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Connected to Asterisk 1.2.9.1 svn rev 34876 currently running on asterisk1 (pid= 31908)
Verbosity is at least 1
Core debug is at least 1
asterisk1*CLI> sip debug ip 192.168.253.3
SIP Debugging Enabled for IP: 192.168.253.3
asterisk1*CLI>
<-- SIP read from 192.168.253.3:5060:
INVITE sip:useip@192.168.253.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bK2f63d20524e118d0646a6a0bf72f3c5c
From: "07766087677" <sip:1001@192.168.253.3:5060>;tag=5563b413
To: <sip:useip@192.168.253.15>
Contact: <sip:1001@192.168.253.3:5060>
Call-ID: 524f4cf527540e4052821b212ed1b90e@192.168.253.3
CSeq: 801 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
User-Agent: CMI CM5K
Content-Length: 329
v=0
o=- 65072 0 IN IP4 192.168.253.3
s=SIP CALL
c=IN IP4 192.168.253.3
t=0 0
m=audio 60000 RTP/AVP 0 8 18 4 23 22 2 21
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=sendrecv
--- (12 headers 15 lines)---
Using INVITE request as basis request - 524f4cf527540e4052821b212ed1b90e@192.168.253.3
Sending to 192.168.253.3 : 5060 (NAT)
Found peer '103'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 23
Found RTP audio format 22
Found RTP audio format 2
Found RTP audio format 21
Peer audio RTP is at port 192.168.253.3:60000
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G723
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format G726-40
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for useip in from-trunk (domain 192.168.253.15)
Reliably Transmitting (no NAT) to 192.168.253.3:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bK2f63d20524e118d0646a6a0b
f72f3c5c;received=192.168.253.3
From: "07766087677" <sip:1001@192.168.253.3:5060>;tag=5563b413
To: <sip:useip@192.168.253.15>;tag=as5f6c5340
Call-ID: 524f4cf527540e4052821b212ed1b90e@192.168.253.3
CSeq: 801 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:useip@192.168.253.15>
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.253.3:5060:
ACK sip:useip@192.168.253.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.3:5060;branch=z9hG4bK2f63d20524e118d0646a6a0bf72f3c5c
From: "07766087677" <sip:1001@192.168.253.3:5060>;tag=5563b413
To: <sip:useip@192.168.253.15>;tag=as5f6c5340
Call-ID: 524f4cf527540e4052821b212ed1b90e@192.168.253.3
CSeq: 801 ACK
Content-Length: 0
--- (7 headers 0 lines)---
Destroying call '524f4cf527540e4052821b212ed1b90e@192.168.253.3'asterisk1*CLI>
<-- SIP read from 192.168.253.3:5060:
REGISTER sip:192.168.253.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bKf7471f35409a49fd0b00bde8e451b684
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>
Contact: <sip:103@192.168.253.3:5060>
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 4 REGISTER
Expires: 300
Authorization: Digest username="103",realm="asterisk",nonce="57bbbd11",response="058e3ba0017350a5761dbd5d265a4ade",uri="sip:192.168.253.15"
User-Agent: CMI CM5K
Content-Length: 0
--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.253.3 : 5060 (NAT)
Transmitting (no NAT) to 192.168.253.3:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bKf7471f35409a49fd0b00bde8e451b684;received=192.168.253.3
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:103@192.168.253.15>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.253.3:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bKf7471f35409a49fd0b00bde8e451b684;received=192.168.253.3
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>;tag=as34161848
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:103@192.168.253.15>
WWW-Authenticate: Digest realm="asterisk", nonce="7de054f9"
Content-Length: 0
---
Scheduling destruction of call '6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3'in 15000 ms
asterisk1*CLI>
<-- SIP read from 192.168.253.3:5060:
REGISTER sip:192.168.253.15 SIP/2.0
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bK99d83a8281d70d9efb4e9cc5bcfdd03f
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>
Contact: <sip:103@192.168.253.3:5060>
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 5 REGISTER
Expires: 300
Authorization: Digest username="103",realm="asterisk",nonce="7de054f9",response="78639875b78ef5c29bfac91e08226b6e",uri="sip:192.168.253.15"
User-Agent: CMI CM5K
Content-Length: 0
--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.253.3 : 5060 (NAT)
Transmitting (no NAT) to 192.168.253.3:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bK99d83a8281d70d9efb4e9cc5bcfdd03f;received=192.168.253.3
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:103@192.168.253.15>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.253.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.253.3:5060;rport;branch=z9hG4bK99d83a8281d70d9efb4e9cc5bcfdd03f;received=192.168.253.3
From: <sip:103@192.168.253.15>;tag=2fb0a274
To: <sip:103@192.168.253.15>;tag=as34161848
Call-ID: 6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 300
Contact: <sip:103@192.168.253.3:5060>;expires=300
Date: Fri, 21 Jul 2006 14:21:21 GMT
Content-Length: 0
---
Scheduling destruction of call '6e6c88d50184fabe2b633ae169a80a2f@192.168.253.3'in 15000 ms
asterisk1*CLI> sip no debug
SIP Debugging Disabled
asterisk1*CLI>