[SR-Users] No BYE from Called party

Hello All. I have kamailio with provider connection (trunk) When I call to external number through my provider call extablished Ok. But when i try hangup call from external number no BYE sended to me. When I hangup call from my kamailio (internal num) I send by to exteral number and it respond me Ok so session if fully complete. I guess that BYE from external number not recieves to me because I have wrong routing header fields at my INVITe or ACK messages, but can not find any information what what header must recieve info to external number where send BYE at hangup or thomething like this. This is my little dump for situation wherer I hangup from internal number and BYE finished successfully: IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599 E....< .@.'. ...6........G.RINVITE sip:12345678900@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Max-Forwards: 70 From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068> Contact:<TrunkNum@my.kamailio.com:5068> Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 04 Sep 2014 21:53:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com", nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri=" sip:12345678900@my.provider.com:5060", qop=auth, nc=00000001, cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5 v=0 o=root 1022912010 1022912010 IN IP4 my.kamailio.com s=Asterisk PBX 12.5.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30032 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=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30033 a=ice-ufrag:3o8JrqkF a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2 a=candidate:TgT1dfTnI3kBgWQ IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882 E..... ./...6... ........b..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE Contact: <sip:12345678900@externail.number.end.ip:5060;transport=udp> User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 746 X-provider agentOutboundGateway: sip:5574012345678900@62.93.147.149 X-provider agentOutboundCarrierID: 23705946361020 X-provider agentCarrierRate: 0.20180 X-provider agentCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com v=0 o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip s=FreeSWITCH c=IN IP4 externail.number.end.ip t=0 0 a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP m=audio 23216 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=ssrc:2990874569 cname:pRs5xP IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596 E..p.@..@.K\ ...6........\`.ACK sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617 E....D..@.KC ...6........q.ZBYE sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: Asterisk PBX 12.5.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568 E..T....-.676... ........@..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Thanks for help.