All pastes #594006 Raw Edit

Miscellany

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#594006 ·published 2007-06-28 17:45 UTC
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User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 28 Jun 2007 17:44:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 4297 4297 IN IP4 91.103.25.29
s=session
c=IN IP4 91.103.25.29
t=0 0
m=audio 12670 RTP/AVP 4 18 3 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Transmitting (no NAT) to 80.93.57.101:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101
From: <sip:5646414@80.93.57.101>;tag=ffffffff1d1a3c1
To: <sip:577266@91.103.25.29>;tag=as6bbe00cb
Call-ID: ffffffff1d1a3c1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:577266@91.103.25.29>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 91.103.25.26:5060:
                                     SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---

<-- SIP read from 91.103.25.26:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 91.103.25.26:5060:
                                     SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---

<-- SIP read from 91.103.25.26:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
asterisk*CLI>
<-- SIP read from 91.103.25.26:5060:
                                     SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Contact: <sip:22@192.168.0.222:5060>
Content-Type: application/sdp
Content-Length: 130

v=0
o=- 8829 8829 IN IP4 192.168.0.222
s=session
c=IN IP4 192.168.0.222
t=0 0
m=audio 10006 RTP/AVP 4
a=rtpmap:4 G723/8000

--- (9 headers 7 lines) ---
Found RTP audio format 4
Peer audio RTP is at port 192.168.0.222:10006
Found description format G723
Capabilities: us - 0x80103 (g723|gsm|g729|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:22@192.168.0.222:5060>
set_destination: Parsing <sip:22@192.168.0.222:5060> for address/port to send to
set_destination: set destination to 192.168.0.222, port 5060
Transmitting (NAT) to 91.103.25.26:5060:
                                        ACK sip:22@192.168.0.222:5060 SIP/2.0
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK012ab263;rport
From: "NOC" <sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Contact: <sip:5646414@91.103.25.29>
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 91.103.25.29 port 19006
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 80.93.57.101:5060:
                                                    SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101
From: <sip:5646414@80.93.57.101>;tag=ffffffff1d1a3c1
To: <sip:577266@91.103.25.29>;tag=as6bbe00cb
Call-ID: ffffffff1d1a3c1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:577266@91.103.25.29>
Content-Type: application/sdp
Content-Length: 211

v=0
o=root 4297 4297 IN IP4 91.103.25.29
s=session
c=IN IP4 91.103.25.29
t=0 0
m=audio 19006 RTP/AVP 4 96
a=rtpmap:4 G723/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 91.103.25.26:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b
From: "NOC"<sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 INVITE
Contact: <sip:22@192.168.0.222:5060>
Content-Type: application/sdp
Content-Length: 130

v=0
o=- 8829 8829 IN IP4 192.168.0.222
s=session
c=IN IP4 192.168.0.222
t=0 0
m=audio 10006 RTP/AVP 4
a=rtpmap:4 G723/8000

--- (9 headers 7 lines) ---
Found RTP audio format 4
Peer audio RTP is at port 192.168.0.222:10006
Found description format G723
Capabilities: us - 0x80103 (g723|gsm|g729|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:22@192.168.0.222:5060>
set_destination: Parsing <sip:22@192.168.0.222:5060> for address/port to send to
set_destination: set destination to 192.168.0.222, port 5060
Transmitting (NAT) to 91.103.25.26:5060:
ACK sip:22@192.168.0.222:5060 SIP/2.0
Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK012ab263;rport
From: "NOC" <sip:5646414@91.103.25.29>;tag=as4c006281
To: <sip:22@192.168.0.222:5060>;tag=31c3ef99
Contact: <sip:5646414@91.103.25.29>
Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 91.103.25.29 port 19006
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 80.93.57.101:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101
From: <sip:5646414@80.93.57.101>;tag=ffffffff1d1a3c1
To: <sip:577266@91.103.25.29>;tag=as6bbe00cb
Call-ID: ffffffff1d1a3c1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:577266@91.103.25.29>
Content-Type: application/sdp
Content-Length: 211

v=0
o=root 4297 4297 IN IP4 91.103.25.29
s=session
c=IN IP4 91.103.25.29
t=0 0
m=audio 19006 RTP/AVP 4 96
a=rtpmap:4 G723/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -

---
asterisk*CLI>
<-- SIP read from 80.93.57.101:5060:
                                     ACK sip:577266@91.103.25.29 SIP/2.0
Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091
From: <sip:5646414@80.93.57.101>;tag=ffffffff1d1a3c1
To: <sip:577266@91.103.25.29>;tag=as6bbe00cb
Call-ID: ffffffff1d1a3c1
Contact: <sip:5646414@80.93.57.101>
CSeq: 1 ACK
Accept: application/sdp
Max-Forwards: 70
User-Agent: IXC SS
Content-Length: 0


--- (11 headers 0 lines) ---

<-- SIP read from 80.93.57.101:5060:
ACK sip:577266@91.103.25.29 SIP/2.0
Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091
From: <sip:5646414@80.93.57.101>;tag=ffffffff1d1a3c1
To: <sip:577266@91.103.25.29>;tag=as6bbe00cb
Call-ID: ffffffff1d1a3c1
Contact: <sip:5646414@80.93.57.101>
CSeq: 1 ACK
Accept: application/sdp
Max-Forwards: 70
User-Agent: IXC SS
Content-Length: 0