[Serusers] Missing something !!!!

Hi, I set up my ser+asterisk in order to make it scalable suggested. The amazing thing is that when lookup(location) failed, call is forward to asterisk as I asked (see diagnostic below) but when I got a client is not responding, calls are not forwarding to asterisk. Where do I miss something ? Thanks in advance --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); rewritehostport("10.0.0.13:5070"); t_relay_to_udp("10.0.0.13","5070"); break; }; # THIS IS WORKING !!! IT'S FOR TEST PURPOSE }; if (!t_relay()) { sl_reply_error(); }; if (method=="INVITE"){ t_on_failure("1"); t_relay(); break; } } # THIS IS NOT WORKING AT ALL !!! route[1]{ if(uri=~"^sip:72[0-9]{2}@*"){ revert_uri(); rewritehostport("10.0.0.73:5070"); append_branch(); t_relay_to_udp("10.0.0.13","5070"); } } --------------------------------------------------- LOGS SHOW WHEN TRYING TO CALL USER 7200 WHO IS TURN OFF ---------------------------------------------------- Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0 Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B Record-Route: <sip:7200@10.0.0.242;ftag=1412313924;lr=on> From: Jean <sip:7201@10.0.0.242>;tag=1412313924 To: <sip:7200@10.0.0.242>;tag=as25d200ae Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B(a)10.0.0.155 CSeq: 47883 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:7200@10.0.0.13:5070> Content-Type: application/sdp Content-Length: 255 v=0 o=root 5586 5587 IN IP4 10.0.0.13 s=session c=IN IP4 10.0.0.13 t=0 0 m=audio 10602 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.0.0.242:5060 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0 Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B Record-Route: <sip:7200@10.0.0.242;ftag=1412313924;lr=on> From: Jean <sip:7201@10.0.0.242>;tag=1412313924 To: <sip:7200@10.0.0.242>;tag=as25d200ae Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B(a)10.0.0.155 CSeq: 47883 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:7200@10.0.0.13:5070> Content-Type: application/sdp Content-Length: 255 v=0 o=root 5586 5587 IN IP4 10.0.0.13 s=session c=IN IP4 10.0.0.13 t=0 0 m=audio 10602 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.0.0.242:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0 Via: SIP/2.0/UDP 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B Record-Route: <sip:7200@10.0.0.242;ftag=1412313924;lr=on> From: Jean <sip:7201@10.0.0.242>;tag=1412313924 To: <sip:7200@10.0.0.242>;tag=as25d200ae Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B(a)10.0.0.155 CSeq: 47883 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:7200@10.0.0.13:5070> Content-Type: application/sdp Content-Length: 255 v=0 o=root 5586 5587 IN IP4 10.0.0.13 s=session c=IN IP4 10.0.0.13 t=0 0 m=audio 10602 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.0.0.242:5060 Nov 29 15:22:39 WARNING[31452080]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 240C974F-FFF5-4EB7-B9D9-C8B69484018B(a)10.0.0.155 for seqno 47883 (Non-critical Response) Nov 29 15:22:43 NOTICE[125635504]: res_musiconhold.c:306 monmp3thread: Request to schedule in the past?!?! -- Playing 'vm-password' (language 'en') -- Incorrect password '' for user '7201' (context = <any>) -- Playing 'vm-incorrect-mailbox' (language 'en') -- Playing 'vm-password' (language 'en') -- Incorrect password '' for user '7201' (context = <any>) -- Playing 'vm-incorrect-mailbox' (language 'en') -- Playing 'vm-password' (language 'en') -- Incorrect password '' for user '7201' (context = <any>) -- Playing 'vm-incorrect' (language 'en') -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup("SIP/10.0.0.242-085da938", "") in new stack == Spawn extension (default, 7200, 4) exited non-zero on 'SIP/10.0.0.242-085da938' Destroying call &#39;240C974F-FFF5-4EB7-B9D9-C8B69484018B(a)10.0.0