[SR-Users] (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Alexandr Usov
blessendor at gmail.com
Tue Aug 6 12:36:50 CEST 2013
Sorry, It was call wothout answering.
I'm disable rtp debug and got full sip trace on asterisk side.
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101 at sip1.domain.com.ua SIP/2.0
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Max-Forwards: 16
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>
Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 461
v=0
o=- 3584774018 3584774018 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:45033
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (18 headers 21 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Sending to 2.2.2.2:5060 (no NAT)
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
Found peer '101' for '101' from 2.2.2.2:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
Looking for 101 in 1-internal (domain sip1.domain.com.ua)
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Length: 0
<------------>
-- Executing [101 at 1-internal:1] Macro("SIP/101-00000510",
"1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack
-- Executing [s at macro-1-internal:1] NoOp("SIP/101-00000510", "") in new
stack
-- Executing [s at macro-1-internal:2] Dial("SIP/101-00000510", "SIP/
101 at sip1.domain.com.ua,60,rTt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14084
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101 at sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241
Max-Forwards: 70
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101 at sip1.domain.com.ua>
Contact: <sip:101 at 192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk Cloud PBX 1.0
Date: Tue, 06 Aug 2013 10:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 2136064201 2136064201 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 14084 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101 at sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101 at sip1.domain.com.ua>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101 at sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-00000511 is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101 at sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 57312 RTP/AVP 0 101
a=rtcp:57313
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:57312
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port to
send to
set_destination: set destination to 2.2.2.2:5060
Transmitting (no NAT) to 2.2.2.2:5060:
ACK sip:101 at CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101 at sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
Contact: <sip:101 at 192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
CSeq: 102 ACK
User-Agent: Asterisk Cloud PBX 1.0
Content-Length: 0
---
-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 294
v=0
o=root 794877266 794877266 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
> 0x7f2b60530ea0 -- Probation passed - setting RTP source address to
1.1.1.1:57312
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101 at 192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU
Max-Forwards: 16
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101 at 192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
Max-Forwards: 16
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 0 101
a=rtcp:45033
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (15 headers 13 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Length: 0
<------------>
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101 at 192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 270
v=0
o=root 794877266 794877267 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
> 0x7f2b60141b30 -- Probation passed - setting RTP source address to
1.1.1.1:45032
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101 at 192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i
Max-Forwards: 16
From: "101" <sip:101 at sip1.domain.com.ua
>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:101 at 192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
Max-Forwards: 16
From: <sip:101 at sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
CSeq: 18603 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Scheduling destruction of SIP dialog '
025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua' in 32000 ms (Method:
BYE)
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
From: <sip:101 at sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc at sip1.domain.com.ua
CSeq: 18603 BYE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (macro-1-internal, s, 2) exited non-zero on
'SIP/101-00000510' in macro '1-internal'
== Spawn extension (1-internal, 101, 1) exited non-zero on
'SIP/101-00000510'
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in
6400 ms (Method: ACK)
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:101 at CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101 at sip1.domain.com.ua>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
BYE sip:101 at CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101 at sip1.domain.com.ua>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101 at sip1.domain.com.ua>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK
2013/8/6 SamyGo <govoiper at gmail.com>
> Hi again,
>
> Still Missing 200OK for this call. It'll be helpful to send a complete
> trace for the call coming in to the Asterisk at first place and then
> Dialing out to the B-leg whose trace which you've just shared.
>
>
>
>
> On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor at gmail.com>wrote:
>
>>
>>
>> <------------>
>> Dial (.......) in new stack
>>
>>
>> == Using SIP RTP CoS mark 5
>> Audio is at 19614
>> Adding codec 100003 (ulaw) to SDP
>> Adding codec 100002 (gsm) to SDP
>> Adding codec 100004 (alaw) to SDP
>> Adding codec 100017 (testlaw) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>>
>> Reliably Transmitting (no NAT) to 2.2.2.2:5060:
>> INVITE sip:101 at sip1.domain.com.ua SIP/2.0
>> Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
>> Max-Forwards: 70
>> From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
>> To: <sip:101 at sip1.domain.com.ua>
>> Contact: <sip:101 at 2.2.2.101:5080>
>> Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
>> CSeq: 102 INVITE
>> User-Agent: Asterisk
>> Date: Tue, 06 Aug 2013 10:18:03 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 319
>>
>> v=0
>> o=root 1885227245 1885227245 IN IP4 2.2.2.101
>> s=Asterisk
>> c=IN IP4 2.2.2.101
>> t=0 0
>> m=audio 19614 RTP/AVP 0 3 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> ---
>> -- Called SIP/101 at sip1.domain.com.ua
>>
>> <--- Transmitting (no NAT) to 2.2.2.2:5060 --->
>> SIP/2.0 180 Ringing
>> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
>> Via: SIP/2.0/UDP 192.168.10.240:52396
>> ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
>> Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
>> Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
>> From: "101" <sip:101 at sip1.domain.com.ua
>> >;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
>> To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1bd39f9d
>> Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
>> CSeq: 10050 INVITE
>> Server: Asterisk
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Session-Expires: 1800;refresher=uas
>> Contact: <sip:101 at 2.2.2.101:5080>
>> Content-Length: 0
>>
>>
>> <------------>
>>
>> <--- SIP read from UDP:2.2.2.2:5060 --->
>> SIP/2.0 100 trying -- your call is important to us
>> Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
>> From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
>> To: <sip:101 at sip1.domain.com.ua>
>> Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
>> CSeq: 102 INVITE
>> Server: kamailio (4.0.2 (x86_64/linux))
>> Content-Length: 0
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>>
>> <--- SIP read from UDP:2.2.2.2:5060 --->
>> SIP/2.0 180 Ringing
>> Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
>> Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
>> Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
>> Call-ID: 2ac37537499c919f01683582349522d6 at sip1.domain.com.ua
>> From: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1b8070ba
>> To: <sip:101 at sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
>> CSeq: 102 INVITE
>> Contact: "101" <sip:101 at CLIENT.GW.PUB.IP:17303;ob>
>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
>> MESSAGE, OPTIONS
>> Content-Length: 0
>>
>> <------------->
>> --- (11 headers 0 lines) ---
>> list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
>> list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
>> -- SIP/sip1.domain.com.ua-0000050f is ringing
>>
>> <--- Transmitting (no NAT) to 2.2.2.2:5060 --->
>> SIP/2.0 180 Ringing
>> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
>> Via: SIP/2.0/UDP 192.168.10.240:52396
>> ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
>> Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
>> Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
>> From: "101" <sip:101 at sip1.domain.com.ua
>> >;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
>> To: "Bomber" <sip:101 at sip1.domain.com.ua>;tag=as1bd39f9d
>> Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
>> CSeq: 10050 INVITE
>> Server: Asterisk
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Session-Expires: 1800;refresher=uas
>> Contact: <sip:101 at 2.2.2.101:5080>
>> Content-Length: 0
>>
>>
>> <------------>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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