<div dir="ltr"><div>Note:<br></div>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).<br><br><br></div><div class="gmail_extra"><br><br>
<div class="gmail_quote">2013/8/6 Alexandr Usov <span dir="ltr"><<a href="mailto:blessendor@gmail.com" target="_blank">blessendor@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div>Sorry, It was call wothout answering.<br><br></div>I'm disable rtp debug and got full sip trace on asterisk side.<br><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
INVITE <a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a> SIP/2.0<br>Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0<br>
Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5<br>Max-Forwards: 16<br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>
CSeq: 12894 INVITE<br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Supported: replaces, 100rel, timer, norefersub<br>Session-Expires: 1800<br>Min-SE: 90<br>User-Agent: Telephone 1.0.4<br>
Content-Type: application/sdp<br>Content-Length: 461<br><br>v=0<br>o=- 3584774018 3584774018 IN IP4 1.1.1.1<br>s=pjmedia<br>c=IN IP4 1.1.1.1<br>t=0 0<br>a=X-nat:0<br>m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101<br>a=rtcp:45033<br>
a=rtpmap:103 speex/16000<br>a=rtpmap:102 speex/8000<br>a=rtpmap:104 speex/32000<br>a=rtpmap:109 iLBC/8000<br>a=fmtp:109 mode=30<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:9 G722/8000<br>
a=sendrecv<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=nortpproxy:yes<br><-------------><br>--- (18 headers 21 lines) ---<br>Sending to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> (no NAT)<br>
Sending to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> (no NAT)<br>
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>Found peer '101' for '101' from <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a><br> == Using SIP RTP CoS mark 5<br>
Found RTP audio format 103<br>
Found RTP audio format 102<br>Found RTP audio format 104<br>Found RTP audio format 109<br>Found RTP audio format 3<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found RTP audio format 9<br>Found RTP audio format 101<br>
Found audio description format speex for ID 103<br>Found audio description format speex for ID 102<br>Found audio description format speex for ID 104<br>Found audio description format iLBC for ID 109<br>Found audio description format GSM for ID 3<br>
Found audio description format PCMU for ID 0<br>Found audio description format PCMA for ID 8<br>Found audio description format G722 for ID 9<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<br>Peer audio RTP is at port <a href="http://1.1.1.1:45032" target="_blank">1.1.1.1:45032</a><br>
Looking for 101 in 1-internal (domain <a href="http://sip1.domain.com.ua" target="_blank">sip1.domain.com.ua</a>)<br>
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5<br>Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12894 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>
Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>Content-Length: 0<br><br><br><------------><br> -- Executing [101@1-internal:1] Macro("SIP/101-00000510", "1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack<br>
-- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "") in new stack<br> -- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/<a href="mailto:101@sip1.domain.com.ua" target="_blank">101@sip1.domain.com.ua</a>,60,rTt") in new stack<br>
== Using SIP RTP CoS mark 5<br>Audio is at 14084<br>Adding codec 100003 (ulaw) to SDP<br>Adding codec 100002 (gsm) to SDP<br>Adding codec 100004 (alaw) to SDP<br>Adding codec 100017 (testlaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>
Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br>INVITE <a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241<br>
Max-Forwards: 70<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>
Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk Cloud PBX 1.0<br>Date: Tue, 06 Aug 2013 10:33:43 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 319<br><br>v=0<br>o=root 2136064201 2136064201 IN IP4 192.168.144.101<br>s=Asterisk Cloud PBX 1.0<br>c=IN IP4 192.168.144.101<br>t=0 0<br>m=audio 14084 RTP/AVP 0 3 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>
a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called SIP/<a href="mailto:101@sip1.domain.com.ua" target="_blank">101@sip1.domain.com.ua</a><br>
<br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12894 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>
Content-Length: 0<br><br><br><------------><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 100 trying -- your call is important to us<br>Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080<br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>
Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>Server: kamailio (4.0.2 (x86_64/linux))<br>Content-Length: 0<br><br><-------------><br>--- (8 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241<br>Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes><br>Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes><br>Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO<br>
CSeq: 102 INVITE<br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Content-Length: 0<br><br><-------------><br>
--- (11 headers 0 lines) ---<br>list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes><br>list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes><br> -- SIP/sip1.domain.com.ua-00000511 is ringing<br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12894 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>
Content-Length: 0<br><br><br><------------><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241<br>
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes><br>Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes><br>Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO<br>
CSeq: 102 INVITE<br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Supported: replaces, 100rel, timer, norefersub<br>
Content-Type: application/sdp<br>Content-Length: 253<br><br>v=0<br>o=- 3584774018 3584774019 IN IP4 1.1.1.1<br>s=pjmedia<br>c=IN IP4 1.1.1.1<br>t=0 0<br>a=X-nat:0<br>m=audio 57312 RTP/AVP 0 101<br>a=rtcp:57313<br>a=rtpmap:0 PCMU/8000<br>
a=sendrecv<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=nortpproxy:yes<br><-------------><br>--- (13 headers 13 lines) ---<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Found audio description format PCMU for ID 0<br>
Found audio description format telephone-event for ID 101<br>Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<br>
Peer audio RTP is at port <a href="http://1.1.1.1:57312" target="_blank">1.1.1.1:57312</a><br>list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes><br>list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes><br>set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port to send to<br>
set_destination: set destination to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a><br>Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br>ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4<br>
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes><br>Max-Forwards: 70<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>
To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO<br>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>
Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk Cloud PBX 1.0<br>Content-Length: 0<br>
<br><br>
---<br> -- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510<br>Audio is at 18570<br>Adding codec 100003 (ulaw) to SDP<br>Adding codec 100004 (alaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br>
<--- Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12894 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>
Content-Type: application/sdp<br>Require: timer<br>Content-Length: 294<br><br>v=0<br>o=root 794877266 794877266 IN IP4 192.168.144.101<br>s=Asterisk Cloud PBX 1.0<br>c=IN IP4 192.168.144.101<br>t=0 0<br>m=audio 18570 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><------------><br> > 0x7f2b60530ea0 -- Probation passed - setting RTP source address to <a href="http://1.1.1.1:57312" target="_blank">1.1.1.1:57312</a><br>
<br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>ACK <a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a> SIP/2.0<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX<br>
Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU<br>Max-Forwards: 16<br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12894 ACK<br>Content-Length: 0<br>
<br><-------------><br>
--- (9 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>INVITE <a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a> SIP/2.0<br>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0<br>
Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4<br>Max-Forwards: 16<br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>
CSeq: 12895 INVITE<br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Supported: replaces, 100rel, timer, norefersub<br>Session-Expires: 1800;refresher=uas<br>Min-SE: 90<br>Content-Type: application/sdp<br>
Content-Length: 253<br><br>v=0<br>o=- 3584774018 3584774019 IN IP4 1.1.1.1<br>s=pjmedia<br>c=IN IP4 1.1.1.1<br>t=0 0<br>a=X-nat:0<br>m=audio 45032 RTP/AVP 0 101<br>a=rtcp:45033<br>a=rtpmap:0 PCMU/8000<br>a=sendrecv<br>a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>a=nortpproxy:yes<br><-------------><br>--- (15 headers 13 lines) ---<br>Sending to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> (no NAT)<br>Found RTP audio format 0<br>Found RTP audio format 101<br>
Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<br>
Peer audio RTP is at port <a href="http://1.1.1.1:45032" target="_blank">1.1.1.1:45032</a><br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2<br>
Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4<br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12895 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>
Content-Length: 0<br><br><br><------------><br>Audio is at 18570<br>Adding codec 100003 (ulaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4<br>
From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12895 INVITE<br>Server: Asterisk Cloud PBX 1.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>
Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><br>Content-Type: application/sdp<br>Require: timer<br>Content-Length: 270<br><br>v=0<br>o=root 794877266 794877267 IN IP4 192.168.144.101<br>
s=Asterisk Cloud PBX 1.0<br>c=IN IP4 192.168.144.101<br>t=0 0<br>m=audio 18570 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br>
<br><------------><br> > 0x7f2b60141b30 -- Probation passed - setting RTP source address to <a href="http://1.1.1.1:45032" target="_blank">1.1.1.1:45032</a><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
ACK <a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a> SIP/2.0<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i<br>
Max-Forwards: 16<br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 12895 ACK<br>Content-Length: 0<br><br><-------------><br>--- (9 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
BYE <a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a> SIP/2.0<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq<br>
Max-Forwards: 16<br>From: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO<br>To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>
Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>CSeq: 18603 BYE<br>User-Agent: Telephone 1.0.4<br>Content-Length: 0<br>
<br><-------------><br>
--- (10 headers 0 lines) ---<br>Sending to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> (no NAT)<br>Scheduling destruction of SIP dialog '<a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a>' in 32000 ms (Method: BYE)<br>
<br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq<br>
From: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO<br>To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as19dbc43e<br>
Call-ID: <a href="mailto:025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua" target="_blank">025be3512347f9f21b00d1930fa8f4bc@sip1.domain.com.ua</a><br>CSeq: 18603 BYE<br>Server: Asterisk Cloud PBX 1.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br> == Spawn extension (macro-1-internal, s, 2) exited non-zero on 'SIP/101-00000510' in macro '1-internal'<br> == Spawn extension (1-internal, 101, 1) exited non-zero on 'SIP/101-00000510'<br>
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in 6400 ms (Method: ACK)<br>set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for address/port to send to<br>set_destination: set destination to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a><br>
Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br>BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0<br>Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754<br>Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes><br>
Max-Forwards: 70<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>To: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 102 BYE<br>User-Agent: Asterisk Cloud PBX 1.0<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br>Retransmitting #1 (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br>
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0<br>Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754<br>Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes><br>Max-Forwards: 70<br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>To: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>CSeq: 102 BYE<br>User-Agent: Asterisk Cloud PBX 1.0<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754<br>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as7db5bb42<br>
To: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC<br>CSeq: 102 BYE<br>Content-Length: 0<br><br><-------------><br>
--- (7 headers 0 lines) ---<br>
SIP Response message for INCOMING dialog BYE arrived<br>Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/8/6 SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Hi again,</div><div><br></div>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.<div>
<br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <span dir="ltr"><<a href="mailto:blessendor@gmail.com" target="_blank">blessendor@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><br><------------><br> Dial (.......) in new stack<br> <br> <br> == Using SIP RTP CoS mark 5<br>
Audio is at 19614<br>Adding codec 100003 (ulaw) to SDP<br>Adding codec 100002 (gsm) to SDP<br>Adding codec 100004 (alaw) to SDP<br>
Adding codec 100017 (testlaw) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<div><br>Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br></div>INVITE <a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a> SIP/2.0<br>
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299<br>Max-Forwards: 70<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>
To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>
Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk<br>Date: Tue, 06 Aug 2013 10:18:03 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 319<br><br>v=0<br>o=root 1885227245 1885227245 IN IP4 2.2.2.101<br>s=Asterisk<br>c=IN IP4 2.2.2.101<br>t=0 0<br>m=audio 19614 RTP/AVP 0 3 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called SIP/<a href="mailto:101@sip1.domain.com.ua" target="_blank">101@sip1.domain.com.ua</a><br>
<br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1bd39f9d<br>Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9<br>CSeq: 10050 INVITE<br>Server: Asterisk<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Content-Length: 0<br><br><br><------------><br>
<br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 100 trying -- your call is important to us<br>Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080<br>From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>
To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
CSeq: 102 INVITE<br>Server: kamailio (4.0.2 (x86_64/linux))<br>Content-Length: 0<br><br><-------------><br>--- (8 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299<br>Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes><br>Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes><br>Call-ID: <a href="mailto:2ac37537499c919f01683582349522d6@sip1.domain.com.ua" target="_blank">2ac37537499c919f01683582349522d6@sip1.domain.com.ua</a><br>
From: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1b8070ba<br>To: <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV<br>
CSeq: 102 INVITE<br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br>Content-Length: 0<br><br><-------------><br>
--- (11 headers 0 lines) ---<br>list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes><br>list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes><br> -- SIP/sip1.domain.com.ua-0000050f is ringing<br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2<br>Via: SIP/2.0/UDP 192.168.10.240:52396;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-<br>
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes><br>Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes><br>From: "101" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD<br>
To: "Bomber" <<a href="mailto:sip%3A101@sip1.domain.com.ua" target="_blank">sip:101@sip1.domain.com.ua</a>>;tag=as1bd39f9d<br>Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9<br>CSeq: 10050 INVITE<br>Server: Asterisk<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:101@2.2.2.101:5080" target="_blank">sip:101@2.2.2.101:5080</a>><br>Content-Length: 0<br><br><br><------------></div>
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