Public feed #528961

jkiff

public text v1 · immutable
Raw Edit
·published 2007-06-01 19:08 UTC
rendered paste body

    -- Executing Answer("SIP/299-081dd1a8", "") in new stack
    -- Executing Wait("SIP/299-081dd1a8", "2") in new stack
    -- Executing SetCIDNum("SIP/299-081dd1a8", "8594558658") in new stack
    -- Executing Dial("SIP/299-081dd1a8", "Zap/g1/18594558658") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g1/18594558658
    -- Zap/1-1 is proceeding passing it to SIP/299-081dd1a8
    -- Accepting call from '8594558658' to '8594558658' on channel 0/2, span 1
    -- Executing Answer("Zap/2-1", "") in new stack
    -- Executing Wait("Zap/2-1", "1") in new stack
    -- Zap/1-1 answered SIP/299-081dd1a8
    -- Executing GotoIfTime("Zap/2-1", "08:00-17:10|mon-fri|*|*?receptionist|s|1") in new stack
    -- Goto (receptionist,s,1)
    -- Executing Answer("Zap/2-1", "") in new stack
    -- Executing Wait("Zap/2-1", "2") in new stack
    -- Executing Set("Zap/2-1", "__TRANSFER_CONTEXT=internal") in new stack
    -- Executing Queue("Zap/2-1", "reception-queue|rtn|||30") in new stack
    -- outgoing agentcall, to agent '213', on 'Local/213@agents-6f0e,1'
    -- Called Agent/213
    -- Called SIP/201
    -- Executing Dial("Local/213@agents-6f0e,2", "SIP/213") in new stack
We're at 192.168.1.42 port 19838
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.1.222:5061:
INVITE sip:213@192.168.1.222:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK0c7a7a80;rport
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
To: <sip:213@192.168.1.222:5061>
Contact: <sip:8594558658@192.168.1.42>
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 01 Jun 2007 19:02:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 3663 3663 IN IP4 192.168.1.42
s=session
c=IN IP4 192.168.1.42
t=0 0
m=audio 19838 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=silenceSupp:off - - - -

---
    -- Called 213
    -- SIP/201-081e6608 is ringing
asterisk*CLI>
<-- SIP read from 192.168.1.222:5061:
SIP/2.0 100 Trying
To: <sip:213@192.168.1.222:5061>
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK0c7a7a80
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (8 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 192.168.1.222:5061:
SIP/2.0 180 Ringing
To: <sip:213@192.168.1.222:5061>;tag=90ccaf84a8321afi1
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK0c7a7a80
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (8 headers 0 lines) ---
    -- SIP/213-081ef008 is ringing
    -- Agent/213 is ringing
asterisk*CLI>
<-- SIP read from 192.168.1.222:5061:
SIP/2.0 200 OK
To: <sip:213@192.168.1.222:5061>;tag=90ccaf84a8321afi1
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK0c7a7a80
Contact: John Kiffmeyer <sip:213@192.168.1.222:5061>
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 30948 30948 IN IP4 192.168.1.222
s=-
c=IN IP4 192.168.1.222
t=0 0
m=audio 16442 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.222:16442
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:213@192.168.1.222:5061>
set_destination: Parsing <sip:213@192.168.1.222:5061> for address/port to send to
set_destination: set destination to 192.168.1.222, port 5061
Transmitting (no NAT) to 192.168.1.222:5061:
ACK sip:213@192.168.1.222:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK0cafa105;rport
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
To: <sip:213@192.168.1.222:5061>;tag=90ccaf84a8321afi1
Contact: <sip:8594558658@192.168.1.42>
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/213-081ef008 answered Local/213@agents-6f0e,2
    -- Agent/213 answered Zap/2-1
  == Spawn extension (agents, 213, 1) exited non-zero on 'Local/213@agents-6f0e,2'
<-- SIP read from 192.168.1.222:5061:
NOTIFY sip:192.168.1.42 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.222:5061:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db;received=192.168.1.222
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>;tag=as2359dee7
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 192.168.1.222:5061:
NOTIFY sip:192.168.1.42 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.222:5061:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db;received=192.168.1.222
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>;tag=as764a6ae0
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
<-- SIP read from 192.168.1.222:5061:
NOTIFY sip:192.168.1.42 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.222:5061:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db;received=192.168.1.222
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>;tag=as2d700c5e
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Scheduling destruction of call '524b1555635eec4b110402c97dd3778b@192.168.1.42' in 32000 ms
set_destination: Parsing <sip:213@192.168.1.222:5061> for address/port to send to
set_destination: set destination to 192.168.1.222, port 5061
Reliably Transmitting (no NAT) to 192.168.1.222:5061:
BYE sip:213@192.168.1.222:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK6e2c1c82;rport
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
To: <sip:213@192.168.1.222:5061>;tag=90ccaf84a8321afi1
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (receptionist, s, 4) exited non-zero on 'Zap/2-1'
    -- Hungup 'Zap/2-1'
<-- SIP read from 192.168.1.222:5061:
SIP/2.0 200 OK
To: <sip:213@192.168.1.222:5061>;tag=90ccaf84a8321afi1
From: "COMMERCE,VOLTAIC" <sip:8594558658@192.168.1.42>;tag=as24e9b083
Call-ID: 524b1555635eec4b110402c97dd3778b@192.168.1.42
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK6e2c1c82
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '524b1555635eec4b110402c97dd3778b@192.168.1.42'
    -- Channel 0/1, span 1 got hangup request
    -- Hungup 'Zap/1-1'
  == Spawn extension (internal, 4558658, 4) exited non-zero on 'SIP/299-081dd1a8'
<-- SIP read from 192.168.1.222:5061:
NOTIFY sip:192.168.1.42 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2000-2.0.13(g)
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.222:5061:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 192.168.1.222:5061;branch=z9hG4bK-49a5a8db;received=192.168.1.222
From: John Kiffmeyer <sip:213@192.168.1.42>;tag=adc1b503b629cedco1
To: <sip:192.168.1.42>;tag=as4605f592
Call-ID: d1806bcb-919946b4@192.168.1.222
CSeq: 7 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.222:5061:
OPTIONS sip:213@192.168.1.222:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK77838edc;rport
From: "asterisk" <sip:asterisk@192.168.1.42>;tag=as35166b5d
To: <sip:213@192.168.1.222:5061>
Contact: <sip:asterisk@192.168.1.42>
Call-ID: 0bd95ba12d5cdaad3dad49bb487b1fa2@192.168.1.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 01 Jun 2007 19:02:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
<-- SIP read from 192.168.1.222:5061:
SIP/2.0 200 OK
To: <sip:213@192.168.1.222:5061>;tag=2ac8414fdae5cf00i1
From: "asterisk" <sip:asterisk@192.168.1.42>;tag=as35166b5d
Call-ID: 0bd95ba12d5cdaad3dad49bb487b1fa2@192.168.1.42
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK77838edc
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


--- (10 headers 0 lines) ---
Destroying call '0bd95ba12d5cdaad3dad49bb487b1fa2@192.168.1.42'