Couple of things to try....

1 - Check out xlog() so you can print out the URI, Contact, etc.

2 - Are you beheind NAT?

3 - This is all you should be concerend about... What is the last log message you are getting from here? Then use xlog() to tell you what you need to know. If you are not even getting the first log message then SER is not aware that it is supposed to respond for whatever the REGISTER has requested. You will need to add it as an "alias" global config option at the top of the file.

    if (uri==myself) {

            log(1,"in the served domain");
            if (method=="REGISTER") {
                    log(1,"do the register work");

#Uncomment this if you want to use digest authentication if (!www_authorize("Ip address of sip server", "subscriber")) { www_challenge("Ip address of sip server", "0"); log(1,"do www_challenge"); break; };

                    save("location");
                    log(1,"save in the location");
                    break;
            };

            # native SIP destinations are handled using 

our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1,"not found in the location"); break; }; }else{ log(1,"not in the domain"); };

----------------------------------------

Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike@bwsys.net Customer Service: (877) 976-0711

-----Original Message----- From: serusers-bounces@lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of yilan@cs.dal.ca Sent: Monday, September 27, 2004 7:19 PM To: serusers@lists.iptel.org Subject: [Serusers] Ser Register Problem

Dear All,

I have the latest ser (0.8.14) on fedora 2 as the sip server ,1 Mitel sip phone and xpro soft phone as the sip phones. The problem is that either the mitel or xpro sip phones can't register in ser.

I used "serctl ul show",nothing is registered.

I used "sipsak -vv -s sip:6000@ip address of sipserver",it shows "SIP/2.0 404 Not Found"

Then I added some log functions in ser.cfg where route locates (listed below).I found out when I used "sipsak -vv -s sip:6000@ip address of sipserver",the log functions execute first at uri==myself, then (!lookup("location")) and I can see the logs in the terminal.

But when I started the phone to communicate with the ser, those logs weren't show although I did see the register request from sip phones to ser from ngrep.

So my first question is when the ser.cfg is executed. When I used sipsak, I could see ser.cfg is executed. When I started the sip phone, I can't see its execution.

My second question is how to find out the reason for not registering. I used the IP address for the domain name. is it right to use? Because I did the test in local area network without dns.

route{

    log(1,"Begin route\n");
    if (!mf_process_maxfwd_header("10")) {
            sl_send_reply("483","Too Many Hops");
            log(1,"Too many hops\n");
            break;
    };
    if ( msg:len > max_len ) {
            sl_send_reply("513", "Message too big");
            log(1,"Message too big\n");
            break;
    };

    record_route();
    log(1,"record_route\n");
    # loose-route processing
    if (loose_route()) {
            t_relay();
            log(1,"loose route\n");
            break;
    };

    if (uri==myself) {

            log(1,"in the served domain");
            if (method=="REGISTER") {
                    log(1,"do the register work");

#Uncomment this if you want to use digest authentication if (!www_authorize("Ip address of sip server", "subscriber")) { www_challenge("Ip address of sip server", "0"); log(1,"do www_challenge"); break; };

                    save("location");
                    log(1,"save in the location");
                    break;
            };

            # native SIP destinations are handled using 

our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); log(1,"not found in the location"); break; }; }else{ log(1,"not in the domain"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); };

}

Thanks, Yilan

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

The RFC's and experience have been worth way more to me than any book. The RFC's are not a good place to start but good for finding the "official" answer. Try www.voip-info.com and the www.iptel.org site has a good SIP primer too. The rest of your learning will come from ngrep :)

----------------------------------------

Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike@bwsys.net Customer Service: (877) 976-0711

-----Original Message----- From: serusers-bounces@lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Kiko Vives Sent: Tuesday, September 28, 2004 1:24 AM To: serusers@lists.iptel.org Subject: [Serusers] SIP Books

Hi all.

I would like to buy a good book about SIP.

I have read in this list that "Internet Communication Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol" by Henry Sinnreich and Alan B. Johnston is a really good one.

Does anyone know another better ?

Thanks in advance.

Kiko

PS: please, someone answers my previous post which ask if with NATHELPER + RTPROXY is possible to establish videoconference sessions and not only audio sessions and if this is solved with Mediaproxy.

A bunch of thanks !

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers