Hi ! After specific time I redirect (revert_uri and append_branch) call to another sip address. Everythig is ok for UA like ATA, Kphone and C7960). When the call is started from Grandstream after the pick up second site (Asterisk IVR- after redirection), connection is terminated afer a few seconds. This situations takes place (only for GS ) also when I redirect calls from one sip domain to another depends on prefix call (for client doesn't support URL sip addresses like GS, ATA) In logs ACK message directed to ser I see differences between UA. Originated destination sip address is 3000, when no answer, call is redirected to 4000 For ATA I have: 192.168.0.83:5060 -> 192.168.0.1:5060 ACK sip:3000@192.168.0.1 SIP/2.0.. Route: <sip:4000@192.168.0.81;branch=0>,<sip:4000@192.168.0.1:6060>.. Via: SIP/2.0/UDP 192.168.0.83:5060.. From: radan <sip:3100@sip.router.pl;user=phone>;tag=3207317092.. To: <sip:3000@sip.router.pl;user=phone>;tag=as2d60db53.. Call-ID: 3934861712(a)192.168.0.83.. CSeq: 1 ACK.. User-Agent: Cisco ATA 186 v3.0.0 atasip (031210A).. Content-Length: 0.... For GS I have: 192.168.0.84:5060 -> 192.168.0.1:5060 ACK sip:4000@192.168.0.1:6060 SIP/2.0.. Via: SIP/2.0/UDP 192.168.0.84.. Route:<sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>.. Route: <sip:4000@192.168.0.81;branch=0>.. From: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844.. To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868.. Contact:<sip:3102@192.168.0.84;user=phone>.. Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84.. CSeq: 65090 ACK.. User-Agent: Grandstream SIP UA 1.0.3.81.. Max-Forwards: 70.. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.. Content-Length: 0.... Two different calls are confirmed. for GS I have then following info a few times (5 or 6) 192.168.0.1:5060 -> 192.168.0.84:5060 SIP/2.0 200 OK.. Via: SIP/2.0/UDP 192.168.0.84.. Record-Route: <sip:4000@192.168.0.81;branch=0>.. Record-Route: <sip:3000@192.168.0.1;ftag=f6e3b058-8afd-fac2-e60b-e493a7d83844;lr>.. From: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844.. To: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868.. Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84.. CSeq: 65090 INVITE.. User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact: <sip:4000@192.168.0.1:6060>.. Content-Type: application/sdp.. Probably GS is not able to send ACK After them the ser sends BYE to the GS 192.168.0.1:5060 -> 192.168.0.84:5060 BYE sip:3102@192.168.0.84;user=phone SIP/2.0.. Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>.. Max-Forwards: 9 .. Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0.. Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0.. Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add.. From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868.. To: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844.. Contact: <sip:4000@192.168.0.1:6060>.. Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84.. CSeq: 102 BYE..User-Agent: Asterisk PBX Content-Length: 0.... a GS talks that this connection doesn't exist 192.168.0.84:5060 -> 192.168.0.1:5060 SIP/2.0 481 .. Via: SIP/2.0/UDP 192.168.0.1;branch=z9hG4bK2743.08055687.0.. Via: SIP/2.0/UDP 192.168.0.81;branch=z9hG4bKcc8e.cc7088e2.0.. Via: SIP/2.0/UDP 192.168.0.1:6060;branch=z9hG4bK0ff02add.. Record-Route: <sip:3000@192.168.0.1;ftag=as18e54868;lr>.. From: <sip:3000@sip.router.pl;user=phone>;tag=as18e54868.. To: "radan - grandstream" <sip:3102@sip.router.pl;user=phone>;tag=f6e3b058-8afd-fac2-e60b-e493a7d83844.. Call-ID: 683b3311-0ebd-55c5-4191-5469b058efb1(a)192.168.0.84.. CSeq: 102 BYE.. User-Agent: Grandstream SIP UA 1.0.3.81.. Content-Length: 0.... It is a some bug in soft for GS, or do I have to add something special in configuration file for GS ? Thanks Andrzej