All pastes #732998 Raw Edit

Something

public text v1 · immutable
·published 2007-10-11 13:23 UTC
rendered paste body
ee*CLI> 
<--- SIP read from 192.168.11.123:5062 --->
INVITE sip:693977@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F
CSeq: 390 INVITE
To: <sip:693977@192.168.11.124>
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=2D4F9484
Call-ID: 1665297040@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123:5062;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32782 RTP/AVP 8 97 3 0
m=audio 32782 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<------------->
--- (11 headers 12 lines) ---
Sending to 192.168.11.123 : 5062 (no NAT)
Using INVITE request as basis request - 1665297040@192.168.11.123

<--- Reliably Transmitting (no NAT) to 192.168.11.123:5062 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=2D4F9484
To: <sip:693977@192.168.11.124>;tag=as77643dba
Call-ID: 1665297040@192.168.11.123
CSeq: 390 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d44f613"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1665297040@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214077'
bee*CLI> 
<--- SIP read from 192.168.11.123:5062 --->
ACK sip:693977@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK9A26C9F
CSeq: 390 ACK
To: <sip:693977@192.168.11.124>;tag=as77643dba
From: <sip:214077@192.168.11.124>;tag=2D4F9484
Call-ID: 1665297040@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123:5062;transport=udp>


<------------->
--- (9 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.123:5062 --->
INVITE sip:693977@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9
CSeq: 391 INVITE
To: <sip:693977@192.168.11.124>
Proxy-Authorization: Digest username="214077", realm="asterisk", nonce="3d44f613", uri="sip:693977@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="4c2041fc1e939228ce2a67958c9ff4a7", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214077@192.168.11.124>;tag=2D4F9484
Call-ID: 1665297040@192.168.11.123
Subject: sip:214077@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123:5062;transport=udp>

v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32782 RTP/AVP 8 97 3 0
m=audio 32782 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

<------------->
--- (12 headers 12 lines) ---
Sending to 192.168.11.123 : 5062 (no NAT)
Using INVITE request as basis request - 1665297040@192.168.11.123
Found user '214077'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 192.168.11.123:32782
Found description format PCMU for ID 0
Found description format GSM for ID 3
Found description format PCMA for ID 8
Found description format iLBC for ID 97
Capabilities: us - 0x8 (alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32782
Looking for 693977 in from-sip (domain 192.168.11.124)
list_route: hop: <sip:214077@192.168.11.123:5062;transport=udp>

<--- Transmitting (no NAT) to 192.168.11.123:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=2D4F9484
To: <sip:693977@192.168.11.124>
Call-ID: 1665297040@192.168.11.123
CSeq: 391 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:693977@192.168.11.124>
Content-Length: 0


<------------>
Audio is at 192.168.11.124 port 15138
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.128:5060:
INVITE sip:693977@192.168.11.128 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as39d21cfe
To: <sip:693977@192.168.11.128>
Contact: <sip:214077@192.168.11.124>
Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 11 Oct 2007 13:23:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 21640 21640 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 15138 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
bee*CLI> 
<--- SIP read from 192.168.11.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as39d21cfe
To: <sip:693977@192.168.11.128>;tag=246DD5-1A40
Date: Mon, 01 Mar 1993 00:39:47 GMT
Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
bee*CLI> 
<--- SIP read from 192.168.11.128:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as39d21cfe
To: <sip:693977@192.168.11.128>;tag=246DD5-1A40
Date: Mon, 01 Mar 1993 00:39:47 GMT
Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.11.128:5060:
ACK sip:693977@192.168.11.128 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK1549241c;rport
From: "214077" <sip:214077@192.168.11.124>;tag=as39d21cfe
To: <sip:693977@192.168.11.128>;tag=246DD5-1A40
Contact: <sip:214077@192.168.11.124>
Call-ID: 0f35ed3461aa30c61e108d723fb794d6@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
bee*CLI> 
<--- Transmitting (no NAT) to 192.168.11.123:5062 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9;received=192.168.11.123
From: <sip:214077@192.168.11.124>;tag=2D4F9484
To: <sip:693977@192.168.11.124>;tag=as0592e4e8
Call-ID: 1665297040@192.168.11.123
CSeq: 391 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:693977@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34


<------------>
bee*CLI> 
<--- SIP read from 192.168.11.123:5062 --->
ACK sip:693977@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123:5062;branch=z9hG4bK1C2782E9
CSeq: 391 ACK
To: <sip:693977@192.168.11.124>;tag=as0592e4e8
From: <sip:214077@192.168.11.124>;tag=2D4F9484
Call-ID: 1665297040@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214077@192.168.11.123:5062;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '0f35ed3461aa30c61e108d723fb794d6@192.168.11.124' Method: INVITE
Really destroying SIP dialog '1665297040@192.168.11.123' Method: ACK
bee*CLI>