[SR-Users] (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)

Tue Aug 6 12:41:31 CEST 2013

Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network  (I am missed to change before copy-pasting here).

2013/8/6 Alexandr Usov <blessendor at gmail.com>

> 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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