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"; #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> 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> > 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 >> OpenSIPS Summit 27-30 Sept 2022, Athens >> 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 >> >> _______________________________________________ >> Users mailing >> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >>
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