All pastes #74022 Raw Edit

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#74022 ·published 2006-06-28 15:42 UTC
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Reliably Transmitting (no NAT) to 192.168.0.254:5060:
INVITE sip:16562226071845@192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
From: "1656222" <sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>
Contact: <sip:1656222@192.168.0.1>
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Jun 2006 15:37:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 3131 3131 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 19030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #1 (no NAT) to 192.168.0.254:5060:
INVITE sip:16562226071845@192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
From: "1656222" <sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>
Contact: <sip:1656222@192.168.0.1>
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Jun 2006 15:37:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 3131 3131 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 19030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 100 Trying
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 INVITE
From: "1656222"<sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>;tag=c0a800fe-ebb
User-Agent: Quintum/1.0.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
Quintum: 0b023335


--- (8 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 100 Trying
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 INVITE
From: "1656222"<sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>;tag=c0a800fe-ebb
User-Agent: Quintum/1.0.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
Quintum: 0b023335


--- (8 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 183 Session Progress
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
Content-Length: 160
Content-Type: application/sdp
CSeq: 102 INVITE
From: "1656222"<sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>;tag=c0a800fe-ebb
User-Agent: Quintum/1.0.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
Quintum: 070e00000003008f6506001e03808081

v=0
o=Quintum 4 3131 IN IP4 192.168.0.254
s=VoipCall
c=IN IP4 192.168.0.254
t=0 0
m=audio 10240 RTP/AVP 0
c=IN IP4 192.168.0.254
a=rtpmap:0 pcmu/8000/1

--- (10 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 192.168.0.254:10240
Found description format pcmu
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 180 Ringing
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
Content-Length: 160
Content-Type: application/sdp
CSeq: 102 INVITE
From: "1656222"<sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>;tag=c0a800fe-ebb
User-Agent: Quintum/1.0.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport

v=0
o=Quintum 5 3131 IN IP4 192.168.0.254
s=VoipCall
c=IN IP4 192.168.0.254
t=0 0
m=audio 10240 RTP/AVP 0
c=IN IP4 192.168.0.254
a=rtpmap:0 pcmu/8000/1

--- (9 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 192.168.0.254:10240
Found description format pcmu
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Reliably Transmitting (no NAT) to 192.168.0.254:5060:
CANCEL sip:16562226071845@192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport
From: "1656222" <sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>
Contact: <sip:1656222@192.168.0.1>
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 200 OK
Call-ID: 6816d1ce4fa1f89624fe4f6d3a805830@192.168.0.1
CSeq: 102 CANCEL
From: "1656222"<sip:1656222@192.168.0.1>;tag=as7369da55
To: <sip:16562226071845@192.168.0.254>
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK31d5c8c5;rport


--- (6 headers 0 lines)---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.254:5060:
OPTIONS sip:192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ce6aaf6;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as32b17854
To: <sip:192.168.0.254>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 0725474745878aad152aaf22033d9997@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Jun 2006 15:38:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 192.168.0.254:5060:
SIP/2.0 200 OK
Call-ID: 0725474745878aad152aaf22033d9997@192.168.0.1
CSeq: 102 OPTIONS
From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as32b17854
To: <sip:192.168.0.254>
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ce6aaf6;rport