All pastes #194687 Raw Edit

Something

public text v1 · immutable
·published 2006-10-08 04:52 UTC
rendered paste body
asterisk*CLI> sip debug
SIP Debugging enabled
Asterisk Ready.

<-- SIP read from 192.168.1.3:5060:
INVITE sip:14@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1
Content-Length: 335
Contact: <sip:ivan@192.168.1.3:5060>
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
Content-Type: application/sdp
CSeq: 1 INVITE
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
Max-Forwards: 70
To: <sip:14@192.168.1.2>
User-Agent: SJphone/1.60.289a (SJ Labs)

v=0
o=- 3369271910 3369271910 IN IP4 192.168.1.3
s=SJphone
c=IN IP4 192.168.1.3
t=0 0
a=direction:active
m=audio 49158 RTP/AVP 3 97 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (11 headers 15 lines)---
Using INVITE request as basis request - 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
Sending to 192.168.1.3 : 5060 (NAT)
Reliably Transmitting (no NAT) to 192.168.1.3:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1;received=192.168.1.3
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
To: <sip:14@192.168.1.2>;tag=as2f733647
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14@192.168.1.2>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6253989a"
Content-Length: 0


---
Scheduling destruction of call '0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3' in 15000 ms
Found user 'ivan'
asterisk*CLI>
<-- SIP read from 192.168.1.3:5060:
ACK sip:14@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e60000464c000000e1
Content-Length: 0
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
CSeq: 1 ACK
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
Max-Forwards: 70
To: <sip:14@192.168.1.2>;tag=as2f733647
User-Agent: SJphone/1.60.289a (SJ Labs)


--- (9 headers 0 lines)---
asterisk*CLI>
<-- SIP read from 192.168.1.3:5060:
INVITE sip:14@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2
Content-Length: 335
Contact: <sip:ivan@192.168.1.3:5060>
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
Content-Type: application/sdp
CSeq: 2 INVITE
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
Max-Forwards: 70
To: <sip:14@192.168.1.2>
User-Agent: SJphone/1.60.289a (SJ Labs)
Proxy-Authorization: Digest username="ivan",realm="asterisk",nonce="6253989a",uri="sip:14@192.168.1.2",response="d24aa324b832cabec5c14996cd7b55a5",algorithm="MD5"

v=0
o=- 3369271910 3369271910 IN IP4 192.168.1.3
s=SJphone
c=IN IP4 192.168.1.3
t=0 0
a=direction:active
m=audio 49158 RTP/AVP 3 97 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (12 headers 15 lines)---
Using INVITE request as basis request - 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
Sending to 192.168.1.3 : 5060 (NAT)
Found user 'ivan'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.3:49158
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format PCMA
Found description format PCMU
Found description format telephone-event
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: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 14 in outbound-local (domain 192.168.1.2)
Reliably Transmitting (no NAT) to 192.168.1.3:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2;received=192.168.1.3
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
To: <sip:14@192.168.1.2>;tag=as2f733647
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14@192.168.1.2>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 192.168.1.3:5060:
ACK sip:14@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000054452883e700004005000000e2
Content-Length: 0
Call-ID: 0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3
CSeq: 2 ACK
From: "Pablo"<sip:ivan@192.168.1.2>;tag=16954548418643
Max-Forwards: 70
To: <sip:14@192.168.1.2>;tag=as2f733647
User-Agent: SJphone/1.60.289a (SJ Labs)


--- (9 headers 0 lines)---
Destroying call '0A922184-949F-40E2-B98D-E6BB72BB2AD4@192.168.1.3'
asterisk*CLI>
<-- SIP read from 192.168.1.3:5060:
OPTIONS sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000010452883f100007f0c000000e4
Content-Length: 0
Call-ID: 4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3
CSeq: 56 OPTIONS
From: <sip:ivan@192.168.1.2>;tag=16955590619931
Max-Forwards: 70
To: <sip:192.168.1.2>


--- (8 headers 0 lines)---
Looking for s in default (domain 192.168.1.2)
Transmitting (no NAT) to 192.168.1.3:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.3;rport;branch=z9hG4bKc0a8010300000010452883f100007f0c000000e4;received=192.168.1.3
From: <sip:ivan@192.168.1.2>;tag=16955590619931
To: <sip:192.168.1.2>;tag=as4932bcc4
Call-ID: 4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3
CSeq: 56 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.2>
Accept: application/sdp
Content-Length: 0


---
Destroying call '4320915F-9E26-4BD7-ACBB-6173AA90A5B4@192.168.1.3'
asterisk*CLI> sip no debug
SIP Debugging Disabled
asterisk*CLI>