[Users] pstn-to-openser, 403 Forbiden

[Ion Minzu]

Hello ,

I have connected openser with pstn through cisco. when I make a call from voip network to pstn it's ok.
but from pstn to voip I have a problem:openser answers 403 forbiden.
in openser I do the authorisation on mysql, I have disabled authorisation on sip
gateway:

if (src_ip!=X.X.X.X) {
        if (!www_authorize("DOMAIN.COM","subscriber")) {
        www_challenge("DOMAIN.COM","0");
        exit;
        }
        };

What is the problem?

 X.X.X.X is cisco
 
U X.X.X.X:54177 -> 172.17.6.2:5060
  INVITE sip:820022 at 172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP
  X.X.X.X:5060..From: <sip:022250699 at X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022 at 172.1
  7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID:
  906DA628-6E4F11DB-9034EA4F-E981BA1F at X.X.X.X..Supported: timer,100rel..Min-SE:  1800..Cisco-Guid
  : 2422905184-1850675675-2419190351-3917593119..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBS
  CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:022250699 at X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: 116
  2983794..Contact: <sip:022250699 at X.X.X.X:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 235....v=
  0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP
  Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4
  X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:19 CN/8000..a=ptime:20..
#
U 172.17.6.2:5060 -> X.X.X.X:5060
  SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP  X.X.X.X:5060..From:
  <sip:022250699 at X.X.X.X>;tag=1A0FBC30-1472..To:
  <sip:820022 at 172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F at X.X.X.X..CSeq: 101 INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C
  ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy feedback tells:  pid=32240 req_src_ip=X.X.X.X req_src_port=54177 in_uri=sip:820022 at 172.17.6.2:5
  060 out_uri=sip:820022 at 172.17.6.2:5060 via_cnt==1"....
#
U X.X.X.X:54177 -> 172.17.6.2:5060
  ACK sip:820022 at 172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP
  X.X.X.X:5060..From: <sip:022250699 at X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022 at 172.17.6
  .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov
  2006 11:03:14 GMT..Call-ID:
  906DA628-6E4F11DB-9034EA4F-E981BA1F at X.X.X.X..Max-Forward
  s: 6..Content-Length: 0..CSeq: 101 ACK....


Best regards,
Ion Minzu,
Specialist Tehnologii Informationale,
Administrator de sistem al Centrului de certificare,
Administrator VoIP,
I.S."Centrul de Telecomunicatii Speciale",
tel:250-517 (office), 069501208 (mob), 382869185 (ICQ)
mailto:ion.minzu at cts.md