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

Tue Aug 6 14:50:50 CEST 2013

As I posted resently:

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 SamyGo <govoiper at gmail.com>

> 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
>>
>>
>
> _______________________________________________
> 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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