All pastes #914094 Raw Edit

Someone

public text v1 · immutable
·published 2008-02-22 16:46 UTC
rendered paste body
asterisk1*CLI> sip debug peer providertrunk
SIP Debugging Enabled for IP: 209.183.xxx.xxx:5060
asterisk1*CLI>
<-- SIP read from 209.183.xxx.xxx:5060:
INVITE sip:479242xxxx@72.18.xxx.xxx:5060;transport=udp SIP/2.0
To: <sip:479242xxxx@72.18.xxx.xxx>
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 INVITE
Contact: "LASTNAME,FIRSTNAME"<sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060>
Expires: 180
Max-Forwards: 69
Content-Type: application/sdp
Organization:
Supported: 100rel
Allow-Events: message-summary, refer, dialog, line-seize, presence
Content-Length: 171

v=0
o=- 1107660232 1079319116 IN IP4 209.183.xxx.xxx
s=-
c=IN IP4 209.183.xxx.xxx
t=0 0
m=audio 35442 RTP/AVP 2 0 101
a=ptime:20
a=rtpmap:101 telephone-event/8000

--- (14 headers 8 lines) ---
Using INVITE request as basis request - 7513591450075b58a823083aa4abaf77
Sending to 209.183.xxx.xxx : 5060 (NAT)
Found peer 'providerTrunk'
Found RTP audio format 2
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 209.183.xxx.xxx:35442
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Looking for 479242xxxx in from-sip-external (domain 72.18.xxx.xxx)
list_route: hop: <sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060>
Transmitting (no NAT) to 209.183.xxx.xxx:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
To: <sip:479242xxxx@72.18.xxx.xxx>
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:479242xxxx@72.18.xxx.xxx>
Content-Length: 0


---
  dialparties.agi: Starting New Dialparties.agi
  dialparties.agi: Caller ID name is 'LASTNAME,FIRSTNAME' number is '479652xxxx'
  dialparties.agi: Methodology of ring is  'ringall'
Transmitting (no NAT) to 209.183.xxx.xxx:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
To: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:479242xxxx@72.18.xxx.xxx>
Content-Length: 0


---
  dialparties.agi: Starting New Dialparties.agi
  dialparties.agi: Caller ID name is 'LASTNAME,FIRSTNAME' number is '479652xxxx'
  dialparties.agi: Methodology of ring is  'ringall'
Transmitting (no NAT) to 209.183.xxx.xxx:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
To: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:479242xxxx@72.18.xxx.xxx>
Content-Length: 0


---
We're at 72.18.xxx.xxx port 16758
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-fddb482d9dc4ff50dbac-1-cHAwMTYwYjYzYWIzMTAwNTVhOGIzYg..-d87543-;received=209.183.xxx.xxx
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
To: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:479242xxxx@72.18.xxx.xxx>
Content-Type: application/sdp
Content-Length: 158

v=0
o=root 3039 3039 IN IP4 72.18.xxx.xxx
s=session
c=IN IP4 72.18.xxx.xxx
t=0 0
m=audio 16758 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
asterisk1*CLI>
<-- SIP read from 209.183.xxx.xxx:5060:
ACK sip:479242xxxx@72.18.xxx.xxx SIP/2.0
To: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
From: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-eaee5b198d7d897e-1-cHA0MmVkMWQxZWE4OWI2MDYyMTQ3Yg..-d87543-
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 369683301 ACK
Contact: "LASTNAME,FIRSTNAME"<sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060>
Max-Forwards: 69
Organization:
Allow-Events: message-summary, refer, dialog, line-seize, presence
Content-Length: 0


--- (11 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060:
OPTIONS sip:voip.myprovider.com SIP/2.0
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9;rport
From: "Unknown" <sip:Unknown@72.18.xxx.xxx>;tag=as548d1adb
To: <sip:voip.myprovider.com>
Contact: <sip:Unknown@72.18.xxx.xxx>
Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Feb 2008 16:41:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (no NAT) to 209.183.xxx.xxx:5060:
OPTIONS sip:voip.myprovider.com SIP/2.0
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9;rport
From: "Unknown" <sip:Unknown@72.18.xxx.xxx>;tag=as548d1adb
To: <sip:voip.myprovider.com>
Contact: <sip:Unknown@72.18.xxx.xxx>
Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Feb 2008 16:41:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 209.183.xxx.xxx:5060:
SIP/2.0 200 OK
To: <sip:voip.myprovider.com>;tag=metaswitch.providertelecom.net+1+0+6e023f86
From: "Unknown"<sip:Unknown@72.18.xxx.xxx>;tag=as548d1adb
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK5d4a93d9
Call-ID: 1c159e7356e093f413c6225415506193@72.18.xxx.xxx
CSeq: 102 OPTIONS
Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed
Accept-Encoding: identity
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER
Organization:
Server: DC-SIP/2.0
Supported: 100rel
Allow-Events: message-summary, refer, dialog, line-seize, presence
Content-Length: 0


--- (14 headers 0 lines) ---
Destroying call '1c159e7356e093f413c6225415506193@72.18.xxx.xxx'
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 209.183.xxx.xxx:5060:
REGISTER sip:voip.myprovider.com SIP/2.0
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a;rport
From: <sip:479242xxxx@voip.myprovider.com>;tag=as18f07382
To: <sip:479242xxxx@voip.myprovider.com>
Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1
CSeq: 116 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:s@72.18.xxx.xxx>
Event: registration
Content-Length: 0


---
Retransmitting #1 (NAT) to 209.183.xxx.xxx:5060:
REGISTER sip:voip.myprovider.com SIP/2.0
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a;rport
From: <sip:479242xxxx@voip.myprovider.com>;tag=as18f07382
To: <sip:479242xxxx@voip.myprovider.com>
Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1
CSeq: 116 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:s@72.18.xxx.xxx>
Event: registration
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 209.183.xxx.xxx:5060:
SIP/2.0 200 OK
To: <sip:479242xxxx@voip.myprovider.com>;tag=182a807b
From: <sip:479242xxxx@voip.myprovider.com>;tag=as18f07382
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK2bc2c28a
Call-ID: 52cc5f9158fdf085350ac5b341b833f9@127.0.0.1
CSeq: 116 REGISTER
Contact: <sip:s@209.183.xxx.xxx:5060>;expires=30
Content-Length: 0


--- (8 headers 0 lines) ---
Scheduling destruction of call '52cc5f9158fdf085350ac5b341b833f9@127.0.0.1' in 32000 ms
Scheduling destruction of call '7513591450075b58a823083aa4abaf77' in 32000 ms
set_destination: Parsing <sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060> for address/port to send to
set_destination: set destination to 209.183.xxx.xxx, port 5060
Reliably Transmitting (no NAT) to 209.183.xxx.xxx:5060:
BYE sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK76008b77;rport
From: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
To: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 209.183.xxx.xxx:5060:
SIP/2.0 200 OK
To: "LASTNAME,FIRSTNAME"<sip:479652xxxx@192.168.xxx.xxx:5060;transport=udp>;tag=c9cdc102
From: <sip:479242xxxx@72.18.xxx.xxx>;tag=as55f1d89b
Via: SIP/2.0/UDP 72.18.xxx.xxx:5060;branch=z9hG4bK76008b77
Call-ID: 7513591450075b58a823083aa4abaf77
CSeq: 102 BYE
Contact: "LASTNAME,FIRSTNAME"<sip:7aJA2wDjHwat39SzpfB2OApBBg7IEJLp0roXJgh8w@209.183.xxx.xxx:5060>
Organization:
Server: DC-SIP/2.0
Supported: 100rel
Allow-Events: message-summary, refer, dialog, line-seize, presence
Content-Length: 0


--- (12 headers 0 lines) ---
Destroying call '7513591450075b58a823083aa4abaf77'
asterisk1*CLI> sip no debug