rendered paste body? ? ???????? ?????? ?? ???????? ??????, ??? ???? ??? ???????? ? ????????
<-- SIP read from 192.168.1.43:5060:
INVITE sip:74957757566@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKncdolnst
Max-Forwards: 70
To: <sip:74957757566@192.168.1.6>
From: "201" <sip:201@192.168.1.6>;tag=isiyh
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 962 INVITE
Contact: <sip:201@192.168.1.43>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304
v=0
o=201 1770785376 570897782 IN IP4 192.168.1.43
s=-
c=IN IP4 192.168.1.43
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
--- (13 headers 14 lines) ---
Using INVITE request as basis request - cjlqmdbgmjanzth@192.168.1.43
Sending to 192.168.1.43 : 5060 (NAT)
Reliably Transmitting (no NAT) to 192.168.1.43:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKncdolnst;received=192.168.1.43;rport=5060
From: "201" <sip:201@192.168.1.6>;tag=isiyh
To: <sip:74957757566@192.168.1.6>;tag=as197c8170
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 962 INVITE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38433a2b"
Content-Length: 0
---
Scheduling destruction of call 'cjlqmdbgmjanzth@192.168.1.43' in 15000 ms
Found user '201'
<-- SIP read from 192.168.1.43:5060:
ACK sip:74957757566@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKncdolnst
Max-Forwards: 70
To: <sip:74957757566@192.168.1.6>;tag=as197c8170
From: "201" <sip:201@192.168.1.6>;tag=isiyh
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 962 ACK
User-Agent: Twinkle/1.1
Content-Length: 0
--- (9 headers 0 lines) ---
<-- SIP read from 192.168.1.43:5060:
INVITE sip:74957757566@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKsyufaoni
Max-Forwards: 70
Proxy-Authorization: Digest username="201",realm="asterisk",nonce="38433a2b",uri="sip:74957757566@192.168.1.6",response="a5f436e27fd3e6e3fb7044eb0fc485c6",algorithm=MD5
To: <sip:74957757566@192.168.1.6>
From: "201" <sip:201@192.168.1.6>;tag=isiyh
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 INVITE
Contact: <sip:201@192.168.1.43>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 304
v=0
o=201 1770785376 570897782 IN IP4 192.168.1.43
s=-
c=IN IP4 192.168.1.43
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
--- (14 headers 14 lines) ---
Using INVITE request as basis request - cjlqmdbgmjanzth@192.168.1.43
Sending to 192.168.1.43 : 5060 (NAT)
Found user '201'
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.43:8000
Peer video RTP is at port 192.168.1.43:65535
Found description format speex
Found description format speex
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/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 74957757566 in office (domain 192.168.1.6)
list_route: hop: <sip:201@192.168.1.43>
Transmitting (no NAT) to 192.168.1.43:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060
From: "201" <sip:201@192.168.1.6>;tag=isiyh
To: <sip:74957757566@192.168.1.6>
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 INVITE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:74957757566@192.168.1.6>
Content-Length: 0
---
We're at 212.122.x.x port 12684
Video is at 212.122.x.x port 10784
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:74957757566@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>
Contact: <sip:user@212.122.x.x>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 102 INVITE
User-Agent: Usr
Max-Forwards: 70
Date: Mon, 14 Apr 2008 13:50:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 18815 18815 IN IP4 212.122.x.x
s=session
c=IN IP4 212.122.x.x
t=0 0
m=audio 12684 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport=5060
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 102 INVITE
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (8 headers 0 lines) ---
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport=5060
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=9A4C65AB
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="etc.tario.ru",nonce="1E6BFA84146B441D1C62",opaque="opaqueData",qop="auth",algorithm=MD5
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:74957757566@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK7aadd3b0;rport
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=9A4C65AB
Contact: <sip:user@212.122.x.x>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 102 ACK
User-Agent: Usr
Max-Forwards: 70
Content-Length: 0
---
We're at 212.122.x.x port 12684
Video is at 212.122.x.x port 10784
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
NVITE sip:74957757566@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK774b3c23;rport
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>
Contact: <sip:user@212.122.x.x>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 INVITE
User-Agent: Usr
Max-Forwards: 70
Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:74957757566@sipnet.ru", nonce="1E6BFA84146B441D1C62", response="509c3ba604e4c4a5076900a151c3b753", opaque="opaqueData", qop=auth, cnonce="67a0e2f6", nc=00000001
Date: Mon, 14 Apr 2008 13:50:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 18815 18816 IN IP4 212.122.x.x
s=session
c=IN IP4 212.122.x.x
t=0 0
m=audio 12684 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK774b3c23;rport=5060
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (8 headers 0 lines) ---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:312653-212.53.40.71.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.71:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 INVITE
Server: TarioSoftswitch/3.2.10
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.43:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060
From: "201" <sip:201@192.168.1.6>;tag=isiyh
To: <sip:74957757566@192.168.1.6>;tag=as589b0e88
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 INVITE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:74957757566@192.168.1.6>
Content-Length: 0
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:312653-212.53.40.71.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.71:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 INVITE
Content-Type: application/sdp
Server: TarioSoftswitch/3.2.10
Content-Length: 178
v=0
o=Tario-Softswitch 7115 101 IN IP4 212.53.40.82
s=SIP Call
c=IN IP4 212.53.40.82
t=0 0
m=audio 21588 RTP/AVP 8
c=IN IP4 212.53.40.82
a=rtpmap:8 PCMA/8000
a=ptime:20
--- (12 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.82:21588
Peer video RTP is at port 212.53.40.82:65535
Found description format PCMA
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
We're at 192.168.1.6 port 16014
Video is at 192.168.1.6 port 18442
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.1.43:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060
From: "201" <sip:201@192.168.1.6>;tag=isiyh
To: <sip:74957757566@192.168.1.6>;tag=as589b0e88
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 INVITE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:74957757566@192.168.1.6>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 18815 18815 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 16014 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK774b3c23
Record-Route: <sip:212.53.35.244:5060;lr>,<sip:312653-212.53.40.71.dialog.cgatepro;lr>
Record-Route: <sip:212.53.40.71:5060;lr>
Record-Route: <sip:212.53.40.40:5060;lr>
From: <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 INVITE
Contact: <sip:proc-136174@212.53.35.244>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS
Server: TarioSoftswitch/3.2.10
Content-Length: 178
v=0
o=Tario-Softswitch 7115 101 IN IP4 212.53.40.82
s=SIP Call
c=IN IP4 212.53.40.82
t=0 0
m=audio 21588 RTP/AVP 8
c=IN IP4 212.53.40.82
a=rtpmap:8 PCMA/8000
a=ptime:20
--- (14 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 212.53.40.82:21588
Peer video RTP is at port 212.53.40.82:65535
Found description format PCMA
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:212.53.40.40:5060;lr>
list_route: hop: <sip:212.53.40.71:5060;lr>
list_route: hop: <sip:312653-212.53.40.71.dialog.cgatepro;lr>
list_route: hop: <sip:212.53.35.244:5060;lr>
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-136174@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK13c47bee;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.71:5060;lr>,<sip:312653-212.53.40.71.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Contact: <sip:user@212.122.x.x>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 103 ACK
User-Agent: Usr
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.1.6 port 16014
Video is at 192.168.1.6 port 18442
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.43;branch=z9hG4bKsyufaoni;received=192.168.1.43;rport=5060
From: "201" <sip:201@192.168.1.6>;tag=isiyh
To: <sip:74957757566@192.168.1.6>;tag=as589b0e88
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 INVITE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:74957757566@192.168.1.6>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 18815 18816 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 16014 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
We're at 212.122.x.x port 12684
Video is at 212.122.x.x port 10784
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x200 (speex) to SDP
14 headers, 11 lines
sip-gw*CLI>
<-- SIP read from 192.168.1.43:5060:
ACK sip:74957757566@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.43;rport;branch=z9hG4bKiqivsnmf
Max-Forwards: 70
Proxy-Authorization: Digest username="201",realm="asterisk",nonce="38433a2b",uri="sip:74957757566@192.168.1.6",response="a5f436e27fd3e6e3fb7044eb0fc485c6",algorithm=MD5
To: <sip:74957757566@192.168.1.6>;tag=as589b0e88
From: "201" <sip:201@192.168.1.6>;tag=isiyh
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 963 ACK
User-Agent: Twinkle/1.1
Content-Length: 0
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:201@192.168.1.43> for address/port to send to
set_destination: set destination to 192.168.1.43, port 5060
We're at 192.168.1.6 port 16014
Video is at 192.168.1.6 port 18442
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.1.43:5060:
INVITE sip:201@192.168.1.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK72d35c20;rport
From: <sip:74957757566@192.168.1.6>;tag=as589b0e88
To: "201" <sip:201@192.168.1.6>;tag=isiyh
Contact: <sip:74957757566@192.168.1.6>
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 102 INVITE
User-Agent: Usr
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 18815 18817 IN IP4 212.53.40.82
s=session
c=IN IP4 212.53.40.82
t=0 0
m=audio 21588 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
sip-gw*CLI>
<-- SIP read from 192.168.1.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;branch=z9hG4bK72d35c20
To: "201" <sip:201@192.168.1.6>;tag=isiyh
From: <sip:74957757566@192.168.1.6>;tag=as589b0e88
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 102 INVITE
Contact: <sip:201@192.168.1.43>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.1
Supported: replaces,norefersub
Content-Length: 202
v=0
o=201 1770785376 570897783 IN IP4 192.168.1.43
s=-
c=IN IP4 192.168.1.43
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.43:8000
Peer video RTP is at port 192.168.1.43:65535
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:201@192.168.1.43>
set_destination: Parsing <sip:201@192.168.1.43> for address/port to send to
set_destination: set destination to 192.168.1.43, port 5060
Transmitting (no NAT) to 192.168.1.43:5060:
ACK sip:201@192.168.1.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7af62bf0;rport
From: <sip:74957757566@192.168.1.6>;tag=as589b0e88
To: "201" <sip:201@192.168.1.6>;tag=isiyh
Contact: <sip:74957757566@192.168.1.6>
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 102 ACK
User-Agent: Usr
Max-Forwards: 70
Content-Length: 0
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1db4436c;rport=5060
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 104 INVITE
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (8 headers 0 lines) ---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.122.x.x:5060;rport=5060;branch=z9hG4bK1db4436c
From: <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 104 INVITE
Server: TarioSoftswitch/3.2.10
Content-Length: 0
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:212.53.40.40:5060;lr> for address/port to send to
set_destination: set destination to 212.53.40.40, port 5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:proc-136174@212.53.35.244 SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1db4436c;rport
Route: <sip:212.53.40.40:5060;lr>,<sip:212.53.40.71:5060;lr>,<sip:312653-212.53.40.71.dialog.cgatepro;lr>,<sip:212.53.35.244:5060;lr>
From: "SipPhone" <sip:user@sipnet.ru>;tag=as3442dc46
To: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
Contact: <sip:user@212.122.x.x>
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 104 ACK
User-Agent: Usr
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call 'cjlqmdbgmjanzth@192.168.1.43' in 32000 ms
set_destination: Parsing <sip:201@192.168.1.43> for address/port to send to
set_destination: set destination to 192.168.1.43, port 5060
Reliably Transmitting (no NAT) to 192.168.1.43:5060:
BYE sip:201@192.168.1.43 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK27a3b839;rport
From: <sip:74957757566@192.168.1.6>;tag=as589b0e88
To: "201" <sip:201@192.168.1.6>;tag=isiyh
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 103 BYE
User-Agent: Usr
Max-Forwards: 70
Content-Length: 0
---
sip-gw*CLI>
<-- SIP read from 192.168.1.43:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;branch=z9hG4bK27a3b839
To: "201" <sip:201@192.168.1.6>;tag=isiyh
From: <sip:74957757566@192.168.1.6>;tag=as589b0e88
Call-ID: cjlqmdbgmjanzth@192.168.1.43
CSeq: 103 BYE
Server: Twinkle/1.1
Content-Length: 0
--- (8 headers 0 lines) ---
Destroying call '7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru'
Destroying call 'cjlqmdbgmjanzth@192.168.1.43'
sip-gw*CLI> ~~~~~~~~~~~~~~~
No such command '~~~~~~~~~~~~~~~' (type 'help' for help)
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
sip-gw*CLI>
Apr 15 00:50:49 NOTICE[18815]: chan_sip.c:5460 sip_reregister: -- Re-registration for user@sipnet.ru
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK00df1a32;rport
From: <sip:user@sipnet.ru>;tag=as6e0e9def
To: <sip:user@sipnet.ru>
Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.43
CSeq: 116 REGISTER
User-Agent: Usr
Max-Forwards: 70
Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="1E6BFA84146B441D1C62", response="c7e34d3459edae65d95fb78737b29dfb", opaque="opaqueData", qop=auth, cnonce="521aa385", nc=00000002
Expires: 120
Contact: <sip:4235623@212.122.x.x>
Event: registration
Content-Length: 0
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK00df1a32;rport=5060
From: <sip:user@sipnet.ru>;tag=as6e0e9def
To: <sip:user@sipnet.ru>;tag=636BD494
Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6
CSeq: 116 REGISTER
Expires: 120
Contact: <sip:4235623@212.122.x.x>;expires=120
Event: registration
Date: Mon, 14 Apr 2008 13:50:49 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (15 headers 0 lines) ---
Scheduling destruction of call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6' in 32000 ms
Apr 15 00:50:49 NOTICE[18815]: chan_sip.c:9979 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 120 sec (Scheduling reregistration in 105 s)
Destroying call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6'
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
BYE sip:user@212.122.x.x SIP/2.0
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK741122-lnqxjfj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport
Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK2014814026
Via: SIP/2.0/TCP 212.53.35.244:50666
Max-Forwards: 68
From: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
To: <sip:user@sipnet.ru>;tag=as3442dc46
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 53496 BYE
User-Agent: TarioSoftswitch/3.2.10
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 212.53.40.40:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK741122-lnqxjfj;cgp=etc.tario.ru;upaddr=212.53.35.244;received=212.53.40.40;rport=5060
Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK2014814026
Via: SIP/2.0/TCP 212.53.35.244:50666
From: <sip:74957757566@sipnet.ru>;tag=710a1ca5-136174
To: <sip:user@sipnet.ru>;tag=as3442dc46
Call-ID: 7b6fcb1505b80e2a222b041b2f7329b8@sipnet.ru
CSeq: 53496 BYE
User-Agent: Usr
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Apr 15 00:52:34 NOTICE[18815]: chan_sip.c:5460 sip_reregister: -- Re-registration for user@sipnet.ru
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1245854e;rport
From: <sip:user@sipnet.ru>;tag=as7c66f301
To: <sip:user@sipnet.ru>
Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6
CSeq: 117 REGISTER
User-Agent: Usr
Max-Forwards: 70
Authorization: Digest username="user", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="1E6BFA84146B441D1C62", response="2b7cc1f06931754357723c2255d320b7", opaque="opaqueData", qop=auth, cnonce="50e65be7", nc=00000003
Expires: 120
Contact: <sip:4235623@212.122.x.x>
Event: registration
Content-Length: 0
---
sip-gw*CLI>
<-- SIP read from 212.53.40.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.122.x.x:5060;branch=z9hG4bK1245854e;rport=5060
From: <sip:user@sipnet.ru>;tag=as7c66f301
To: <sip:user@sipnet.ru>;tag=07737923
Call-ID: 3fa188a21eb4b93353d294f07baaba69@192.168.1.6
CSeq: 117 REGISTER
Expires: 120
Contact: <sip:4235623@212.122.x.x>;expires=120
Event: registration
Date: Mon, 14 Apr 2008 13:52:34 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.2.2p
Content-Length: 0
--- (15 headers 0 lines) ---
Scheduling destruction of call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6' in 32000 ms
Apr 15 00:52:34 NOTICE[18815]: chan_sip.c:9979 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 120 sec (Scheduling reregistration in 105 s)
Destroying call '3fa188a21eb4b93353d294f07baaba69@192.168.1.6'