I dont know if any of you use the SIPBroker service to do multi enum lookups, but I'm hoping there are some on this list who do. I have been using SIPBroker for dialing with sip codes only so far and did my enum lookups on my own. But now, I decided to set ser up to try sipbroker first for all external calls and then fall back to my different gateways if no route can be found via SIPBroker. SIPBroker, in prinicpal, works like this. I send an Invite for any enum number to SIPBroker. If sipbroker finds a route for that number to another voipprovider it will proxy the call to the found provider. If sipbroker can not find a route, it will reply with a redirect of the call to my self and I am supposed to handle the call setup my self, through my gateways. My question is how to handle this redirect message? Any one who has a working failure route to handle this situation, and are willing to share? Here is an actual SIP conversation, initiated from Asterisk via SER, of a failed Enum lookup through SIPBroker: # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 INVITE sip:4640240252@sipbroker.com SIP/2.0. Record-Route: <sip:212.247.91.XXX;ftag=as5e811304;lr=on>. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>. Contact: <sip:330000@212.247.91.XXZ>. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sat, 29 Jul 2006 22:42:23 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 336. . v=0. o=root 850 851 IN IP4 212.247.91.XXZ. s=session. c=IN IP4 212.247.91.XXZ. t=0 0. m=audio 34852 RTP/AVP 0 8 18 97 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:97 iLBC/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. ## U 24.196.79.163:5060 -> 212.247.91.XXX:5060 SIP/2.0 300 Redirect. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. Contact: sip:4640240252@sip.serverhallen.com. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 24.196.79.163:5060 "Noisy feedback tells: pid=15326 req_src_ip=212.247.91.XXXreq_src_port=5060 in_uri=sip:4640240252@sipbroker.com out_uri=sip:4640240252@sip.serverhallen.com via_cnt==2". . # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 ACK sip:4640240252@sipbroker.com SIP/2.0. Via: SIP/2.0/UDP 212.247.91.237;branch=z9hG4bK7307.c7d88007.0. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. CSeq: 103 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/freebsd)). Content-Length: 0. .