[Serusers] Missing something !!!!

[Ahmed Boreau]

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 at 10.0.0.242;ftag=1412313924;lr=on>
From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
CSeq: 47883 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7200 at 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 at 10.0.0.242;ftag=1412313924;lr=on>
From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
CSeq: 47883 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7200 at 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 at 10.0.0.242;ftag=1412313924;lr=on>
From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
CSeq: 47883 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7200 at 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 at 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 '240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0