Hello, I have a problem that my calls get rejected by he pstn gateway as my gateway is expecting a DID number instead of a 4 digit extension as the 'from user". Let me explain  my situation:

1. I have set callfwd_busy in my user preference table. 2. In my failure route I check to see if callfwd_busy has been set, if it has then I execute the following script:

if (t_check_status("486|408")) {

$avp(dst_number) = $rU;                     if (avp_db_load("$avp(dst_number)", "$avp(callfwd_busy)")) {                         xlog("LOG: avp(callfwd_busy)=$avp(callfwd_busy), avp(dst_number)=$avp(dst_number)\n");                         }

xlog("LOG: Failure route with $rU $avp(dst_number) \n");                         $rU = $avp(callfwd_busy);                         $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);                          route(RELAY);                         exit;               }

When I call from extension 7004 to ext 7002 ( this has callfwd_busy set to a DID number) the trace looks as follows:

U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 54.200.xx.yy:5060;branch=z9hG4bKee75.ab791c5.0;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.219.63;branch=z9hG4bK209939211;rport=1025. From: "7004" sip:7004@ajfmc1.myDomain.net;tag=1830267367. To: sip:7002@ajfmc1.myDomain.net;tag=as17cd0b40. Call-ID: 876294364-5060-74@BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0.

U 172.10.30.15:5060 -> 172.10.30.15:5080 INVITE sip:18455981234@64.136.xx.yy:5060 SIP/2.0. Record-Route: sip:54.200.xx.xx;lr=on;ftag=1830267367;nat=yes. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" sip:7004@ajfmc1.myDomain.net;tag=1830267367. To: sip:7002@ajfmc1.myDomain.net. Call-ID: 876294364-5060-74@BJC.BGI.B.I. CSeq: 731 INVITE. Contact: "7004" sip:7004@71.252.xx.yy:1025. Max-Forwards: 16. User-Agent: Grandstream HT701 1.0.4.8. Privacy: none. P-Preferred-Identity: "7004" sip:7004@ajfmc1.myDomain.net. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length:   453. . v=0. o=7004 8000 8000 IN IP4 172.10.30.1554.200.xx.xx. s=SIP Call. c=IN IP4 172.10.30.1554.200.xx.xx. t=0 0. m=audio 6248262482 RTP/AVP 0 18 4 8 2 97 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16,32-36,54. a=nortpproxy:yes. a=nortpproxy:yes.

# U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 503 Unavailable. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" sip:7004@ajfmc1.myDomain.net;tag=1830267367. To: sip:7002@ajfmc1.myDomain.net;tag=as17cd0b40. Call-ID: 876294364-5060-74@BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0.

It looks like my gateway is expecting to see a DID number in the from header. Can someone let me know how I can dot this? Thank you.

Arun