Re: [OpenSIPS-Users] OpenSIPS Control Panel

[Bogdan-Andrei Iancu]

Hi Nitesh,

Do uac_replace_from( , "$avp(furi)");

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
  https://www.opensips.org/events/Summit-2022Athens/

On 10/14/22 5:55 PM, Nitesh Divecha wrote:

Bogdan-Andrei,

So I gave it a try and got error:

Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]  0. /usr/local/etc/opensips/opensips_residential_2022-10-11.cfg Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003] CRITICAL:core:yyerror: parse error in /usr/local/etc/opensips/opensips_residential_2022-10-11.cfg:456:22-23: too few param> Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]         xlog("new branch at change_from $ru to $avp(furi)

Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: ");

Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]         if ($avp(furi) != NULL) { Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]                 uac_replace_from("$avp(furi)"); Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]                 ^~ Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003]         } Oct 14 14:42:22 ip-172-31-29-47 opensips[60994]: Oct 14 14:42:22 [61003] }

Here is my code:

### uac_registrant module
loadmodule "uac.so"
modparam("uac","restore_mode", "auto")  #auto


####### Routing Logic ########
# main request routing logic
route{
if (dp_translate(10 ,$rU ,$rU) ) {

                $avp(furi) = "sip:1xxxxxxx...@gothamcity.com <mailto:sip%3a1xxxxxxx...@gothamcity.com>";

                #strip(1);

                if (!do_routing(0)) {
                        send_reply(500,"No PSTN Route found");
                        exit;
                }

                t_on_branch("change_from");
                route(relay);
                exit;
        }
}

branch_route[change_from] {
        xlog("new branch at change_from $ru to $avp(furi)\n");
        if ($avp(furi) != NULL) {
                uac_replace_from("$avp(furi)");
        }
}

NOTE: This is my first time playing with C-style code... Trying to learn what branches do/return, etc...

Rest of the code is default residential cfg.

Cheers,
Nitesh

On Thu, Oct 13, 2022 at 10:19 AM Nitesh Divecha <aviator.nites...@gmail.com <mailto:aviator.nites...@gmail.com>> wrote:

    Bogdan-Andrei,

    Thanks for your feedback...

    Yes, the remote SIP server expects FROM HEADER (Calling Identity)
    in order to authenticate the caller to make outbound calls.

    Where in cfg do I implement uac_replace_from() ? Is it in Routing
    Logic or where modules are declared?

    Also what happens if multiple DID providers are implemented in
    future? How will it affect the cfg file? Can we just implement via
    Control Panel?

    Cheers,
    Nitesh





    On Mon, Oct 10, 2022 at 9:42 AM Bogdan-Andrei Iancu
    <bog...@opensips.org <mailto:bog...@opensips.org>> wrote:

        Hi Nitesh,

        In SIP, registration is done for receiving calls, it does not
        impact sending calls. So, define the remote server as GW in
        Dynamic Route and simply route the calls to it.  Note that
        maybe the remote server will expect you to use the as FROM hdr
        (calling identity) the AOR (SIP address) you are registering
        with, so maybe you should be an uac_replace_from() in cfg when
        sending to the GW.

        Best regards,

        Bogdan-Andrei Iancu

        OpenSIPS Founder and Developer
           https://www.opensips-solutions.com  
<https://www.opensips-solutions.com>
        OpenSIPS Summit 27-30 Sept 2022, Athens
           https://www.opensips.org/events/Summit-2022Athens/  
<https://www.opensips.org/events/Summit-2022Athens/>

        On 10/7/22 9:00 PM, Nitesh Divecha wrote:

        Hello All,

        Anyone using OpenSIPS CP 9.3.2? Need small help!

        I got OpenSIPS 3.3.1 running and I can make calls out to the
        gateway (SIP trunk) without any problems.

        My provider issued me a DID with user/pass and I was able to
        configure them under "UAC Registrant" and registered to a
        remote server.

        Question is - how can I route calls to "UAC Registrant"? From
        the "Dynamic Routing" menu I can only route calls to Gateway.
        How can I route calls to "UAC Registrant"?

        Any suggestions?

        Thank you in advance!

        Cheers,
        Nitesh

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