<div dir="ltr">I wonder who this belongs to : <span style="font-family:arial,sans-serif;font-size:13px">c=IN IP4 192.168.144.101</span><div><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div><div><span style="font-family:arial,sans-serif;font-size:13px">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.</span></div>
<div><br></div><div><span style="font-family:arial,sans-serif;font-size:13px">Is your kamailio set to have "mhomed=yes"; just wanted to know.</span></div><div><span style="font-family:arial,sans-serif;font-size:13px"><br>
</span></div><div><font face="arial, sans-serif">---</font></div><div><span style="font-family:arial,sans-serif;font-size:13px">Sammy </span></div><div><span style="font-family:arial,sans-serif;font-size:13px"><br></span></div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 3:36 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"><div>Sorry, It was call wothout answering.<br><br></div>I'm disable rtp debug and got full sip trace on asterisk side.<div class="im">
<br><br><br><--- SIP read from UDP:<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<div class="im"><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>
</div>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>><div class="im"><br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br></div>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>
CSeq: 12894 INVITE<div class="im"><br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br></div>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><div class="im"><br> == Using SIP RTP CoS mark 5<br>
</div>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><div class="im"><br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
</div>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<div class="im"><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></div>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<div class="im"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>
Session-Expires: 1800;refresher=uas<br></div>
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<div class="im">
<br>
== Using SIP RTP CoS mark 5<br></div>Audio is at 14084<div class="im"><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></div>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<div class="im"><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></div>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<div class="im"><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></div>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<div class="im">
<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></div>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<div class="im">
<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br></div>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><div class="im">
<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></div>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><div class="im"><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></div>Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241<div class="im"><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>
</div>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<div class="im">
<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></div> -- SIP/sip1.domain.com.ua-00000511 is ringing<div class="im"><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></div>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<div class="im">
<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></div>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<div class="im">
<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br></div>Contact: <<a href="http://sip:101@192.168.144.101:5080" target="_blank">sip:101@192.168.144.101:5080</a>><div class="im">
<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></div>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241<div class="im">
<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></div>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<div class="im"><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></div>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><div class="im"><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>
</div>set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port to send to<div class="im"><br>
set_destination: set destination to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a><br></div><div class="im">Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br></div>
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<div class="im"><br>Adding codec 100003 (ulaw) to SDP<br></div><div class="im">Adding codec 100004 (alaw) to SDP<br></div><div class="im">
Adding non-codec 0x1 (telephone-event) to SDP<br><br></div>
<--- 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<div class="im">
<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></div>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<div class="im">
<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br></div>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<div class="im"><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></div><------------><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><div class="im">
<br>
<br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br></div>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) ---<div class="im"><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br></div>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<div class="im"><br>Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob><br>
</div>Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii<br>
CSeq: 12895 INVITE<div class="im"><br>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS<br></div>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><div class="im"><br><br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
</div>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<div class="im">
<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br></div>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<div class="im"><br>Adding codec 100003 (ulaw) to SDP<br></div><div class="im">Adding non-codec 0x1 (telephone-event) to SDP<br><br></div><--- 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<div class="im"><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>
Session-Expires: 1800;refresher=uas<br></div>
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<div class="im"><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></div><------------><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><div class="im"><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
</div>
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) ---<div class="im"><br><br><--- SIP read from UDP:<a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br>
</div>
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)<div class="im">
<br>
<br><--- Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a> ---><br></div>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<div class="im"><br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br></div>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<div class="im"><br>
set_destination: set destination to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a><br></div><div class="im">
Reliably Transmitting (no NAT) to <a href="http://2.2.2.2:5060" target="_blank">2.2.2.2:5060</a>:<br></div>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<div class="im"><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></div>
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="HOEnZb"><div class="h5"><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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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
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