rendered paste body -- SIP/1001-085dd138 is making progress passing it to SIP/3024-085d91d0
-- Executing [7775@incoming:1] Goto("SIP/1005-085c1578", "answer|1") in new stack
-- Goto (incoming,answer,1)
-- Executing [answer@incoming:1] Answer("SIP/1005-085c1578", "") in new stack
-- Executing [answer@incoming:2] Wait("SIP/1005-085c1578", "1") in new stack
-- Executing [answer@incoming:3] NoOp("SIP/1005-085c1578", "1005") in new stack
-- Executing [answer@incoming:4] NoOp("SIP/1005-085c1578", "") in new stack
-- Executing [answer@incoming:5] NoOp("SIP/1005-085c1578", "") in new stack
-- Executing [answer@incoming:6] Set("SIP/1005-085c1578", "TIMEOUT(digit)=7") in new stack
-- Digit timeout set to 7
-- Executing [answer@incoming:7] BackGround("SIP/1005-085c1578", "11") in new stack
-- <SIP/1005-085c1578> Playing '11' (language 'ru')
-- SIP/1001-085dd138 answered SIP/3024-085d91d0
-- Executing [answer@incoming:8] SetMusicOnHold("SIP/1005-085c1578", "default") in new stack
-- Executing [answer@incoming:9] WaitExten("SIP/1005-085c1578", "7") in new stack
asterisk*CLI>
-- Timeout on SIP/1005-085c1578, going to 't'
-- Executing [t@incoming:1] Goto("SIP/1005-085c1578", "in_call|s|1") in new stack
-- Goto (in_call,s,1)
-- Executing [s@in_call:1] Dial("SIP/1005-085c1578", "SIP/3000|15|tr") in new stack
Audio is at 192.168.4.2 port 14634
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.150:5060:
INVITE sip:3000@192.168.0.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport
From: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
To: <sip:3000@192.168.0.150:5060>
Contact: <sip:1005@192.168.4.2>
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Jun 2007 10:21:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 2666 2666 IN IP4 192.168.4.2
s=session
c=IN IP4 192.168.4.2
t=0 0
m=audio 14634 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 3000
asterisk*CLI>
<--- SIP read from 192.168.0.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport
From: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
To: <sip:3000@192.168.0.150:5060>
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.16
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from 192.168.0.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport
From: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
To: <sip:3000@192.168.0.150:5060>;tag=6c573764c0b95fc9
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:3000@192.168.0.150:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/3000-085d7c50 is ringing
asterisk*CLI>
<--- SIP read from 192.168.0.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK4deb49e9;rport
From: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
To: <sip:3000@192.168.0.150:5060>;tag=6c573764c0b95fc9
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.16
Contact: <sip:3000@192.168.0.150:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 210
v=0
o=3000 8000 8000 IN IP4 192.168.0.150
s=SIP Call
c=IN IP4 192.168.0.150
t=0 0
m=audio 5004 RTP/AVP 0 96
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-11
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port 192.168.0.150:5004
Found description format PCMU for ID 0
Found description format telephone-event for ID 96
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.150:5004
list_route: hop: <sip:3000@192.168.0.150:5060>
set_destination: Parsing <sip:3000@192.168.0.150:5060> for address/port to send to
set_destination: set destination to 192.168.0.150, port 5060
Transmitting (no NAT) to 192.168.0.150:5060:
ACK sip:3000@192.168.0.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:5060;branch=z9hG4bK33e41028;rport
From: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
To: <sip:3000@192.168.0.150:5060>;tag=6c573764c0b95fc9
Contact: <sip:1005@192.168.4.2>
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/3000-085d7c50 answered SIP/1005-085c1578
-- SIP/3007-085f2c18 answered SIP/3012-085c54e0
== Spawn extension (macro-office_dial, s, 1) exited non-zero on 'SIP/3012-085c54e0' in macro 'office_dial'
== Spawn extension (macro-office_dial, s, 1) exited non-zero on 'SIP/3012-085c54e0'
-- Executing [3007@default:1] Macro("SIP/3012-085bfff8", "office_dial|3007|20") in new stack
-- Executing [s@macro-office_dial:1] Dial("SIP/3012-085bfff8", "SIP/3007|20|tr") in new stack
-- Called 3007
-- SIP/3007-085e6490 is ringing
-- Started music on hold, class 'default', on SIP/1005-085c1578
-- <SIP/3000-085d7c50> Playing 'pbx-transfer' (language 'ru')
-- Executing [600@default:1] Playback("Local/600@default-fd27,2", "demo-echotest") in new stack
-- <Local/600@default-fd27,2> Playing 'demo-echotest' (language 'en')
[Jun 22 13:21:28] WARNING[3478]: cdr.c:767 ast_cdr_init: CDR already initialized on 'Local/600@default-fd27,1'
-- Executing [980296922041@default:1] Macro("SIP/3007-085f50e8", "velcom_dial|6922041") in new stack
-- Executing [s@macro-velcom_dial:1] Dial("SIP/3007-085f50e8", "Sip/1001/80296922041|60|t") in new stack
-- Called 1001/80296922041
asterisk*CLI> quit
<--- SIP read from 192.168.0.150:5060 --->
BYE sip:1005@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bKcc9952e5e5ae3857
From: <sip:3000@192.168.0.150:5060>;tag=6c573764c0b95fc9
To: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 22251 BYE
User-Agent: Grandstream GXP2000 1.1.0.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.0.150 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bKcc9952e5e5ae3857;received=192.168.0.150
From: <sip:3000@192.168.0.150:5060>;tag=6c573764c0b95fc9
To: "1005" <sip:1005@192.168.4.2>;tag=as2c5c71d7
Call-ID: 10b26a620096b56756fa788339a7a150@192.168.4.2
CSeq: 22251 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1005@192.168.4.2>
Content-Length: 0
<------------>
-- Stopped music on hold on SIP/1005-085c1578
-- <Local/600@default-fd27,1> Playing 'beep' (language 'en')
== Spawn extension (in_call, s, 1) exited non-zero on 'Transfered/SIP/1005-085c1578<ZOMBIE>'
Really destroying SIP dialog '10b26a620096b56756fa788339a7a150@192.168.4.2' Method: BYE
-- SIP/1001-085f2c18 is making progress passing it to SIP/3007-085f50e8
asterisk*CLI> quit
[root@asterisk ~]#