All pastes #21672 Raw Edit

Stuff

public text v1 · immutable
#21672 ·published 2005-08-31 00:07 UTC
rendered paste body

--- (12 headers 20 lines)---
Using INVITE request as basis request - 0bcdee0973f7618803ee258d5426da3e@66.250.69.11
Sending to 66.250.69.11 : 5060 (non-NAT)
Found peer 'switch2'
Found RTP audio format 10
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 66.250.69.11:16622
Found description format L16
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8008831849 in incoming
Reliably Transmitting (no NAT) to 66.250.69.11:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK65fc5a1c
From: "7542351245" <sip:7542351245@66.250.69.11>;tag=as738fe796
To: <sip:8008831849@64.135.90.5>;tag=as2c0b43bb
Call-ID: 0bcdee0973f7618803ee258d5426da3e@66.250.69.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:8008831849@64.135.90.5>
Content-Length: 0


---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as4c3f77ea
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as413cad04
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
taiko*CLI>
<-- SIP read from 66.250.69.11:5060:
ACK sip:8008831849@64.135.90.5 SIP/2.0
Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK65fc5a1c;rport
From: "7542351245" <sip:7542351245@66.250.69.11>;tag=as738fe796
To: <sip:8008831849@64.135.90.5>;tag=as2c0b43bb
Contact: <sip:7542351245@66.250.69.11>
Call-ID: 0bcdee0973f7618803ee258d5426da3e@66.250.69.11
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '0bcdee0973f7618803ee258d5426da3e@66.250.69.11'
Retransmitting #1 (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as4c3f77ea
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
Retransmitting #1 (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as413cad04
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 64.135.11.85:5060:
OPTIONS sip:64.135.11.85 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK3d0b3fdb
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as284686f6
To: <sip:64.135.11.85>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 6cd4126a3f3c7f333921997036644be8@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
taiko*CLI>
<-- SIP read from 64.135.11.85:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK3d0b3fdb
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as284686f6
To: <sip:64.135.11.85>;tag=as7a9798d8
Call-ID: 6cd4126a3f3c7f333921997036644be8@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: InnerVoIP-GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:64.135.11.85>
Accept: application/sdp
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '6cd4126a3f3c7f333921997036644be8@64.135.90.5'
11 headers, 0 lines
Reliably Transmitting (no NAT) to 64.135.11.85:5060:
OPTIONS sip:64.135.11.85 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK23f0da7b
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as5ca2a7c7
To: <sip:64.135.11.85>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 35adee4b59dd6c91275b994900765796@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
taiko*CLI>
<-- SIP read from 64.135.11.85:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK23f0da7b
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as5ca2a7c7
To: <sip:64.135.11.85>;tag=as29a90f91
Call-ID: 35adee4b59dd6c91275b994900765796@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: InnerVoIP-GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:64.135.11.85>
Accept: application/sdp
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '35adee4b59dd6c91275b994900765796@64.135.90.5'
Retransmitting #2 (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK778a4220
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as4c3f77ea
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 7d04ad223bcf19c3128becea003617a8@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
Destroying call '7d04ad223bcf19c3128becea003617a8@64.135.90.5'
Retransmitting #2 (no NAT) to 64.135.20.204:5060:
OPTIONS sip:64.135.20.204 SIP/2.0
Via: SIP/2.0/UDP 64.135.90.5:5060;branch=z9hG4bK6076e601
From: "asterisk" <sip:asterisk@64.135.90.5>;tag=as413cad04
To: <sip:64.135.20.204>
Contact: <sip:asterisk@64.135.90.5>
Call-ID: 42af110744577fb66d3b43ab1bb42bd4@64.135.90.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 31 Aug 2005 00:06:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0


---
Destroying call '42af110744577fb66d3b43ab1bb42bd4@64.135.90.5'
taiko*CLI>
<-- SIP read from 64.135.113.4:5060:

--- (0 headers 0 lines) Nat keepalive ---
taiko*CLI>