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

[SamyGo]

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
>
>
> <------------>
>
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