All pastes #68443 Raw Edit

Miscellany

public text v1 · immutable
#68443 ·published 2006-06-21 17:03 UTC
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asterisk*CLI> 

--- (8 headers 0 lines)---

asterisk*CLI> 
Looking for s in default (domain 192.168.0.10)

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.103:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.103;rport;branch=z9hG4bKc0a8006700000010449977674d65efbe00004fbf;received=192.168.0.103
From: <sip:tsr003@192.168.0.10>;tag=41519955411501524070
To: <sip:192.168.0.10:5060>;tag=as6da0be45
Call-ID: 227F676E-1DD2-11B2-A0A9-EDAE4F980423@192.168.0.103
CSeq: 8014 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0


---
Destroying call '227F676E-1DD2-11B2-A0A9-EDAE4F980423@192.168.0.103'

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3dd352a8;rport
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as4a3aabda
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as051bec49
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Call-ID: 0dd66ce37e895a6732bd016d723d5257@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



asterisk*CLI> 

--- (10 headers 0 lines)---

asterisk*CLI> 
Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT)

asterisk*CLI> 
Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3dd352a8;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as4a3aabda
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as051bec49
Call-ID: 0dd66ce37e895a6732bd016d723d5257@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---

asterisk*CLI> 
  == Spawn extension (rca, **90080817914807480034, 6) exited non-zero on 'SIP/tsr002-699c'

asterisk*CLI> 
set_destination: Parsing <sip:tsr002@192.168.0.102:5060> for address/port to send to

asterisk*CLI> 
set_destination: set destination to 192.168.0.102, port 5060

asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
BYE sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK5dee13c4;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as509756c9
To: <sip:tsr002@192.168.0.102:5060>;tag=4151813532565042683
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 0954a3a6090263c06ca685ba2cb730be@192.168.0.10
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK5dee13c4
Content-Length: 0
Call-ID: 0954a3a6090263c06ca685ba2cb730be@192.168.0.10
CSeq: 103 BYE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as509756c9
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=4151813532565042683



---
Destroying call '507ffb9b33b4f669472620795c524744@XXX.XXX.XXX.XXX(VOIP Server)'

asterisk*CLI> 
Destroying call '597D789E-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109'

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
OPTIONS sip:192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499776533e7297f0000001e
Content-Length: 0
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 11 OPTIONS
From: <sip:tsr002@192.168.0.10>;tag=4151993217731562104
Max-Forwards: 70
To: <sip:192.168.0.10:5060>



--- (8 headers 0 lines)---

asterisk*CLI> 
Looking for s in default (domain 192.168.0.10)

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499776533e7297f0000001e;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=4151993217731562104
To: <sip:192.168.0.10:5060>;tag=as27d9de9b
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 11 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0


---

asterisk*CLI> 
Destroying call 'B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102'

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
REGISTER sip:192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020
Content-Length: 0
Contact: <sip:tsr002@192.168.0.102:5060>
Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 3 REGISTER
From: <sip:tsr002@192.168.0.10>;tag=41519936791451098011
Max-Forwards: 70
To: <sip:tsr002@192.168.0.10>



asterisk*CLI> 

--- (9 headers 0 lines)---

asterisk*CLI> 
Using latest REGISTER request as basis request

asterisk*CLI> 
Sending to 192.168.0.102 : 5060 (NAT)

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=41519936791451098011
To: <sip:tsr002@192.168.0.10>
Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:tsr002@192.168.0.10>
Content-Length: 0


---

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977655472823300000020;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=41519936791451098011
To: <sip:tsr002@192.168.0.10>;tag=as5687ef9f
Call-ID: A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:tsr002@192.168.0.102:5060>;expires=120
Date: Wed, 21 Jun 2006 16:44:21 GMT
Content-Length: 0


---

asterisk*CLI> 
Scheduling destruction of call 'A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102' in 15000 ms

asterisk*CLI> 
    -- Attempting call on SIP/tsr002 for **90080817924807480034@rca:1 (Retry 1)

asterisk*CLI> 
We're at 192.168.0.10 port 18450

asterisk*CLI> 
Adding codec 0x2 (gsm) to SDP

asterisk*CLI> 
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

asterisk*CLI> 
13 headers, 11 lines

asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
INVITE sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK38ed3e5b;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as1861d1de
To: <sip:tsr002@192.168.0.102:5060>
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jun 2006 16:44:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 5521 5521 IN IP4 192.168.0.10
s=session
c=IN IP4 192.168.0.10
t=0 0
m=audio 18450 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b
Content-Length: 0
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as1861d1de
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767



asterisk*CLI> 

--- (7 headers 0 lines)---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b
Content-Length: 0
Contact: <sip:tsr002@192.168.0.102:5060>
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as1861d1de
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767



--- (8 headers 0 lines)---

asterisk*CLI> 
<-- SIP read from 192.168.0.183:62832: 
OPTIONS sip:192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499775f0000780e00000279
Content-Length: 0
Call-ID: 0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102
CSeq: 476 OPTIONS
From: <sip:kyle@192.168.0.10>;tag=4333353113333
Max-Forwards: 70
To: <sip:192.168.0.10:5060>



--- (8 headers 0 lines)---
Looking for s in default (domain 192.168.0.10)
Transmitting (no NAT) to 192.168.0.183:62832:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499775f0000780e00000279;received=192.168.0.183
From: <sip:kyle@192.168.0.10>;tag=4333353113333
To: <sip:192.168.0.10:5060>;tag=as0efb349a
Call-ID: 0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102
CSeq: 476 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0


---
Destroying call '0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102'

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK38ed3e5b
Content-Length: 220
Contact: <sip:tsr002@192.168.0.102:5060>
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
Content-Type: application/sdp
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as1861d1de
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767

v=0
o=- 3359897061 3359897061 IN IP4 192.168.0.102
s=SJphone
c=IN IP4 192.168.0.102
t=0 0
a=direction:active
m=audio 49154 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16


asterisk*CLI> 

--- (9 headers 10 lines)---

asterisk*CLI> 
Found RTP audio format 3

asterisk*CLI> 
Found RTP audio format 101

asterisk*CLI> 
Peer audio RTP is at port 192.168.0.102:49154

asterisk*CLI> 
Found description format GSM
Found description format telephone-event

asterisk*CLI> 
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)

asterisk*CLI> 
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

asterisk*CLI> 
list_route: hop: <sip:tsr002@192.168.0.102:5060>

asterisk*CLI> 
set_destination: Parsing <sip:tsr002@192.168.0.102:5060> for address/port to send to

asterisk*CLI> 
set_destination: set destination to 192.168.0.102, port 5060

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
ACK sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK08184819;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as1861d1de
To: <sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

asterisk*CLI> 
    -- Executing SetCIDNum("SIP/tsr002-01ba", "8778994498") in new stack
    -- Executing SetCIDName("SIP/tsr002-01ba", "PMG") in new stack
    -- Executing SetVar("SIP/tsr002-01ba", "CALLFILENAME=9008081792") in new stack
    -- Executing SetVar("SIP/tsr002-01ba", "CALLPATH=/var/calls") in new stack
    -- Executing Monitor("SIP/tsr002-01ba", "wav|/var/calls/9008081792|mb") in new stack
    -- Executing Dial("SIP/tsr002-01ba", "SIP/14807480034@cvcdial|999|r") in new stack
We're at XXX.XXX.XXX.XXX(Asterisk Server) port 19578
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
INVITE sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;rport
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jun 2006 16:44:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 5521 5521 IN IP4 XXX.XXX.XXX.XXX(Asterisk Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(Asterisk Server)
t=0 0
m=audio 19578 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 14807480034@cvcdial

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Length: 0



--- (10 headers 0 lines)---

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as1546cb93
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 23728 23728 IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
t=0 0
m=audio 16950 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


asterisk*CLI> 

--- (12 headers 0 lines)---

asterisk*CLI> 
Scheduling destruction of call '3cbee32f1d4f45f158ec08635cfb6a2a@127.0.0.1' in 32000 ms

asterisk*CLI> 
Jun 21 09:44:31 NOTICE[5788]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for 66X235X234X150 is 120 sec (Scheduling reregistration in 105 s)

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK0deb0424;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as1546cb93
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 23728 23729 IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
t=0 0
m=audio 16950 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


asterisk*CLI> 

--- (11 headers 11 lines)---

asterisk*CLI> 
Found RTP audio format 0
Found RTP audio format 3

asterisk*CLI> 
Found RTP audio format 101

asterisk*CLI> 
Peer audio RTP is at port XXX.XXX.XXX.XXX(VOIP Server):16950

asterisk*CLI> 
Found description format PCMU

asterisk*CLI> 
Found description format GSM
Found description format telephone-event

asterisk*CLI> 
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)

asterisk*CLI> 
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

asterisk*CLI> 
list_route: hop: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>

asterisk*CLI> 
set_destination: Parsing <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)> for address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX(VOIP Server), port 5060
Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
ACK sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK6aca22b1;rport
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as1546cb93
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---

asterisk*CLI> 
Destroying call '70CFB4BC-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109'

asterisk*CLI> 
Destroying call 'A187E66E-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102'

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
OPTIONS sip:192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499777937492daf00000025
Content-Length: 0
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 12 OPTIONS
From: <sip:tsr002@192.168.0.10>;tag=41520132181645676057
Max-Forwards: 70
To: <sip:192.168.0.10:5060>



--- (8 headers 0 lines)---
Looking for s in default (domain 192.168.0.10)
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499777937492daf00000025;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=41520132181645676057
To: <sip:192.168.0.10:5060>;tag=as282ea764
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 12 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0

---
Destroying call '0488B243-9367-41F7-B989-E169C1AE9B46@192.168.100.102'

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK5ded3311;rport
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as1546cb93
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



asterisk*CLI> 

--- (10 headers 0 lines)---

asterisk*CLI> 
Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT)

asterisk*CLI> 
Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK5ded3311;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as1546cb93
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as66fe032e
Call-ID: 5a9d853967f57abe0ebbf71b3d34d9ec@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---

asterisk*CLI> 
  == Spawn extension (rca, **90080817924807480034, 6) exited non-zero on 'SIP/tsr002-01ba'

asterisk*CLI> 
set_destination: Parsing <sip:tsr002@192.168.0.102:5060> for address/port to send to

asterisk*CLI> 
set_destination: set destination to 192.168.0.102, port 5060

asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
BYE sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4446f5a4;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as1861d1de
To: <sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4446f5a4
Content-Length: 0
Call-ID: 34dca93c3d224c2461eb737f29d6a56f@192.168.0.10
CSeq: 103 BYE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as1861d1de
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=41519938851223188767

---
Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms

asterisk*CLI> 
    -- Attempting call on SIP/tsr002 for **90080817934807480034@rca:1 (Retry 1)

asterisk*CLI> 
We're at 192.168.0.10 port 19212

asterisk*CLI> 
Adding codec 0x2 (gsm) to SDP

asterisk*CLI> 
Adding codec 0x4 (ulaw) to SDP

asterisk*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP

asterisk*CLI> 
13 headers, 11 lines

asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
INVITE sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK74a79ecd;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as2393f69f
To: <sip:tsr002@192.168.0.102:5060>
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jun 2006 16:44:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 5521 5521 IN IP4 192.168.0.10
s=session
c=IN IP4 192.168.0.10
t=0 0
m=audio 19212 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd
Content-Length: 0
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as2393f69f
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119



--- (7 headers 0 lines)---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd
Content-Length: 0
Contact: <sip:tsr002@192.168.0.102:5060>
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as2393f69f
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119



---

asterisk*CLI> 
Transmitting (no NAT) to 192.168.100.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.100;rport;branch=z9hG4bKc0a86464000000104499777900006c8a0000027d;received=192.168.0.183
From: <sip:kyle@192.168.0.10>;tag=4335912516060
To: <sip:kyle@192.168.0.10>;tag=as765ba6a4
Call-ID: 05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102
CSeq: 129 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:kyle@192.168.100.100:5060>;expires=120
Date: Wed, 21 Jun 2006 16:44:49 GMT
Content-Length: 0


---

asterisk*CLI> 
Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK74a79ecd
Content-Length: 220
Contact: <sip:tsr002@192.168.0.102:5060>
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
Content-Type: application/sdp
CSeq: 102 INVITE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as2393f69f
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119

v=0
o=- 3359897088 3359897088 IN IP4 192.168.0.102
s=SJphone
c=IN IP4 192.168.0.102
t=0 0
a=direction:active
m=audio 49156 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16


asterisk*CLI> 

--- (9 headers 10 lines)---

asterisk*CLI> 
Found RTP audio format 3

asterisk*CLI> 
Found RTP audio format 101

asterisk*CLI> 
Peer audio RTP is at port 192.168.0.102:49156

asterisk*CLI> 
Found description format GSM

asterisk*CLI> 
Found description format telephone-event

asterisk*CLI> 
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)

asterisk*CLI> 
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

asterisk*CLI> 
list_route: hop: <sip:tsr002@192.168.0.102:5060>

asterisk*CLI> 
set_destination: Parsing <sip:tsr002@192.168.0.102:5060> for address/port to send to

asterisk*CLI> 
set_destination: set destination to 192.168.0.102, port 5060

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
ACK sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK111b232f;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as2393f69f
To: <sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

asterisk*CLI> 
    -- Executing SetCIDNum("SIP/tsr002-1986", "8778994498") in new stack
    -- Executing SetCIDName("SIP/tsr002-1986", "PMG") in new stack

asterisk*CLI> 
    -- Executing SetVar("SIP/tsr002-1986", "CALLFILENAME=9008081793") in new stack

asterisk*CLI> 
    -- Executing SetVar("SIP/tsr002-1986", "CALLPATH=/var/calls") in new stack

asterisk*CLI> 
    -- Executing Monitor("SIP/tsr002-1986", "wav|/var/calls/9008081793|mb") in new stack

asterisk*CLI> 
    -- Executing Dial("SIP/tsr002-1986", "SIP/14807480034@cvcdial|999|r") in new stack

asterisk*CLI> 
We're at XXX.XXX.XXX.XXX(Asterisk Server) port 14492

asterisk*CLI> 
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP

asterisk*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP

asterisk*CLI> 
13 headers, 11 lines

asterisk*CLI> 
Reliably Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
INVITE sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;rport
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Jun 2006 16:44:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 5521 5521 IN IP4 XXX.XXX.XXX.XXX(Asterisk Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(Asterisk Server)
t=0 0
m=audio 14492 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

asterisk*CLI> 
    -- Called 14807480034@cvcdial

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Length: 0



asterisk*CLI> 

--- (10 headers 0 lines)---

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as09aba7a7
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 23728 23728 IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
t=0 0
m=audio 19026 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

asterisk*CLI> 
Destroying call '70CFB4BC-1DD2-11B2-9F5A-CB9D698C9C9E@192.168.0.109'

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK3a784e8f;received=XXX.XXX.XXX.XXX(Asterisk Server);rport=5060
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as09aba7a7
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Content-Type: application/sdp
Content-Length: 241


v=0
o=root 23728 23729 IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
s=session
c=IN IP4 XXX.XXX.XXX.XXX(VOIP Server)
t=0 0
m=audio 19026 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


--- (11 headers 11 lines)---

asterisk*CLI> 
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port XXX.XXX.XXX.XXX(VOIP Server):19026
Found description format PCMU

asterisk*CLI> 
Found description format GSM
Found description format telephone-event

asterisk*CLI> 
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>

asterisk*CLI> 
set_destination: Parsing <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)> for address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX(VOIP Server), port 5060

asterisk*CLI> 
Transmitting (no NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
ACK sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(Asterisk Server):5060;branch=z9hG4bK60abb468;rport
From: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
To: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as09aba7a7
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



---

asterisk*CLI> 
Scheduling destruction of call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102' in 15000 ms

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
OPTIONS sip:192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499778d02bc94950000002a
Content-Length: 0
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 13 OPTIONS
From: <sip:tsr002@192.168.0.10>;tag=4152033219174757187
Max-Forwards: 70
To: <sip:192.168.0.10:5060>



asterisk*CLI> 

--- (8 headers 0 lines)---

asterisk*CLI> 
Looking for s in default (domain 192.168.0.10)

asterisk*CLI> 
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a80066000000104499778d02bc94950000002a;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=4152033219174757187
To: <sip:192.168.0.10:5060>;tag=as26c4cbee
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 13 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0


---

asterisk*CLI> 
Destroying call '2d79d0d712fb28cf350e8b576cde3f69@XXX.XXX.XXX.XXX(VOIP Server)'

asterisk*CLI> 
Destroying call '05528224-8EA7-4FB0-8DFA-3886115CB2EC@192.168.100.102'

asterisk*CLI> 
<-- SIP read from XXX.XXX.XXX.XXX(VOIP Server):5060: 
BYE sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server) SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3fa5a0e2;rport
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as09aba7a7
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
Contact: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



asterisk*CLI> 

--- (10 headers 0 lines)---

asterisk*CLI> 
Sending to XXX.XXX.XXX.XXX(VOIP Server) : 5060 (NAT)

asterisk*CLI> 
Transmitting (NAT) to XXX.XXX.XXX.XXX(VOIP Server):5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX(VOIP Server):5060;branch=z9hG4bK3fa5a0e2;received=XXX.XXX.XXX.XXX(VOIP Server);rport=5060
From: <sip:14807480034@XXX.XXX.XXX.XXX(VOIP Server)>;tag=as09aba7a7
To: "PMG" <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>;tag=as57bbb343
Call-ID: 4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8778994498@XXX.XXX.XXX.XXX(Asterisk Server)>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---

asterisk*CLI> 
  == Spawn extension (rca, **90080817934807480034, 6) exited non-zero on 'SIP/tsr002-1986'

asterisk*CLI> 
set_destination: Parsing <sip:tsr002@192.168.0.102:5060> for address/port to send to

asterisk*CLI> 
set_destination: set destination to 192.168.0.102, port 5060

asterisk*CLI> 
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
BYE sip:tsr002@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK3f496560;rport
From: "fname_lname_10" <sip:asterisk@192.168.0.10>;tag=as2393f69f
To: <sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119
Contact: <sip:asterisk@192.168.0.10>
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK3f496560
Content-Length: 0
Call-ID: 48501c4957c98ac50fd4bf573052cff0@192.168.0.10
CSeq: 103 BYE
From: "fname_lname_10"<sip:asterisk@192.168.0.10>;tag=as2393f69f
To: "unknown"<sip:tsr002@192.168.0.102:5060>;tag=4152020943107930119



--- (7 headers 0 lines)---
Destroying call '4c41d794333fdac3750b768917fc8790@XXX.XXX.XXX.XXX(Asterisk Server)'
Destroying call '48501c4957c98ac50fd4bf573052cff0@192.168.0.10'

asterisk*CLI> 
Jun 21 09:45:19 NOTICE[7154]: pbx_spool.c:279 attempt_thread: Call completed to SIP/tsr002

asterisk*CLI> 
<-- SIP read from 192.168.0.102:5060: 
OPTIONS sip:192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977a12529baf50000002d
Content-Length: 0
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 14 OPTIONS
From: <sip:tsr002@192.168.0.10>;tag=41520532201796165376
Max-Forwards: 70
To: <sip:192.168.0.10:5060>



--- (8 headers 0 lines)---
Looking for s in default (domain 192.168.0.10)
Transmitting (no NAT) to 192.168.0.102:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.102;rport;branch=z9hG4bKc0a8006600000010449977a12529baf50000002d;received=192.168.0.102
From: <sip:tsr002@192.168.0.10>;tag=41520532201796165376
To: <sip:192.168.0.10:5060>;tag=as334b8b91
Call-ID: B9193A8A-1DD1-11B2-8D86-DF3FAA064735@192.168.0.102
CSeq: 14 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.0.10>
Accept: application/sdp
Content-Length: 0