Hello Guys,

Im trying to implement a system to manipulate DID's, the forward for
external address is ok, but in trying now to do the same with  a user that
is registered


i create a new table where i have the did, the account and the destination,
if the destination is null so the opensips will check the account on the
location table.

basically i have this

if(!$avp(91)){
                                        xlog("Did nao encontrado");
                                        sl_send_reply("404", "Not Found");
                                        exit;
                                }else{
                                        xlog("Did encontrado, seguindo
regras para utilizacao em location");
                                        $ru = "sip:" + $avp(91) +
"@IP_ADDRESS:5060";
                                        xlog("Novo destino $ru");
                                }


the avp(91) is the user account, the same that the user use to register


when in this situation, the call go trouhg the location module, and the
system find the correct address.

i have on the log the same information that i have on opensipsctl ul show

Fazendo relay <null> - sip:055011395010100000@IP_ADDR
:5081;rinstance=0f9054bc313f0cf1;transport=UDP


below the output from ul show

    AOR:: 055011395010100000
        Contact::
sip:055011395010100000@IP_ADDR:5081;rinstance=0f9054bc313f0cf1;transport=UDP
Q=
            Expires:: 525
            Callid:: MTZhNzE1ZDYzYWU4Y2ViZDMzZTQzZWU1N2M0ZGFiZjQ.
            Cseq:: 2
            User-agent:: Zoiper Communicator 2.04.10164 rev.10204
            State:: CS_SYNC
            Flags:: 0
            Cflag:: 0
            Socket:: udp:GW_IP_ADDR:5060
            Methods:: 5951



But when i make the call, the ngrep show me the send of the invite, but i
dont see nothing on the other side.


Below you have the invite

U GW_IP_ADDR:5060 -> CUSTOMER_IP_ADDR:5081
INVITE 
sip:055011395010100000@CUSTOMER_IP_ADDR:5081;rinstance=0f9054bc313f0cf1;transport=UDP
SIP/2.0.
Record-Route: <sip:GW_IP_ADDR;lr;ftag=as657116d5;did=5b6.6e3954b;nat=yes>.
Via: SIP/2.0/UDP GW_IP_ADDR:5060;branch=z9hG4bK485d.8be72863.0.
Via: SIP/2.0/UDP
CALLER_IP_ADDR:5060;received=CALLER_IP_ADDR;branch=z9hG4bK1f912c35;rport=5060.
Max-Forwards: 69.
From: "testemike" <sip:testemike@CALLER_IP_ADDR>;tag=as657116d5.
To: <sip:551133992377@GW_IP_ADDR>.
Contact: <sip:testemike@CALLER_IP_ADDR:5060>.
Call-ID: 40d32e5b3c52c58646d996d871ad8471@CALLER_IP_ADDR:5060.
CSeq: 102 INVITE.
User-Agent: SIP.Ultranet.
Date: Mon, 23 Sep 2013 00:20:27 GMT.
Session-Expires: 600.
Min-SE: 90.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 446.
.
v=0.
o=root 985357135 985357135 IN IP4 CALLER_IP_ADDR.
s=SDP.Ultranet.
c=IN IP4 GW_IP_ADDRGW_IP_ADDR.
t=0 0.
m=audio 4229242292 RTP/AVP 8 0 3 111 97 18 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:111 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.
a=nortpproxy:yes.


after the invite, i get a request timeout message

i try to create some firewall rules on the customer side but i dont see any
package, from the opensips is like the package is beeing sended

i made a try using another machine that dont have nat and i cant see the
package in this case too.



anyone have an idea about this ?

Thanks.
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