[Serusers] Dynamic Redirect Issue about SER!

SER gurus: I am an newbie to SER usuage. At present a dynamic redirect issue happens to me. But it is very pity for SER not to support this case!! Problem Description: ==================== When configuring SER as only redirect server, config file regarding redirect routing section is partly writen as follows: #-----------------------request routing logic------------------------- #main routing logic if (method=="INVITE") { #rewrite current URI, which is always part of destination ser rewriteuri("sip:80000000@192.168.0.191:5060"); #redirect now sl_send_reply("301", "Redirect"); break; } #--------------------------------------------------------------------- where, redirecting destination URI is staticly configed into SER, but dynamic destination URI is what our IP-PBX product(RTCCP) want to hope. The following call flow diagram helps to understand above mentioned scenario. GrandStream HandyTone486 SER Our B2BUA IP PBX(RTCCP) IP: 192.168.0.253 IP: 192.168.0.252 IP: 192.168.0.191 +++++++++++++++++++++++++++++++++++++ First Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F1(INVITE)---------->| |<-----------------F2(302)------------>| |-----------------F3(ACK)------------->| |------------------------------------F4(INVITE)------------------------>| ( The subsequent call flow is omitted ) +++++++++++++++++++++++++++++++++++++ Second Call+++++++++++++++++++++++++++++++++++++++++++++++++ |-----------------F5(INVITE)---------->| |<-----------------F6(302)------------>| |-----------------F7(ACK)------------->| |------------------------------------F8(INVITE)------------------------>| ( The subsequent call flow is omitted ) F1: INVITE sip:8000000@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F2: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKb537ec0f1387845f From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=c526ecce44ec6d75 To: <sip:8000000@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f9ed Call-ID: d6169571fe4f59a2(a)192.168.0.233 CSeq: 50105 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6155 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:8000000@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F4: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKc6905874d1906c58 From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=1be4a5d7d169d8f1 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 0d325b0e778321ee(a)192.168.0.233 CSeq: 17809 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F5: INVITE sip:80000002@192.168.0.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 F6: SIP/2.0 301 Redirect Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bKe966424e91ceac8e From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=466d9b06b51ca7b4 To: <sip:80000002@192.168.0.252;user=phone>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7906 Call-ID: c755c6b0908a8ce9(a)192.168.0.233 CSeq: 25248 INVITE Contact: sip:80000000@192.168.0.191:5060 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.252:5060 "Noisy feedback tells: pid=6151 req_src_ip=192.168.0.233 req_src_port=5060 in_uri=sip:80000002@192.168.0.252;user=phone out_uri=sip:80000000@192.168.0.191:5060 via_cnt==1" F8: INVITE sip:80000000@192.168.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.233;branch=z9hG4bK475fff37270b762d From: "60000253" <sip:60000253@192.168.0.252;user=phone>;tag=58a90f4229e70a80 To: <sip:80000000@192.168.0.191:5060> Contact: <sip:60000253@192.168.0.233;user=phone> Supported: replaces Call-ID: 2a79d9ac84e2f72e(a)192.168.0.233 CSeq: 62121 INVITE User-Agent: Grandstream HT487 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 330 v=0 o=60000253 0 8000 IN IP4 192.168.0.233 s=SIP Call c=IN IP4 192.168.0.233 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=20 a=ptime:20 In fact, I think SER should retrieve user portion of header "to" URI of Invite message(for above example it should be 80000000 for first call,whereas 80000002 for second call) to construct userinfo of destination redirect uri for header contact of 3xx response(but hostinfo maybe get from SER config file). Meanwhile I study into sl_send_reply source code, but no progress to issue. So any suggestions or solutions are appreciated!! Thanks in advance!! George Lee(ShenZhen, CHINA)