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 <[email protected] <mailto:[email protected]>> 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" <[email protected]
    <mailto:[email protected]>> 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 <[email protected]
        <mailto:[email protected]>> 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=;[email protected]
                <mailto:[email protected]>;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
            <[email protected]
            <mailto:[email protected]>> 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
                <[email protected] <mailto:[email protected]>> 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
                    <[email protected]
                    <mailto:[email protected]>> 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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