[OpenSER-Users] Privider SIP/PSTN

Marc LEURENT lftsy at free.fr
Tue Jul 31 11:50:12 CEST 2007


Can you help me, I received a SIP/2.0 475 Bad URI (475/SL) message and I don't understand
I have fallowed the tutorial you've sent me: http://docs.huihoo.com/openser/tutorials/uac/ar01s06.html correcting few commands like
t_relay_to_udp("GW_IP","GW_PORT"); --> t_relay("udp:GW_IP:GW_PORT");

You'll find below my conf and a ngrep command

Thanks


NB: I have changed id/password values

# -- uac params --
modparam("uac", "credential", "5674685998:saturne.alsion.com:JG6dzgyfd89F")
modparam("uac","from_restore_mode","auto")              # sequential requests and replies will be automatically updated based on stored original URI

...

if (method==INVITE && uri=~"0[0-9]*") {
                #uac_replace_from("sip:104 at sip.test.fr");
                # set failure route for authentication
                t_on_failure("3");
                # reset flag to mark no authentication yet performed
                resetflag(7);
                # forward to PSTN
                t_relay("udp:saturne.alsion.com:5060");
                exit;
        };
...

failure_route[3]
{
        # authentication reply received?
        if ( t_check_status("401|407") )
        {
                # have we already tried to authenticate?
                if (isflagset(7))
                {
                        t_reply("503","Authentication failed");
                        exit;
                }
                if (uac_auth())
                {
                        # mark that auth was performed
                        setflag(7);
                        # trigger again the failure route
                        t_on_failure("3");
                        # repeat the request with auth response this time
                        append_branch();
                        t_relay();
                }
        }
}





Here is a ngep of the communication (what is interesting is the )


interface: eth1 (192.168.95.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
#
U 192.168.95.70:5060 -> 192.168.95.248:5060
  INVITE sip:0677832974 at sip.wifirst.fr:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4696815358658109209..From: "Bob Wifirst"<sip:102@
  sip.wifirst.fr:5060;user=phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70
  ..CSeq: 1 INVITE..Max-Forwards: 70..Supported: timer, replaces..Session-Expires: 1800..Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPD
  ATE,REFER,REGISTER,INFO..Contact: <sip:102 at 192.168.95.70:5060;user=phone>..User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-C4..Content-Type: appl
  ication/sdp..Content-Length: 271....v=0..o=102 3355109 3355109 IN IP4 192.168.95.70..s=-..c=IN IP4 192.168.95.70..t=0 0..m=audio 41000 RTP/AVP 8 0 18
  4 97..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-15..a=sen
  drecv..
#
U 192.168.95.248:5060 -> 87.98.201.114:5060
  INVITE sip:0677832974 at sip.wifirst.fr:5060 SIP/2.0..Record-Route: <sip:192.168.95.248;lr=on;ftag=c0a80101-3331e4>..Via: SIP/2.0/UDP 192.168.95.248;bran
  ch=z9hG4bKe28.c363a7e.0..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4696815358658109209..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=p
  hone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 INVITE..Max-Forw
  ards: 10..Supported: timer, replaces..Session-Expires: 1800..Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO..C
  ontact: <sip:102 at 192.168.95.70:5060;user=phone>..User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-C4..Content-Type: application/sdp..Content-Lengt
  h: 271....v=0..o=102 3355109 3355109 IN IP4 192.168.95.70..s=-..c=IN IP4 192.168.95.70..t=0 0..m=audio 41000 RTP/AVP 8 0 18 4 97..a=rtpmap:8 PCMA/8000
  ..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a=rtpmap:4 G723/8000..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-15..a=sendrecv..
#
U 192.168.95.248:5060 -> 192.168.95.70:5060
  SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4696815358658109209..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=
  phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 INVITE..Server:
   OpenSER (1.2.1-tls (i386/linux))..Content-Length: 0....
#
U 87.98.201.114:5060 -> 192.168.95.248:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.95.248:5060;branch=z9hG4bKe28.c363a7e.0;received=192.168.95.248..Via: SIP/2.0/UDP 192.168.95.70:5060;bran
  ch=z9hG4bK4696815358658109209..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:506
  0;user=phone>..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  , SUBSCRIBE, NOTIFY..Contact: <sip:0677832974 at 87.98.201.114>..Content-Length: 0....
#
U 87.98.201.114:5060 -> 192.168.95.248:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.95.248:5060;branch=z9hG4bKe28.c363a7e.0;received=192.168.95.248..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z
  9hG4bK4696815358658109209..Record-Route: <sip:192.168.95.248:5060;lr=on;ftag=c0a80101-3331e4>..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=ph
  one>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>;tag=as62a15743..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 IN
  VITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:0677832974 at 87.98.201.114>..Content-T
  ype: application/sdp..Content-Length: 284....v=0..o=root 6041 6041 IN IP4 87.98.201.114..s=session..c=IN IP4 87.98.201.114..t=0 0..m=audio 12322 RTP/A
  VP 0 8 18 97..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-16
  ..a=silenceSupp:off - - - -..
#
U 192.168.95.248:5060 -> 192.168.95.70:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4696815358658109209..Record-Route: <sip:192.168.95.248:5060;lr=on;ftag=c0a80101-3331
  e4>..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>;tag=as62a157
  43..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE
  , NOTIFY..Contact: <sip:0677832974 at 87.98.201.114>..Content-Type: application/sdp..Content-Length: 284....v=0..o=root 6041 6041 IN IP4 87.98.201.114..s
  =session..c=IN IP4 87.98.201.114..t=0 0..m=audio 12322 RTP/AVP 0 8 18 97..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18
   annexb=no..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-16..a=silenceSupp:off - - - -..
#
U 192.168.95.70:5060 -> 192.168.95.248:5060
  ACK sip:0677832974 at 87.98.201.114 SIP/2.0..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4697925369758209219..From: "Bob Wifirst"<sip:102 at sip.wifir
  st.fr:5060;user=phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>;tag=as62a15743..Call-ID: 5d714ab-c0a80101-0-38 at 192.168
  .95.70..CSeq: 1 ACK..Max-Forwards: 70..Route: <sip:192.168.95.248:5060;lr=on;ftag=c0a80101-3331e4>..User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-
  AF-C4..Content-Length: 0....
#
U 192.168.95.248:5060 -> 87.98.201.114:5060
  ACK sip:0677832974 at 87.98.201.114 SIP/2.0..Record-Route: <sip:192.168.95.248;lr=on;ftag=c0a80101-3331e4>..Via: SIP/2.0/UDP 192.168.95.248;branch=z9hG4b
  Ke28.c363a7e.2..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK4697925369758209219..From: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=phone>;tag
  =c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>;tag=as62a15743..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 1 ACK..Max-F
  orwards: 10..User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-C4..Content-Length: 0..P-hint: rr-enforced....
#
U 87.98.201.114:5060 -> 192.168.95.248:5060
  CANCEL :0677832974 at 87.98.201.114 SIP/2.0..Via: SIP/2.0/UDP 87.98.201.114:5060;branch=z9hG4bK7781eedb;rport..Route: <sip:192.168.95.248:5060;lr=on;ftag
  =c0a80101-3331e4>..From: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>..To: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=phone>;tag=c0a80101-3331e
  4..Contact: <sip:0677832974 at 87.98.201.114>..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 101 CANCEL..User-Agent: Asterisk PBX..Max-Forwards: 70
  ..Content-Length: 0....
#
U 192.168.95.248:5060 -> 87.98.201.114:5060
  SIP/2.0 475 Bad URI (475/SL)..Via: SIP/2.0/UDP 87.98.201.114:5060;branch=z9hG4bK7781eedb;rport=5060..From: <sip:0677832974 at sip.wifirst.fr:5060;user=ph
  one>..To: "Bob Wifirst"<sip:102 at sip.wifirst.fr:5060;user=phone>;tag=c0a80101-3331e4..Call-ID: 5d714ab-c0a80101-0-38 at 192.168.95.70..CSeq: 101 CANCEL..S
  erver: OpenSER (1.2.1-tls (i386/linux))..Content-Length: 0....
#
U 192.168.95.70:5060 -> 192.168.95.248:5060
  BYE sip:0677832974 at 87.98.201.114 SIP/2.0..Via: SIP/2.0/UDP 192.168.95.70:5060;branch=z9hG4bK9842036470869314825..From: "Bob Wifirst"<sip:102 at sip.wifir
  st.fr:5060;user=phone>;tag=c0a80101-3331e4..To: <sip:0677832974 at sip.wifirst.fr:5060;user=phone>;tag=as62a15743..Call-ID: 5d714ab-c0a80101-0-38 at 192.168
  .95.70..CSeq: 2 BYE..Max-Forwards: 70..Route: <sip:192.168.95.248:5060;lr=on;ftag=c0a80101-3331e4>..User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-
  AF-C4..Content-Length: 0....
#













Taylor Carpenter a écrit :
> Use uac_auth().   You need modaram() for the credentials and an entry in
> failure route for auth messages.  Check out
> 
>     http://docs.huihoo.com/openser/tutorials/uac/ar01s06.html
> 
> besides the uac module page.
> 
> On Jul 30, 2007, at 11:06 AM, Marc LEURENT wrote:
> 
>> I have a SIP/PSTN provider with an account username/password.
>> The address of the provider is sip.test.com
>>
>> I would like to redirect phone calls to the gateway, but how is it
>> possible to log in on the gateway and forward calls?
>> Thank you for your replies
>>
>> Best Regards,
>>
>>
>> I've tried:
>>
>>  if (uri=~"0[0-9][0-9]*@*") {
>>                
>> rewritehostport("username:password at sip.test.com");             #
>> Rewrite the URI
>>                 route(1);
>>                 exit;
>> }
>>
>> _______________________________________________
>> Users mailing list
>> Users at openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
> 
> 
> _______________________________________________
> Users mailing list
> Users at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users




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