[OpenSIPS-Users] Try to setup ENUM NRENUM support...

[Stefano Pisani]

Ciao Michele,
per uscire con il contact corretto devi usare l'advertised_address / 
advertised_port se il serves ha solo un IP interno.
Questo fa nascere vari problemi perché la correzione viene fatta anche 
per verso i client interni.
Non c'è il concetto di Lan come per asterisk, almeno finora.
Nelle ultime versioni ho visto dei parametri in più quindi le cose 
potrebbero essere migliorate.

La miglior configurazione nel tuo caso sarebbe un server dual homed con 
IP interno ed esterno.

Ciao
s





Il 03/11/2015 10:43, Michele Pinassi ha scritto:
> Hi all,
>
> i'm trying to setup enum (NRENUM) infrastructure for our university. 
> We have a public IP voip server, VOIP01, with a private network for 
> voip phones (172.20.x.x). To let RTP flow through the private and 
> public network, i set up a RTP Proxy:
>
> /onreply_route[enum_answer] {//
> //    if(has_body("application/sdp")) {//
> //        rtpproxy_answer();//
> //    }//
> //}//
> //
> //onreply_route[enum_offer] {//
> //    if(has_body("application/sdp")) {//
> //        rtpproxy_offer();//
> //    }//
> //}//
> //
> //route[enum] {//
> //    xlog("L_INFO","Route to ENUM [$fd/$fu/$rd/$ru/$si/]\n");//
> ////
> //    if (is_method("INVITE")) {//
> //    if(has_body("application/sdp")) {//
> //            if (rtpproxy_offer()) {//
> //                t_on_reply("enum_answer");//
> //            }//
> //        } else {//
> //            t_on_reply("enum_offer");//
> //        }//
> //    }//
> //    if (is_method("ACK") && has_body("application/sdp")) {//
> //        rtpproxy_answer();//
> //    }//
> ////
> //    t_on_failure("pstn");//
> //
> //    if(!t_relay()) {//
> //    sl_reply_error();//
> //    }//
> //    exit;//
> //}/
>
>
> but on establishing call, this is the tcpdump trace between VOIP01 and 
> VOIP02 i get this:
>
> /IP VOIP01.5060 > VOIP02.5060: UDP, length 1170//
> //
> //INVITE sip:86472 at VOIP02 SIP/2.0//
> //Record-Route: <sip:VOIP01;lr;ftag=gujliebxxu;did=a26.6c5c0e13>//
> //Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
> //Via: SIP/2.0/UDP 
> 172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
> //From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
> //To: <sip:009123886472 at VOIP01:5060;user=phone>//
> //Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
> //CSeq: 2 INVITE//
> //Max-Forwards: 69//
> //User-Agent: snom760/8.7.5.17//
> //Contact: <sip:5002 at 172.20.1.47:57907>;reg-id=1//
> //X-Serialnumber: 00041371928A//
> //P-Key-Flags: resolution="31x13", keys="4"//
> //Accept: application/sdp//
> //Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
> PRACK, MESSAGE, INFO, UPDATE//
> //Allow-Events: talk, hold, refer, call-info//
> //Supported: timer, 100rel, replaces, from-change//
> //Session-Expires: 3600//
> //Min-SE: 90//
> //Content-Type: application/sdp//
> //Content-Length: 228//
> //
> //v=0//
> //o=root 846474428 846474428 IN IP4 172.20.1.47//
> //s=call//
> //c=IN IP4 VOIP01//
> //t=0 0//
> //m=audio 63194 RTP/AVP 9 0 8//
> //a=rtpmap:9 G722/8000//
> //a=rtpmap:0 PCMU/8000//
> //a=rtpmap:8 PCMA/8000//
> //a=ptime:20//
> //a=sendrecv//
> //a=nortpproxy:yes//
> //
> //IP VOIP02.5060 > VOIP01.5060: UDP, length 722//
> //
> //SIP/2.0 100 Trying//
> //Via: SIP/2.0/UDP 
> VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
> //Via: SIP/2.0/UDP 
> 172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
> //Record-Route: <sip:VOIP01;lr;ftag=gujliebxxu;did=a26.6c5c0e13>//
> //From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
> //To: <sip:009123886472 at VOIP01:5060;user=phone>//
> //Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
> //CSeq: 2 INVITE//
> //Server: FPBX-2.11.0(11.7.0)//
> //Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH//
> //Supported: replaces, timer//
> //Session-Expires: 1800;refresher=uas//
> //Contact: <sip:86472 at VOIP02:5060>//
> //Content-Length: 0//
> //
> //
> //IP VOIP02.5060 > VOIP01.5060: UDP, length 590//
> //
> //SIP/2.0 603 Declined//
> //Via: SIP/2.0/UDP 
> VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
> //Via: SIP/2.0/UDP 
> 172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//
> //From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
> //To: <sip:009123886472 at VOIP01:5060;user=phone>;tag=as15312d47//
> //Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
> //CSeq: 2 INVITE//
> //Server: FPBX-2.11.0(11.7.0)//
> //Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH//
> //Supported: replaces, timer//
> //Content-Length: 0//
> //
> //
> //IP VOIP01.5060 > VOIP02.5060: UDP, length 375//
> //
> //ACK sip:86472 at VOIP02 SIP/2.0//
> //Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
> //From: "Michele Pinassi" <sip:5002 at VOIP01:5060>;tag=gujliebxxu//
> //Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
> //To: <sip:009123886472 at VOIP01:5060;user=phone>;tag=as15312d47//
> //CSeq: 2 ACK//
> //Max-Forwards: 70//
> //User-Agent: VoIP Unisi.it//
> //Content-Length: 0/
>
> The main doubt is: /a=nortpproxy:yes/ ...why ?
>
> Thanks, Michele
>
>
>
> -- 
> Michele Pinassi
> Responsabile Telefonia di Ateneo
> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
> tel: 0577.(23)5000 -centralino at unisi.it
>
> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo,http://www.faq.unisi.it  
>
>
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