rendered paste bodygarantia-proxy*CLI> sip debug ip 192.168.6.150
SIP Debugging Enabled for IP: 192.168.6.150
-- Executing Dial("SIP/1002-081de6b8", "SIP/2000|60|t") in new stack
We're at 10.0.6.2 port 16636
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
13 headers, 9 lines
Reliably Transmitting (no NAT) to 192.168.6.150:5060:
INVITE sip:2000@192.168.6.150:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.2:5060;branch=z9hG4bK45e9acb3;rport
From: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
To: <sip:2000@192.168.6.150:5060>
Contact: <sip:1002@10.0.6.2>
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 11 Dec 2007 12:09:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 174
v=0
o=root 4144 4144 IN IP4 10.0.6.2
s=session
c=IN IP4 10.0.6.2
t=0 0
m=audio 16636 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
---
-- Called 2000
garantia-proxy*CLI>
<-- SIP read from 192.168.6.150:5060:
SIP/2.0 100 Trying
To: <sip:2000@192.168.6.150:5060>
From: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.6.2:5060;branch=z9hG4bK45e9acb3
Server: Linksys/PAP2-3.1.8(LS)
Content-Length: 0
--- (8 headers 0 lines) ---
garantia-proxy*CLI>
<-- SIP read from 192.168.6.150:5060:
SIP/2.0 180 Ringing
To: <sip:2000@192.168.6.150:5060>;tag=657faf1e1104c709i1
From: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.6.2:5060;branch=z9hG4bK45e9acb3
Server: Linksys/PAP2-3.1.8(LS)
Content-Length: 0
--- (8 headers 0 lines) ---
-- SIP/2000-081e70c0 is ringing
garantia-proxy*CLI>
<-- SIP read from 192.168.6.150:5060:
SIP/2.0 200 OK
To: <sip:2000@192.168.6.150:5060>;tag=657faf1e1104c709i1
From: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.6.2:5060;branch=z9hG4bK45e9acb3
Contact: 2000 <sip:2000@192.168.6.150:5060>
Server: Linksys/PAP2-3.1.8(LS)
Content-Length: 239
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 33007888 33007888 IN IP4 192.168.6.150
s=-
c=IN IP4 192.168.6.150
t=0 0
m=audio 16456 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.6.150:16456
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
list_route: hop: <sip:2000@192.168.6.150:5060>
set_destination: Parsing <sip:2000@192.168.6.150:5060> for address/port to send to
set_destination: set destination to 192.168.6.150, port 5060
Transmitting (no NAT) to 192.168.6.150:5060:
ACK sip:2000@192.168.6.150:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.2:5060;branch=z9hG4bK4d01f4ba;rport
From: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
To: <sip:2000@192.168.6.150:5060>;tag=657faf1e1104c709i1
Contact: <sip:1002@10.0.6.2>
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/2000-081e70c0 answered SIP/1002-081de6b8
garantia-proxy*CLI>
<-- SIP read from 192.168.6.150:5060:
BYE sip:1002@10.0.6.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.150:5060;branch=z9hG4bK-39d5fa05
From: <sip:2000@192.168.6.150:5060>;tag=657faf1e1104c709i1
To: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/PAP2-3.1.8(LS)
Content-Length: 0
--- (9 headers 0 lines) ---
Sending to 192.168.6.150 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.6.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.150:5060;branch=z9hG4bK-39d5fa05;received=192.168.6.150
From: <sip:2000@192.168.6.150:5060>;tag=657faf1e1104c709i1
To: "1002" <sip:1002@10.0.6.2>;tag=as21c39d52
Call-ID: 75e3becf3e0344b90659ee8f1533c195@10.0.6.2
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1002@10.0.6.2>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
== Spawn extension (tronco-vono, 2000, 1) exited non-zero on 'SIP/1002-081de6b8'
-- Executing Hangup("SIP/1002-081de6b8", "") in new stack
== Spawn extension (tronco-vono, h, 1) exited non-zero on 'SIP/1002-081de6b8'
Destroying call '75e3becf3e0344b90659ee8f1533c195@10.0.6.2'
garantia-proxy*CLI>
<-- SIP read from 192.168.6.150:5060:
NOTIFY sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.150:5060;branch=z9hG4bK-c26923e7
From: 2001 <sip:2001@192.168.1.1>;tag=a5df18273495213fo0
To: <sip:192.168.1.1>
Call-ID: 2fff55a7-e9f601bf@192.168.6.150
CSeq: 21712 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/PAP2-3.1.8(LS)
Content-Length: 0
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.6.150:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 192.168.6.150:5060;branch=z9hG4bK-c26923e7;received=192.168.6.150
From: 2001 <sip:2001@192.168.1.1>;tag=a5df18273495213fo0
To: <sip:192.168.1.1>;tag=as4423624e
Call-ID: 2fff55a7-e9f601bf@192.168.6.150
CSeq: 21712 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---