1003>") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1003-00000010", "0?Set(CHANNEL(language)=)") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1003-00000010", "0?continue") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:11] Set("SIP/1003-00000010", "__TTL=64") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1003-00000010", "1?continue") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Goto (macro-user-callerid,s,19)
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/1003-00000010", "Using CallerID "test3" <1003>") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:2] Set("SIP/1003-00000010", "RingGroupMethod=none") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:3] Set("SIP/1003-00000010", "VMBOX=novm") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:4] Set("SIP/1003-00000010", "EXTTOCALL=1002") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:5] Set("SIP/1003-00000010", "CFUEXT=") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:6] Set("SIP/1003-00000010", "CFBEXT=") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:7] Set("SIP/1003-00000010", "RT=""") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/1003-00000010", "record-enable,1002,IN") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/1003-00000010", "1?check") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Goto (macro-record-enable,s,4)
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/1003-00000010", "0?MacroExit()") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/1003-00000010", "0?Group:OUT") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Goto (macro-record-enable,s,15)
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/1003-00000010", "1?IN") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Goto (macro-record-enable,s,20)
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-record-enable:20] ExecIf("SIP/1003-00000010", "1?MacroExit()") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/1003-00000010", "dial,,tr,1002") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-dial:1] GotoIf("SIP/1003-00000010", "1?dial") in new stack
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Goto (macro-dial,s,3)
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-dial:3] AGI("SIP/1003-00000010", "dialparties.agi") in new stack
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: dialparties.agi: Starting New Dialparties.agi
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: dialparties.agi: Caller ID name is 'test3' number is '1003'
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: dialparties.agi: Methodology of ring is 'none'
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: Added extension 1002 to extension map
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: Extension 1002 cf is disabled
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: Extension 1002 do not disturb is disabled
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: dialparties.agi: Extension 1002 has ExtensionState: 0
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: Checking CW and CFB status for extension 1002
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: dbset CALLTRACE/1002 to 1003
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- dialparties.agi: Filtered ARG3: 1002
[Apr 22 11:44:54] VERBOSE[3446] res_agi.c: -- <SIP/1003-00000010>AGI Script dialparties.agi completed, returning 0
[Apr 22 11:44:54] VERBOSE[3446] pbx.c: -- Executing [s@macro-dial:7] Dial("SIP/1003-00000010", "SIP/1002,,tr") in new stack
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using SIP RTP TOS bits 184
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using SIP RTP CoS mark 5
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using SIP VRTP TOS bits 136
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using SIP VRTP CoS mark 6
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using UDPTL TOS bits 184
[Apr 22 11:44:54] VERBOSE[3446] netsock.c: == Using UDPTL CoS mark 5
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Audio is at 192.168.1.150 port 12152
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Video is at 192.168.1.150 port 19162
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding video codec 0x40000 (h261) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding video codec 0x200000 (h264) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:33597:
INVITE sip:1002@192.168.1.3:33597;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK515d6c09;rport
Max-Forwards: 70
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
To: <sip:1002@192.168.1.3:33597;transport=udp>
Contact: <sip:1003@192.168.1.150>
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 22 Apr 2011 04:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 442

v=0
o=root 655491103 655491103 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 12152 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 19162 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

---
[Apr 22 11:44:54] VERBOSE[3446] app_dial.c: -- Called 1002
[Apr 22 11:44:54] VERBOSE[3446] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-3f4b18c555833143-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
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:1002@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:44:55] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK515d6c09;rport=5060
To: <sip:1002@192.168.1.3:33597;transport=udp>
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 102 INVITE
Server: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:44:55] VERBOSE[2907] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK515d6c09;rport=5060
To: <sip:1002@192.168.1.3:33597;transport=udp>;tag=3eb2ec81a4ce99ef
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 102 INVITE
Server: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 184
Content-Type: application/sdp

v=0
o=1002@192.168.1.150 0 0 IN IP4 192.168.1.3
s=Session SIP/SDP
c=IN IP4 192.168.1.3
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: --- (9 headers 8 lines) ---
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Found audio description format PCMU for ID 0
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Capabilities: us - 0x3c010e (gsm|ulaw|alaw|g729|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.3:21000
[Apr 22 11:44:56] VERBOSE[2907] chan_sip.c: Peer doesn't provide video
[Apr 22 11:44:56] VERBOSE[3446] app_dial.c: -- SIP/1002-00000011 is ringing
[Apr 22 11:44:56] VERBOSE[3446] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-3f4b18c555833143-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
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:1002@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:44:56] VERBOSE[3446] app_dial.c: -- SIP/1002-00000011 is making progress passing it to SIP/1003-00000010
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK515d6c09;rport=5060
To: <sip:1002@192.168.1.3:33597;transport=udp>;tag=3eb2ec81a4ce99ef
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 102 INVITE
Contact: <sip:1002@192.168.1.3:33597;transport=udp>
Server: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 184
Content-Type: application/sdp

v=0
o=1002@192.168.1.150 0 0 IN IP4 192.168.1.3
s=Session SIP/SDP
c=IN IP4 192.168.1.3
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: --- (10 headers 8 lines) ---
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: list_route: hop: <sip:1002@192.168.1.3:33597;transport=udp>
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: set_destination: Parsing <sip:1002@192.168.1.3:33597;transport=udp> for address/port to send to
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: set_destination: set destination to 192.168.1.3, port 33597
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: Transmitting (NAT) to 192.168.1.3:33597:
ACK sip:1002@192.168.1.3:33597;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK506922c2;rport
Max-Forwards: 70
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
To: <sip:1002@192.168.1.3:33597;transport=udp>;tag=3eb2ec81a4ce99ef
Contact: <sip:1003@192.168.1.150>
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
[Apr 22 11:44:58] VERBOSE[3446] app_dial.c: -- SIP/1002-00000011 answered SIP/1003-00000010
[Apr 22 11:44:58] VERBOSE[3446] chan_sip.c: Audio is at 192.168.1.150 port 10748
[Apr 22 11:44:58] VERBOSE[3446] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:44:58] VERBOSE[3446] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:44:58] VERBOSE[3446] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:44:58] VERBOSE[3446] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-3f4b18c555833143-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
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:1002@192.168.1.150>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 243773779 243773779 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 10748 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

<------------>
[Apr 22 11:44:58] VERBOSE[3446] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1003-00000010
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
ACK sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-db5dd14bf9c56765-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 2 ACK
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1003",realm="asterisk",nonce="256ac9b8",uri="sip:1002@192.168.1.150",response="baa9dd4669767cad9356cecf438a9520",algorithm=MD5
Content-Length: 0


<------------->
[Apr 22 11:44:58] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:44:58] VERBOSE[3446] abstract_jb.c: -- fixed jitterbuffer created on channel SIP/1002-00000011
[Apr 22 11:45:01] VERBOSE[2820] asterisk.c: -- Remote UNIX connection
[Apr 22 11:45:01] VERBOSE[3451] asterisk.c: -- Remote UNIX connection disconnected
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
INVITE sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-30f726827ecc360d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 3 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
Authorization: Digest username="1003",realm="asterisk",nonce="256ac9b8",uri="sip:1002@192.168.1.150",response="baa9dd4669767cad9356cecf438a9520",algorithm=MD5
Content-Length: 708

v=0
o=- 12947921085741821 2 IN IP4 192.168.1.4
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.4
t=0 0
a=ice-ufrag:271d07
a=ice-pwd:429434487c60060ed3cd922d2cf8728a
m=audio 56798 RTP/AVP 107 0 97 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 56798 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 56799 typ host
m=video 50868 RTP/AVP 34 115
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=2
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=1;CIF=1;VGA=2;I=1;J=1;T=1
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.4 50868 typ host
a=candidate:1 2 UDP 659134 192.168.1.4 50869 typ host

<------------->
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: --- (14 headers 23 lines) ---
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.4 : 53412 (NAT)
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP audio format 107
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP audio format 0
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP audio format 97
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP audio format 8
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP audio format 101
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found audio description format BV32 for ID 107
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found audio description format SPEEX for ID 97
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP video format 34
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found RTP video format 115
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found video description format H263 for ID 34
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Found video description format H263-1998 for ID 115
[Apr 22 11:45:02] VERBOSE[2907] 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)
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Peer audio RTP is at port 192.168.1.4:56798
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Peer video RTP is at port 192.168.1.4:50868
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-30f726827ecc360d-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:1002@192.168.1.150>
Content-Length: 0


<------------>
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Audio is at 192.168.1.150 port 10748
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Video is at 192.168.1.150 port 15668
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-30f726827ecc360d-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:1002@192.168.1.150>
Content-Type: application/sdp
Content-Length: 367

v=0
o=root 243773779 243773780 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 10748 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 15668 RTP/AVP 34 115
a=rtpmap:34 H263/90000
a=rtpmap:115 h263-1998/90000
a=sendrecv

<------------>
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
ACK sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-1e1d8c3512b9ebae-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 3 ACK
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1003",realm="asterisk",nonce="256ac9b8",uri="sip:1002@192.168.1.150",response="baa9dd4669767cad9356cecf438a9520",algorithm=MD5
Content-Length: 0


<------------->
[Apr 22 11:45:02] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:45:03] NOTICE[3446] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:45:03] NOTICE[3446] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:45:03] NOTICE[3446] rtp.c: Unknown RTP codec 126 received from '192.168.1.4'
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->



<------------->
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SUBSCRIBE sip:Unknown@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-7aa333547fc5d5d2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1003@192.168.1.150>;tag=as63c1131c
From: <sip:1003@192.168.1.150>;tag=d5a089cb
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 7 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="076df4a5",uri="sip:Unknown@192.168.1.150",response="7ae54d83e9fe1207746f912a3c936ff2",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: --- (14 headers 0 lines) ---
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.4:53412
[Apr 22 11:45:05] NOTICE[2907] chan_sip.c: Correct auth, but based on stale nonce received from '<sip:1003@192.168.1.150>;tag=d5a089cb'
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-7aa333547fc5d5d2-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=d5a089cb
To: <sip:1003@192.168.1.150>;tag=as63c1131c
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 7 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="14dff5a0", stale=true
Content-Length: 0


<------------>
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog 'Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.' in 6400 ms (Method: SUBSCRIBE)
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SUBSCRIBE sip:Unknown@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-838bf6521f8bafca-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1003@192.168.1.150>;tag=as63c1131c
From: <sip:1003@192.168.1.150>;tag=d5a089cb
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 8 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="14dff5a0",uri="sip:Unknown@192.168.1.150",response="873dfd0dcba94e9de5906c7f9994b40f",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: --- (14 headers 0 lines) ---
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: Found peer '1003' for '1003' from 192.168.1.4:53412
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: Scheduling destruction of SIP dialog 'Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.' in 310000 ms (Method: SUBSCRIBE)
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-838bf6521f8bafca-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=d5a089cb
To: <sip:1003@192.168.1.150>;tag=as63c1131c
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 8 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


<------------>
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.4:53412:
NOTIFY sip:1003@192.168.1.4:53412 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK270b306e;rport
Max-Forwards: 70
Route: <sip:1003@192.168.1.4:53412>
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as63c1131c
To: <sip:1003@192.168.1.4:53412>;tag=d5a089cb
Contact: <sip:Unknown@192.168.1.150>
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 105 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)

---
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK270b306e;rport=5060
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1003@192.168.1.4:53412>;tag=d5a089cb
From: "Unknown"<sip:Unknown@192.168.1.150>;tag=as63c1131c
Call-ID: Yzk1MzQ0NTYxOGRiZDAxNjAxNWE4M2Y0YzFmMTg3ZmI.
CSeq: 105 NOTIFY
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
[Apr 22 11:45:05] VERBOSE[2907] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 22 11:45:07] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->


<------------->
[Apr 22 11:45:08] VERBOSE[2907] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:33597:
OPTIONS sip:1002@192.168.1.3:33597;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7d8a18c6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as0597400a
To: <sip:1002@192.168.1.3:33597;transport=udp>
Contact: <sip:Unknown@192.168.1.150>
Call-ID: 5933491b6f2a9572363d5ec0375cc6b4@192.168.1.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 22 Apr 2011 04:45:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Apr 22 11:45:08] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK7d8a18c6;rport=5060
To: <sip:1002@192.168.1.3:33597;transport=udp>
From: "Unknown" <sip:Unknown@192.168.1.150>;tag=as0597400a
Call-ID: 5933491b6f2a9572363d5ec0375cc6b4@192.168.1.150
CSeq: 102 OPTIONS
Contact: <sip:1002@192.168.1.3:33597;transport=udp>
Content-Length: 0


<------------->
[Apr 22 11:45:08] VERBOSE[2907] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 22 11:45:08] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '5933491b6f2a9572363d5ec0375cc6b4@192.168.1.150' Method: OPTIONS
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.4:53412 --->
BYE sip:1002@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-1606ff81163bc889-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003@192.168.1.4:53412>
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
From: <sip:1003@192.168.1.150>;tag=f8e376f8
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 4 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="1003",realm="asterisk",nonce="256ac9b8",uri="sip:1002@192.168.1.150",response="4c5900441392312a07e992a1b0430ef0",algorithm=MD5
Content-Length: 0


<------------->
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c: --- (11 headers 0 lines) ---
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c: Sending to 192.168.1.4 : 53412 (NAT)
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c:
<--- Transmitting (NAT) to 192.168.1.4:53412 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:53412;branch=z9hG4bK-d8754z-1606ff81163bc889-1---d8754z-;received=192.168.1.4;rport=53412
From: <sip:1003@192.168.1.150>;tag=f8e376f8
To: <sip:1002@192.168.1.150>;tag=as3cdfb970
Call-ID: YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.
CSeq: 4 BYE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [h@macro-dial:1] Macro("SIP/1003-00000010", "hangupcall") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1003-00000010", "1?noautomon") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Goto (macro-hangupcall,s,3)
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/1003-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1003-00000010", "1?skiprg") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Goto (macro-hangupcall,s,7)
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1003-00000010", "1?skipblkvm") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Goto (macro-hangupcall,s,10)
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1003-00000010", "1?theend") in new stack
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Goto (macro-hangupcall,s,12)
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/1003-00000010", "") in new stack
[Apr 22 11:45:25] VERBOSE[3446] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1003-00000010' in macro 'hangupcall'
[Apr 22 11:45:25] VERBOSE[3446] chan_sip.c: Scheduling destruction of SIP dialog '3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150' in 11392 ms (Method: INVITE)
[Apr 22 11:45:25] VERBOSE[3446] chan_sip.c: set_destination: Parsing <sip:1002@192.168.1.3:33597;transport=udp> for address/port to send to
[Apr 22 11:45:25] VERBOSE[3446] chan_sip.c: set_destination: set destination to 192.168.1.3, port 33597
[Apr 22 11:45:25] VERBOSE[3446] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.3:33597:
BYE sip:1002@192.168.1.3:33597;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK39c26a96;rport
Max-Forwards: 70
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
To: <sip:1002@192.168.1.3:33597;transport=udp>;tag=3eb2ec81a4ce99ef
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 22 11:45:25] VERBOSE[3446] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1002-00000011
[Apr 22 11:45:25] VERBOSE[3446] app_macro.c: == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1003-00000010' in macro 'dial'
[Apr 22 11:45:25] VERBOSE[3446] app_macro.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/1003-00000010' in macro 'exten-vm'
[Apr 22 11:45:25] VERBOSE[3446] pbx.c: == Spawn extension (from-internal, 1002, 1) exited non-zero on 'SIP/1003-00000010'
[Apr 22 11:45:25] VERBOSE[3446] abstract_jb.c: -- fixed jitterbuffer destroyed on channel SIP/1003-00000010
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c:
<--- SIP read from UDP:192.168.1.3:33597 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK39c26a96;rport=5060
To: <sip:1002@192.168.1.3:33597;transport=udp>;tag=3eb2ec81a4ce99ef
From: "test3" <sip:1003@192.168.1.150>;tag=as513b3cbb
Call-ID: 3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150
CSeq: 103 BYE
Server: Sipbu/2.0.1 beta/GT-S5830
Content-Length: 0


<------------->
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog '3b2054fb3cdf7f5342d20c3865b8f949@192.168.1.150' Method: INVITE
[Apr 22 11:45:25] VERBOSE[2907] chan_sip.c: Really destroying SIP dialog 'YWM3YTJjZjZiMmZjMDg0ZWU2YmIxM2JkMmU0MWVhZmU.' Method: BYE
[Apr 22 11:45:30] VERBOSE[3454] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:45:30] VERBOSE[3454] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:45:36] VERBOSE[3456] manager.c: == Manager 'admin' logged on from 127.0.0.1
[Apr 22 11:45:36] VERBOSE[3456] manager.c: == Manager 'admin' logged off from 127.0.0.1
[Apr 22 11:45:40] VERBOSE[3458] manager.c: == Manager 'admin' logged on from 127.0.0.1

-------------------------------------------------------------------------------------------------------------------------------------

ไม่รู้ว่าทำถูกรึเปล่าครับ ขอบคุณที่ให้คำแนะนำครับ รบกวนด้วยนะครับ :D
seui
Silver Member
 
โพสต์: 20
ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
ข้างบน

Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย nuiz » 23 เม.ย. 2011 10:32

อืม...
ตอนกดโทรจาก GT-S5830 ก็ใช้ Video Codec เป็น h263-1998 และเวลา x-lite รับสาย มันก็ใช้ Video เป็น h263-1998 ซึ่ง h263-1998 นี้ Asterisk มองว่าเป็น h263p แต่ทำไมไม่มีภาพที่ส่งจาก Sipdroid ไป (ตามที่คุณ seui บอกมา) งงๆอยู่ครับ หรือว่า Sipdroid มันเช็คได้ว่าที่มันกำลังคุยอยู่นี้ไม่ใช่ PBXes มันก็เลยไม่ยอมส่งภาพออกมา (ตามที่อ่านเจอใน FAQ) สงสัยอาจต้องปลอมตัวเป็น PBXes หล่ะครับ แต่ทำไงไม่รู้

ส่วนข้อความที่ว่า rtp.c: Unknown RTP codec 126 received from '192.168.1.4' นี้ เกิดจาก X-Lite เองครับ มันส่งอะไรมาไม่รู้ ทำให้ Asterisk งง แต่ก็ไม่มีผลในการใช้งานครับ

ขอบคุณมากๆสำหรับ SIP Messages นะครับ ได้ความรู้เพิ่มอีกพอสมควรเลย
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
Diamond Member
 
โพสต์: 6995
ลงทะเบียนเมื่อ: 24 มี.ค. 2010 09:33
ข้างบน

Re: ถามปัญหาการส่ง videocall ครับ

โพสต์โดย seui » 23 เม.ย. 2011 12:35

nuiz เขียน:อืม...
ตอนกดโทรจาก GT-S5830 ก็ใช้ Video Codec เป็น h263-1998 และเวลา x-lite รับสาย มันก็ใช้ Video เป็น h263-1998 ซึ่ง h263-1998 นี้ Asterisk มองว่าเป็น h263p แต่ทำไมไม่มีภาพที่ส่งจาก Sipdroid ไป (ตามที่คุณ seui บอกมา) งงๆอยู่ครับ หรือว่า Sipdroid มันเช็คได้ว่าที่มันกำลังคุยอยู่นี้ไม่ใช่ PBXes มันก็เลยไม่ยอมส่งภาพออกมา (ตามที่อ่านเจอใน FAQ) สงสัยอาจต้องปลอมตัวเป็น PBXes หล่ะครับ แต่ทำไงไม่รู้

ส่วนข้อความที่ว่า rtp.c: Unknown RTP codec 126 received from '192.168.1.4' นี้ เกิดจาก X-Lite เองครับ มันส่งอะไรมาไม่รู้ ทำให้ Asterisk งง แต่ก็ไม่มีผลในการใช้งานครับ

ขอบคุณมากๆสำหรับ SIP Messages นะครับ ได้ความรู้เพิ่มอีกพอสมควรเลย



ขอบคุณที่ช่วยดูให้ครับผมมือใหม่ครับก็ดูไม่ค่อยเป็น และขอบคุณสำหรับคำตอบ :D
seui
Silver Member
 
โพสต์: 20
ลงทะเบียนเมื่อ: 19 ส.ค. 2010 15:55
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