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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@sip1.domain.com.ua Contact: "101" sip:101@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@192.168.144.101:5080 Content-Length: 0
<------------> -- Executing [101@1-internal:1] Macro("SIP/101-00000510", "1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack -- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "") in new stack -- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/ 101@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@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@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua Call-ID: 025be3512347f9f21b00d1930fa8f4bc@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@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO CSeq: 102 INVITE Contact: "101" sip:101@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@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@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@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@sip1.domain.com.ua;tag=as19dbc43e To: sip:101@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO Contact: sip:101@192.168.144.101:5080 Call-ID: 025be3512347f9f21b00d1930fa8f4bc@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@sip1.domain.com.ua;tag=as7db5bb42 Contact: "101" sip:101@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@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@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To: "Bomber" sip:101@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@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@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@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@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@sip1.domain.com.ua;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO To: "Bomber" sip:101@sip1.domain.com.ua;tag=as19dbc43e Call-ID: 025be3512347f9f21b00d1930fa8f4bc@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@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@sip1.domain.com.ua;tag=as7db5bb42 To: "101" sip:101@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@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@sip1.domain.com.ua;tag=as7db5bb42 To: "101" sip:101@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@sip1.domain.com.ua;tag=as7db5bb42 To: "101" sip:101@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@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@gmail.comwrote:
<------------> 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@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@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Contact: sip:101@2.2.2.101:5080 Call-ID: 2ac37537499c919f01683582349522d6@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@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@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" sip:101@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@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@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua Call-ID: 2ac37537499c919f01683582349522d6@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@sip1.domain.com.ua From: "Bomber" sip:101@sip1.domain.com.ua;tag=as1b8070ba To: sip:101@sip1.domain.com.ua;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV CSeq: 102 INVITE Contact: "101" sip:101@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@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" sip:101@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@2.2.2.101:5080 Content-Length: 0
<------------>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users