datacompboy
public text v1 · immutableSIP Debugging Enabled for IP: 217.151.230.182:5069
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
INVITE sip:111@sip.call2ru.com SIP/2.0
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>
Contact: <sip:120001@217.151.230.182:5069>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4072 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 145
v=0
o=120001 46618828 46618828 IN IP4 217.151.230.182
s=X-Lite
c=IN IP4 217.151.230.182
t=0 0
m=audio 9000 RTP/AVP 0
a=rtpmap:0 pcmu/8000
--- (11 headers 7 lines)---
Using INVITE request as basis request - 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
Sending to 217.151.230.182 : 5069 (NAT)
Reliably Transmitting (no NAT) to 217.151.230.182:5069:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20;received=217.151.230.182
rom: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as5f63ecd1
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4072 INVITE
User-Agent: Call2Ru service
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:111@195.224.39.103>
Proxy-Authenticate: Digest algorithm=MD5, realm="call2ru.com", nonce="09cd2640"
Content-Length: 0
---
Scheduling destruction of call '354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182' in 15000 ms
Found user '120001'
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
ACK sip:111@sip.call2ru.com SIP/2.0
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK97F00B7308C142E093AC2BB14C713B20
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as5f63ecd1
Contact: <sip:120001@217.151.230.182:5069>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4072 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
INVITE sip:111@sip.call2ru.com SIP/2.0
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>
Contact: <sip:120001@217.151.230.182:5069>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4073 INVITE
Proxy-Authorization: Digest username="120001",realm="call2ru.com",nonce="09cd2640",response="393ba5217012f40e958f44f
9c37e419e",uri="sip:111@sip.call2ru.com",algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 145
v=0
o=120001 46619218 46619218 IN IP4 217.151.230.182
s=X-Lite
c=IN IP4 217.151.230.182
t=0 0
m=audio 9000 RTP/AVP 0
a=rtpmap:0 pcmu/8000
--- (12 headers 7 lines)---
Using INVITE request as basis request - 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
Sending to 217.151.230.182 : 5069 (NAT)
Found user '120001'
Found RTP audio format 0
Peer audio RTP is at port 217.151.230.182:9000
Peer video RTP is at port 217.151.230.182:65535
Found description format pcmu
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 111 in default (domain sip.call2ru.com)
list_route: hop: <sip:120001@217.151.230.182:5069>
Transmitting (no NAT) to 217.151.230.182:5069:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4;received=217.151.230.182
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4073 INVITE
User-Agent: Call2Ru service
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:111@195.224.39.103>
Content-Length: 0
---
-- Executing Answer("SIP/120001-08303fa0", "") in new stack
We're at 195.224.39.103 port 15158
Video is at 195.224.39.103 port 13882
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 217.151.230.182:5069:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK7C09A4CC58D542A8A22F0CD62E921DA4;received=217.151.230.182
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as7602845a
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4073 INVITE
User-Agent: Call2Ru service
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:111@195.224.39.103>
Content-Type: application/sdp
Content-Length: 162
v=0
o=root 8244 8244 IN IP4 195.224.39.103
s=session
c=IN IP4 195.224.39.103
t=0 0
m=audio 15158 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
---
-- Executing Playback("SIP/120001-08303fa0", "tt-weasels") in new stack
Sent RTP packet to 217.151.230.182:9000 (type 0, seq 19347, ts 160, len 160)
-- Playing 'tt-weasels' (language 'en')
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
ACK sip:111@195.224.39.103 SIP/2.0
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK79B1CC727AEB478D98AFDC157704210F
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as7602845a
Contact: <sip:120001@217.151.230.182:5069>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4073 ACK
Max-Forwards: 70
Content-Length: 0
--- (9 headers 0 lines)---
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
BYE sip:111@195.224.39.103 SIP/2.0
Via: SIP/2.0/UDP 217.151.230.182:5069;rport;branch=z9hG4bK352A308E84E84D028454954A8AE1D4FE
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as7602845a
Contact: <sip:120001@217.151.230.182:5069>
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4074 BYE
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0
--- (10 headers 0 lines)---
Sending to 217.151.230.182 : 5069 (NAT)
Transmitting (NAT) to 217.151.230.182:5069:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.151.230.182:5069;branch=z9hG4bK352A308E84E84D028454954A8AE1D4FE;received=217.151.230.182;rport=
5069
From: call2ru <sip:120001@sip.call2ru.com>;tag=2472302208
To: <sip:111@sip.call2ru.com>;tag=as7602845a
Call-ID: 354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182
CSeq: 4074 BYE
User-Agent: Call2Ru service
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:111@195.224.39.103>
Content-Length: 0
---
> cdr_odbc: Query Successful!
Destroying call '354AC419-A3F3-4EA2-ACD4-71041049372A@217.151.230.182'
call2ru*CLI>
<-- SIP read from 217.151.230.182:5069:
--- (0 headers 0 lines) Nat keepalive ---
*** Phone receive one RTP packet (I have heared it)
*** Phone send outgoing RTP, tcpdump on server saw it.