http://www.opensips.org/html/docs/modules/devel/rr.html#id293868

On 08/25/2015 10:20 AM, Nabeel wrote:
Please show me an example of where / how to use record_route_preset() to add the FQDN.

On 25 August 2015 at 16:54, Bogdan-Andrei Iancu <bog...@opensips.org <mailto:bog...@opensips.org>> wrote:

    Hi,

    According to the RFC, in RR header can be IP or FQDN (any kind of
    SIP URI). Even more, the best practice is to actually use IPs in
    RR to be 100% sure that the following requests to hit exactly the
    same box (if using FQDN, subject to DNS resolving, a different IP
    may be lookup up later).

    If you really want to put an IP there, use the
    record_route_preset() function:
    http://www.opensips.org/html/docs/modules/1.11.x/rr.html#id293864

    Regards,

    Bogdan-Andrei Iancu
    OpenSIPS Founder and Developer
    http://www.opensips-solutions.com

    On 25.08.2015 16:47, Nabeel wrote:
    Currently, OpenSIPS is using the actual IP address in the
    record-route URI, but I believe my SIP client needs the domain
    name in the record-route instead.


    For example, it should be:

        Record-Route: <sip:sipdomain.com
        <http://sipdomain.com>;lr;nat=yes;did=29.3daff1f4>


    NOT:

        Record-Route: <sip:162.242.153.259;lr;nat=yes;did=29.3daff1f4>



    How can I make this change in the OpenSIPS config?

    This should solve the problem because in a working setup
    (different SIP server), the logs state /"Resolving host address
    'sipdomain.com <http://sipdomain.com>'"/ and the record route URI
    includes the domain name, but in the OpenSIPS setup the logs
    state /"Resolving host address '162.242.153.259'/ and the record
    route URI contains the IP address.


    On 24 August 2015 at 18:37, Nabeel <nabeelshik...@gmail.com
    <mailto:nabeelshik...@gmail.com>> wrote:

        Hi,

        I see the cause now on the UAC side; I know it seems simple
        to just add some DNS records to the server IP,  but I'm still
        pondering on the best way to solve this and where exactly to
        add the SRV records because:

        1) I already have the SRV records set up on the actual
        hostname / domain, hosted by a DNS service third party, which
        is easier for me to maintain.  However the UAC seems to be
        ignoring this.

        2) I have used the same UAC with another server and did not
        have to set up SRV on the actual server machine IP.

        I'm not sure if this has anything to do with the OpenSIPS
        config but I'll let you know if I solve it.

        On 24 Aug 2015 17:56, "Bogdan-Andrei Iancu"
        <bog...@opensips.org <mailto:bog...@opensips.org>> wrote:

            Hi ,

            So, is the problem solved (by your findings in the UAS
            side) ?

            Regards,

            Bogdan-Andrei Iancu
            OpenSIPS Founder and Developer
            http://www.opensips-solutions.com

            On 24.08.2015 18:25, Nabeel wrote:
            I just discovered that the SIP client logs show an error
            message only on the recipient side, not on the caller's
            side.  I missed this previously because the caller's
            side log does not show any error:

                java.lang.Exception: No DNS SRV or A results found
                for: 162.242.153.259  (IP address of OpenSIPS server).


            I have the SRV records set on the actual
            hostname/domain, but it seems to be looking for SRV at
            the actual IP address itself.

            On 21 August 2015 at 17:57, Nabeel
            <nabeelshik...@gmail.com
            <mailto:nabeelshik...@gmail.com>> wrote:

                The log doesn't show any errors when the Timeout
                occurs, it only shows this:

                    opensips[1842]: ACC: call missed:
                    
timestamp=1440174643;method=INVITE;from_tag=z9hG4bK04147190;to_tag=;call_id=424618310389@10.137.181.237
                    <mailto:424618310389@10.137.181.237>;code=408;reason=Request
Timeout

                This seems to occur sporadically; some calls connect
                without problem but others don't; so perhaps it is a
                genuine timeout... maybe it simply longer to connect
                on some calls?


                On 21 August 2015 at 17:46, Nabeel
                <nabeelshik...@gmail.com
                <mailto:nabeelshik...@gmail.com>> wrote:

                    Sorry to bring this up again, but I still get
                    the 408 Request Timeout on some calls.

                    Isn't there just a way to increase the request
                    timeout limit?

                    Here is the trace:

                    http://pastebin.com/jvCPGYDu

                    There is even an ACK in the trace after the
                    request timeout message, but the call doesn't
                    connect.

                    On 7 August 2015 at 18:10, Bogdan-Andrei Iancu
                    <bog...@opensips.org
                    <mailto:bog...@opensips.org>> wrote:

                        Indeed,

                        Bogdan-Andrei Iancu
                        OpenSIPS Founder and Developer
                        http://www.opensips-solutions.com

                        On 07.08.2015 20:08, Nabeel wrote:
                        You mean like this, right?

                        if (is_method("REGISTER"))

                        {
                        if ( 0 ) setflag(TCP_PERSISTENT);

                        setbflag(SIP_PING_FLAG);

                        if (!save("location"))
                        sl_reply_error();

                        exit;
                        }



                        On 7 August 2015 at 17:52, Bogdan-Andrei
                        Iancu <bog...@opensips.org
                        <mailto:bog...@opensips.org>> wrote:

                            Hi Nabeel,

                            Bogdan-Andrei Iancu
                            OpenSIPS Founder and Developer
                            http://www.opensips-solutions.com

                            On 07.08.2015 19:39, Nabeel wrote:
                            [........]
                            Bogdan,

                            Regarding UDP, I realised that the UDP
                            port could not be in LISTEN state and
                            this was probably preventing my server
                            from fully opening that port. Running
                            nmap on that port showed result
                            "open|filtered", unlike with TCP which
                            showed fully open.  I am not running
                            any firewalls on my server, so this
                            seems to be the default behaviour of
                            my network.
                            A bidirectional traffic through the NAT
                            will keep the NAT pinhole open, while a
                            unidirectional one may not. This is the
                            advantage of the SIP pinging versus
                            simple UDP pinging.

I would like to clarify one thing. You mentioned adding
                            setbflag(SIP_PING_FLAG) before doing
                            save(), but in my config file I don't
                            see save() anywhere, there is only
                            this line: "if (!save("location"))".
                            Where exactly do I add this line?
                            exactly.

                            Regards,
                            Bogdan








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