Public feed #604031

Mine

public text v1 · immutable
Raw Edit
·published 2007-07-05 07:58 UTC
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<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9
CSeq: 3692 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=7C04A406
Call-ID: 1039122264@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;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 32848 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 1039122264@192.168.11.123

<--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=7C04A406
To: <sip:9214020@192.168.11.124>;tag=as3e16b601
Call-ID: 1039122264@192.168.11.123
CSeq: 3692 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26959360"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1039122264@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214059'

<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK2AB2B9A9
CSeq: 3692 ACK
To: <sip:9214020@192.168.11.124>;tag=as3e16b601
From: <sip:214059@192.168.11.124>;tag=7C04A406
Call-ID: 1039122264@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4
CSeq: 3693 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="214059", realm="asterisk", nonce="26959360", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="14699d4868b687d1d6c2cc55a81a8f61", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=7C04A406
Call-ID: 1039122264@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;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 32848 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 1039122264@192.168.11.123
Found user '214059'
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 3
Peer audio RTP is at port 192.168.11.123:32848
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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32848
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214059@192.168.11.123;transport=udp>

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=7C04A406
To: <sip:9214020@192.168.11.124>
Call-ID: 1039122264@192.168.11.123
CSeq: 3693 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0


<------------>
    -- Executing [9214020@local-phones:1] Dial("SIP/214059-0820c130", "SIP/9214020@192.168.11.120|20") in new stack
Audio is at 192.168.11.124 port 10086
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:9214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4b314b04
To: <sip:9214020@192.168.11.120>
Contact: <sip:214059@192.168.11.124>
Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 05 Jul 2007 07:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 4528 4528 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 10086 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 9214020@192.168.11.120

<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4b314b04
To: <sip:9214020@192.168.11.120>;tag=4C5F5BC5-BBC
Date: Thu, 05 Jul 2007 07:58:04 GMT
Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4b314b04
To: <sip:9214020@192.168.11.120>;tag=4C5F5BC5-BBC
Date: Thu, 05 Jul 2007 07:58:04 GMT
Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@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 192.168.11.120:5060:
ACK sip:9214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK71b2fd18;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as4b314b04
To: <sip:9214020@192.168.11.120>;tag=4C5F5BC5-BBC
Contact: <sip:214059@192.168.11.124>
Call-ID: 3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/192.168.11.120-08211040 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [9214020@local-phones:2] Congestion("SIP/214059-0820c130", "") in new stack

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=7C04A406
To: <sip:9214020@192.168.11.124>;tag=as4fb7fa69
Call-ID: 1039122264@192.168.11.123
CSeq: 3693 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1


<------------>
  == Spawn extension (local-phones, 9214020, 2) exited non-zero on 'SIP/214059-0820c130'

<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK4C589DF4
CSeq: 3693 ACK
To: <sip:9214020@192.168.11.124>;tag=as4fb7fa69
From: <sip:214059@192.168.11.124>;tag=7C04A406
Call-ID: 1039122264@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3a02b0ff44a1d55e42c7b5c45030e8d6@192.168.11.124' Method: INVITE
Really destroying SIP dialog '1039122264@192.168.11.123' Method: ACK