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

Tue Aug 6 12:48:29 CEST 2013

I wonder who this belongs to : c=IN IP4 192.168.144.101

Also your Kamailio just sends the c=IN IP4 1.1.1.1 for the very first
incoming call that tells me that RTP proxy function is either running on
1.1.1.1 only or if its in bridging mode then you're not using the right
flag combination to use the 2.2.2.2 IP.

Is your kamailio set to have "mhomed=yes"; just wanted to know.

---
Sammy

On Tue, Aug 6, 2013 at 3:36 AM, Alexandr Usov <blessendor at gmail.com> wrote:

> 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
>>
>>
>
> _______________________________________________
> 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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