[OpenSIPS-Users] Integration with Asterisk/Trixbox

James Lamanna jlamanna at gmail.com
Wed May 20 22:11:59 CEST 2009


On Wed, May 20, 2009 at 7:46 AM, Iñaki Baz Castillo <ibc at aliax.net> wrote:
> 2009/5/20 James Lamanna <jlamanna at gmail.com>:
>> Hi,
>> I want to use OpenSIPs as the registrar (and NAT handler) for an
>> Asterisk/Trixbox installation.
>> I've got things partially working, but I've totally made a mess of my
>> config (I can post it if you would like).
>>
>> Some things that I need:
>>
>> I'm having problems with SIP<->SIP calls because I need asterisk to
>> stay in the media stream, so really the call has to be routed like:
>>
>> phone1 <--> opensips <--> asterisk <--> opensips <--> phone2.
>>
>> Does anyone have any configs that come close to this that I could stare at?
>
> Set "canreinvite=no" for opensips peer in sip.conf.
>
>
>> The ones I've found on the web are useful in some ways, but not in others.
>
> This question is more related to Asterisk.

Not really.
I'm using the basic NAT example with a little rewriting (as shown below).
The problem I have is that the SDP address is not being rewritten, so
on the asterisk box I see audio traces like this:
(x.x.x.x is the phone NAT IP address, y.y.y.y is the asterisk box address).
You'll notice that the first 2 lines are correct, but the second 2 are
not (asterisk sends to the private IP of the phone)

13:04:54.228511 IP (tos 0x0, ttl 118, id 8879, offset 0, flags [none],
proto: UDP (17), length: 200) x.x.x.x.47127 > y.y.y.y.12232: UDP,
length 172
13:04:54.228555 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 200) y.y.y.y.10536 > x.x.x.x.47128: UDP,
length 172
13:04:54.243209 IP (tos 0x0, ttl  54, id 22237, offset 0, flags
[none], proto: UDP (17), length: 200) x.x.x.x.47128 > y.y.y.y.10536:
UDP, length 172
13:04:54.243254 IP (tos 0x0, ttl  64, id 0, offset 0, flags [DF],
proto: UDP (17), length: 200) y.y.y.y.12232 > 10.20.200.219.49154:
UDP, length 172

When asterisk sends an INVITE to connect the called phone, the SDP
isn't getting rewritten for some reason.
How do I make that work?

The setup signaling looks like:


Calling phone sends INVITE --> opensips forwards to asterisk -->
asterisk sends INVITE ---> opensips forwards INVITE to called phone...

-- James

debug=3         # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)

# Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
#debug=4

check_via=no    # (cmd. line: -v)
dns=no           # (cmd. line: -r)
rev_dns=no      # (cmd. line: -R)
port=5060
children=4

listen=udp:z.z.z.z:5060
# ------------------ module loading ----------------------------------

#set module path
#mpath="/usr/local/lib/opensips/modules/"
mpath="/usr/local/lib64/opensips/modules/"

# Uncomment this if you want to use SQL database
#loadmodule "db_mysql.so"

loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "signaling.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "xlog.so"

# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
#loadmodule "auth.so"
#loadmodule "auth_db.so"

# !! Nathelper
loadmodule "nathelper.so"


# ----------------- setting module-specific parameters ---------------

# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# -- usrloc params --
modparam("usrloc", "db_mode",   0)

# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)

# -- auth params --
# Uncomment if you are using auth module
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)

# !! Nathelper
modparam("usrloc","nat_bflag",6)
modparam("nathelper","sipping_bflag",8)
modparam("nathelper", "ping_nated_only", 1)   # Ping only clients behind NAT
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")

# -------------------------  request routing logic -------------------

# main routing logic

route{

^[
    xlog("L_INFO", "New request - Request/failure/branch routes: M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");

    # max_forwards==0, or excessively long requests
    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    };
    if (msg:len >=  2048 ) {
        sl_send_reply("513", "Message too big");
        exit;
    };

    if (is_method("OPTIONS")) {
        sl_send_reply("200", "OK");
        exit;
    }

    # !! Nathelper
    # Special handling for NATed clients; first, NAT test is
    # executed: it looks for via!=received and RFC1918 addresses
    # in Contact (may fail if line-folding is used); also,
    # the received test should, if completed, should check all
    # vias for rpesence of received
    if (nat_uac_test("3")) {
        # Allow RR-ed requests, as these may indicate that
        # a NAT-enabled proxy takes care of it; unless it is
        # a REGISTER

        if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
            xlog("L_INFO", "LOG:Someone trying to register from
private IP, rewriting\n");
            # This will work only for user agents that support symmetric
            # communication. We tested quite many of them and majority is
            # smart enough to be symmetric. In some phones it takes a
            # configuration option. With Cisco 7960, it is called
            # NAT_Enable=Yes, with kphone it is called "symmetric media" and
            # "symmetric signalling".

            # Rewrite contact with source IP of signalling
            fix_nated_contact();
            if ( is_method("INVITE") ) {
                xlog("L_INFO", "NAT: FIXING SDP");
                fix_nated_sdp("1"); # Add direction=active to SDP
            };
            };
            force_rport(); # Add rport parameter to topmost Via
            setbflag(6);    # Mark as NATed

            # if you want sip nat pinging
            # setbflag(8);
        };
    };

    # subsequent messages withing a dialog should take the
    # path determined by record-routing
    if (loose_route()) {
        # mark routing logic in request
        append_hf("P-hint: rr-enforced\r\n");
        route(1);
        exit;
    };

    # we record-route all messages -- to make sure that
    # subsequent messages will go through our proxy; that's
    # particularly good if upstream and downstream entities
    # use different transport protocol
    if (!is_method("REGISTER"))
        record_route();

    if (!uri==myself) {
        # mark routing logic in request
        append_hf("P-hint: outbound\r\n");
        route(1);
        exit;
    };

    # if the request is for other domain use UsrLoc
    # (in case, it does not work, use the following command
    # with proper names and addresses in it)
    if (uri==myself) {

        if (is_method("REGISTER")) {

            # Uncomment this if you want to use digest authentication
            #if (!www_authorize("siphub.org", "subscriber")) {
            #   www_challenge("siphub.org", "0");
            #   return;
            #};

            save("location");
            exit;
        };

        lookup("aliases");
        if (!uri==myself) {
            append_hf("P-hint: outbound alias\r\n");
            route(1);
            exit;
        };

        if (is_method("NOTIFY")) {
            route(2);
        }
        if(is_method("OPTIONS")) {
            # send reply for each options request
            sl_send_reply("200", "ok");
            exit;
        }


        if (from_uri =~ "sip:.*@y\.y\.y\..*") {
            log("request from asterisk\n");
            if (!lookup("location")) {
                sl_send_reply("404", "Not Found");
                exit;
            };
            #fix_nated_contact();
            #if (is_method("INVITE"))
            #   fix_nated_sdp("3");
            #force_rport(); # Add rport parameter to topmost Via
            #setbflag(6);    # Mark as NATed
        } else if (is_method("INVITE")) {
            #} else { #if (uri =~ "sip:1[0-9]{10,10}@z\.z\.z\.z") {
                rewritehostport("y.y.y.y:5061");
            #};
        }

        # native SIP destinations are handled using our USRLOC DB
        #if (!lookup("location")) {
        #   sl_send_reply("404", "Not Found");
        #   exit;
        #};
    };
    append_hf("P-hint: usrloc applied\r\n");

    route(1);
}

route[1]
{
    # !! Nathelper
    if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
        sl_send_reply("479", "We don't forward to private IP addresses");
        exit;
    };

    # if client or server know to be behind a NAT, enable relay
    if (isbflagset(6)) {
        force_rtp_proxy();
    };

    # NAT processing of replies; apply to all transactions (for example,
    # re-INVITEs from public to private UA are hard to identify as
    # NATed at the moment of request processing); look at replies
    t_on_reply("1");

    # send it out now; use stateful forwarding as it works reliably
    # even for UDP2TCP
    if (!t_relay()) {
        sl_reply_error();
    };
}

# !! Nathelper
onreply_route[1] {
    # NATed transaction ?
    if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
        fix_nated_contact();
        force_rtp_proxy();
    # otherwise, is it a transaction behind a NAT and we did not
    # know at time of request processing ? (RFC1918 contacts)
    } else if (nat_uac_test("1")) {
        fix_nated_contact();
    };
}

route[2] {
    if (!t_newtran()) {
        sl_reply_error();
        exit;
    };

    t_reply("200", "OK");
    exit;
}



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