[Serusers] question on Adding aliases

[Eric Haskins]

Ok I was able to get the inbound call to ring to user 5002  I had to add Provider IP to 

                        # authenticate calls
                        if(!www_authorize("voip.mydomain.com","subscriber")) {
                                # you didn't send me credentials Maybe I trust your IP?
                                if (src_ip=~"216.8.xx.*" || src_ip=~"24.206.xxx.*" || src_ip=~"69.28.xxx.*") {
                                        log(1,"Incoming call from trusted IP");
                                } else {
                                        # I don't trust this IP.. ask for credentials
                                        www_challenge("voip.mydomain.com", "0");
                                        break;
                                }
                        }

But once I answer 5002 it is dead air on the 5002 and the PSTN phone keeps ringing until call could not be completed.

No. Time        Source        Destination     Protocol Info
31  2.248714    216.8.xxx.xx  65.175.xxx.xxx  SIP      Status: 200 OK
120 10.499698   69.28.x.xxx   216.8.xxx.xx    SIP/SDP  Request: INVITE sip:16049094251 at 216.8.xxx.xx:5060,
125 10.513795   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 100 trying -- your call is important to us
126 10.513918   216.8.xxx.xx  65.175.xxx.xxx  SIP/SDP  Request: INVITE sip:5002 at 65.175.xxx.xxx:5060,
128 10.580401   65.175.xxx.xxx 216.8.xxx.xx   SIP      Status: 100 Trying
129 10.597017   65.175.xxx.xxx 216.8.xxx.xx   SIP      Status: 180 Ringing
130 10.597219   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 180 Ringing
186 15.502397   69.28.x.xxx   216.8.xxx.xx    SIP  Request: CANCEL sip:16049094251 at 216.8.xxx.xx:5060
187 15.502684   216.8.xxx.xx  65.175.xxx.xxx  SIP      Request: CANCEL sip:5002 at 65.175.xxx.xxx:5060
188 15.502851   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 200 canceling
190 15.569267   65.175.xxx.xxx 216.8.xxx.xx   SIP      Status: 487 Request Terminated
191 15.569386   216.8.xxx.xx  65.175.xxx.xxx  SIP      Request: ACK sip:5002 at 65.175.xxx.xxx:5060
192 15.569550   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 487 Request Terminated
193 15.573890   65.175.xxx.xxx 216.8.xxx.xx   SIP      Status: 200 OK
198 15.929421   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 487 Request Terminated
214 17.308191   65.175.xxx.xxx 216.8.xxx.xx   SIP      Request: NOTIFY sip:216.8.xxx.xx
215 17.308387   216.8.xxx.xx  65.175.xxx.xxx  SIP      Status: 200 OK
223 17.933116   216.8.xxx.xx  69.28.x.xxx     SIP      Status: 487 Request Terminated
397 32.367404   65.175.xxx.xxx 216.8.xxx.xx   SIP      Request: NOTIFY sip:216.8.xxx.xx
398 32.367590   216.8.xxx.xx  65.175.xxx.xxx  SIP      Status: 200 OK

65.175.* IP is my ATA Test Box user 5002
69.28.* is the DID Provider
216.8.* is the SER box

Considering this box was designed orignally for outbound only I am wondering if I have to send a response back to our DID provider to let them know it was answered??  I have searched for example configs utilizing in & outbound and have come up short

Thx again for any help

Eric
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