sip - Opensips byes rejected by client by 481 during redirection -

server opensips server 1.10.0-tls (linux). can handle conversations to/from local stations , has been updated allow stations external systems. changing username, ip , port in $ru if station doesn't exist locally ($tu untouched). works fine invites, calls , similar messages.

the problem i'm having byes coming external server, being passed on local client station, being rejected 481 (call leg/transaction not exist) can confirm coming client software, yet accepts byes local stations on same server without bother. calls local local end ok, calls local external shutdown ok, it's calls external caller local callee won't close (it callee saying 481).

i understand caused transaction matching not occurring due different in tags in to/from , call-id; understand (like $ru script) changes parts of $ru might have effect on hash determine transaction, don't change tags or callid, $ru name make go right ip , station name.

my question how go solving on server without changing client applications? i've included examples of messages being sent below taken wireshark capture on client workstation, i'm unsure i'm doing wrong..i've tried different things on server no luck. there way mark or tell client via sip messages shutdown conversation regardless of transaction matching?

i appreciate aid in i've been pulling hair out time.

message examples external caller (103 on server 5.44) local callee (local name wks2, external ref name 155, on server 3.3, client 3.0). first bye problem, second bye me closing hanging connection on client.

---------- invite sip:895eedf7-4256-44d0-9edf-39785b6ceef0@172.16.3.0:5050 sip/2.0       record-route: <sip:172.16.3.3;lr;ftag=as678f227c>       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bkd066.93df6e05.0       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk50029c58;rport=5060       max-forwards: 69       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       to: <sip:155@172.16.3.3:5060>       contact: <sip:103@172.16.5.44:5060>       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 102 invite       user-agent: asterisk pbx 1.8.13.1~dfsg1-3+deb7u3       date: thu, 25 jun 2015 14:27:31 gmt       allow: invite, ack, cancel, options, bye, refer, subscribe, notify, info, publish       supported: replaces, timer       content-type: application/sdp       content-length: 296       redirect-to: sip:wks2@172.16.3.3:5060   ---------- sip/2.0 180 ringing       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bkd066.93df6e05.0;received=172.16.3.3;rport=5060       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk50029c58;rport=5060       to: <sip:155@172.16.3.3:5060>;tag=1262186908       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 102 invite       allow: ack, bye, cancel, info, invite, notify, options, refer, register, subscribe       content-length: 0   ---------- sip/2.0 200 ok       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bkd066.93df6e05.0;received=172.16.3.3;rport=5060       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk50029c58;rport=5060       to: <sip:155@172.16.3.3:5060>;tag=1262186908       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 102 invite       contact: <sip:172.16.3.0:5050>       record-route: <sip:172.16.3.3;lr;ftag=as678f227c>       server: www.sipsorcery.com       content-length: 161       content-type: application/sdp   ---------- ack sip:172.16.3.0:5050 sip/2.0       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bkd066.93df6e05.2       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk49d179f0;rport=5060       max-forwards: 69       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       to: <sip:155@172.16.3.3:5060>;tag=1262186908       contact: <sip:103@172.16.5.44:5060>       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 102 ack       user-agent: asterisk pbx 1.8.13.1~dfsg1-3+deb7u3       content-length: 0   ---------- bye sip:172.16.3.0:5050 sip/2.0       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bke066.64948323.0       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk1e381a63;rport=5060       max-forwards: 69       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       to: <sip:155@172.16.3.3:5060>;tag=1262186908       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 103 bye       user-agent: asterisk pbx 1.8.13.1~dfsg1-3+deb7u3       x-asterisk-hangupcause: normal clearing       x-asterisk-hangupcausecode: 16       content-length: 0   ---------- sip/2.0 481 calllegtransactiondoesnotexist       via: sip/2.0/udp 172.16.3.3:5060;branch=z9hg4bke066.64948323.0;received=172.16.3.3;rport=5060       via: sip/2.0/udp 172.16.5.44:5060;received=172.16.5.44;branch=z9hg4bk1e381a63;rport=5060       to: <sip:155@172.16.3.3:5060>;tag=1262186908       from: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 103 bye       allow: ack, bye, cancel, info, invite, notify, options, refer, register, subscribe       content-length: 0   ---------- bye sip:103@172.16.5.44:5060 sip/2.0       via: sip/2.0/udp 172.16.3.0:5050;branch=z9hg4bkfe13e63d99524b06846bde0fedbd8a69;rport       to: "station 103   " <sip:103@172.16.5.44>;tag=as678f227c       from: <sip:155@172.16.3.3:5060>;tag=1262186908       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 103 bye       max-forwards: 70       route: <sip:172.16.3.3;lr;ftag=as678f227c>       content-length: 0   ---------- sip/2.0 481 call leg/transaction not exist       via: sip/2.0/udp 172.16.3.0:5050;received=172.16.3.0;branch=z9hg4bkfe13e63d99524b06846bde0fedbd8a69;rport=5050       from: <sip:155@172.16.3.3:5060>;tag=1262186908       to: "station 103 " <sip:103@172.16.5.44>;tag=as678f227c       call-id: 2697521655d00bab6de06a5941ddf8b4@172.16.5.44:5060       cseq: 103 bye       server: asterisk pbx 1.8.13.1~dfsg1-3+deb7u3       allow: invite, ack, cancel, options, bye, refer, subscribe, notify, info, publish       supported: replaces, timer       content-length: 0   ---------- 

thank :-)

see comment above, issue client app following non-standard behaviour checking bye messages (it checked to, not to-tag). issue being resolved through client app change.