Hi Nitesh,
Where you have the loose_route() in your script right now, use
topology_hiding_match()
https://opensips.org/html/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
On 10/27/22 3:51 PM, Nitesh Divecha wrote:
Hey Bogdan-Andrei,
Thanks for your feedback.
To be honest - I don't know!
I followed some online tutorials and tried to implement
"topology_hiding()" after is_method("INVITE") since the first message
sent to Outbound Provider is INVITE.
I never tried topology_hiding_match().
What is the right way to implement topology_hiding module? My scenario
is simple, I'm using the default cfg and making ATA register to
OpenSIPS to make Outbound PSTN calls.
I guess Bootcamp is the answer :-)
Cheers,
Nitesh
On Thu, Oct 27, 2022 at 4:02 AM Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
Hi Nitesh,
Do you do topology_hiding_match() for the sequential requests
(instead of the typical loose_route()) ? As it seems you cfg
fails to properly handle the in-dialog / sequential requests.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com <https://www.opensips-solutions.com>
OpenSIPS Bootcamp 5-16 Dec 2022, online
https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
<https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/>
On 10/26/22 9:05 PM, Nitesh Divecha wrote:
Hello All,
I don't know if this is by design or me not implementing
correctly! I'm a newbie to OpenSIPS...
So I generated a new opensips_residential.cfg file and only
edited it with topology_hiding("UC"); under INVITE. Plus all the
extras to make ATA register successfully and SIP trunk.
Every time I make a call two INVITE dialogs are created. One from
ATA to OpenSIPS and second one from OpenSIPS to Outbound
Provider. When a PSTN call is answered, the Outbound Provider
sends 200 OK back to OpenSIPS, and OpenSIPS sends back 200 OK to
ATA. Then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS
fails to send ACK to the Outbound Provider. So Outbound Provider
sends 200 OK again to OpenSIPS and OpenSIPS sends 200 OK to ATA
then ATA acknowledges with ACK back to OpenSIPS BUT OpenSIPS
fails to send back ACK... This dance goes on for 30 secs and
until Outbound Provider drops the call due to no ACK.
On the sngrep it shows that call from ATA to OpenSIPS "COMPLETED"
while OpenSIPS to Outbound Provider "CALL SETUP".
If I comment out topology_hiding("UC"); then everything works
perfectly and sngrep shows only one INVITE dialog from ATA to
OpenSIPS to Outbound Provider except the Outbound Provider can
see everything that is ATA's IP info, etc...
Here is my cfg:
route {
...
if (dp_translate(10 ,$rU ,$rU) ) {
xlog("*** Dial plan translate from source $avp(src)
to $rU ***\n");
$avp(furi) = "sip:xxxxxxx...@gothamcity.com
<mailto:sip%3axxxxxxx...@gothamcity.com>";
uac_replace_from( , "$avp(furi)");
#strip(1);
if (!do_routing(0)) {
send_reply(500,"No PSTN Route found");
exit;
}
# t_on_branch("change_from");
route(relay);
exit;
}
...
}
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
# create_dialog();
topology_hiding("UC");
if(remove_hf("User-Agent")){
xlog("*** 4. User-Agent found and removed.
***\n");
}
if (isflagset("NAT") && has_body("application/sdp")) {
rtpproxy_offer("ro");
}
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}
if (isflagset("NAT")) {
add_rr_param(";nat=yes");
}
if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}
My eyes are sore and my head is spinning... Any help will be
highly appreciated... Thanks!
Cheers,
Nitesh
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