Hi Alistiar,
The BYE is a sequential/indialog request, so you need to check your
script on how these are handled. Usually they are routed via
record_route() (if the routing mechanism is used in your cfg). See
https://github.com/OpenSIPS/opensips/blob/3.6/etc/opensips.cfg#L110
IF you do not understand you cfg flow, maybe you can add this script
tracing function on top of your cfg, so you can see how the execution
goes step by step via the cfg:
https://www.opensips.org/Documentation/Script-CoreFunctions-3-6#script_trace
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com
On 30.01.2026 18:54, Alistair Cleminson wrote:
Thanks for the tip, it turns out I'd completely misunderstood the
process. This configuration doesn't generate the errors & passes calls:
if (is_method("INVITE")) {
$acc_extra(si) = $si;
$acc_extra(fU) = $fU;
$acc_extra(fd) = $fd;
$acc_extra(rU) = $rU;
$acc_extra(rd) = $rd;
do_accounting("log|db","cdr|missed|failed");
if (get_source_group($var(si))) {
if (!alias_db_lookup("dbaliases")) {
xlog("L_ERR", "LOG: DDI Alias not found,
time=[$Ts] from uri=[$fu] to uri=[$ru]");
sl_send_reply(404, "Not Found");
};
$avp(site)=$(ruri.user{s.substr,0,5});
route(ALLOW);
exit;
}
else {
log(1, "Permissions RECORD FAIL\n");
sl_send_reply(403, "Forbidden");
exit;
};
};
I'm still not passing BYE packets successfully though. I'll dig a
little more before I ask again.
thanks
a.
*From: *johan de clercq <[email protected]>
*Date: *Thursday, 29 January 2026 at 9:41 am
*To: *Alistair Cleminson <[email protected]>
*Subject: *Re: [OpenSIPS-Users] Problems passing BYE signal in v3.7
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maybe changing the 2 vars to avp's will fix the issue.
On 1/27/26 17:27, Alistair Cleminson wrote:
Hi,
I've nearly completed the process of upgrading our ancient v1.7
deployment to v3.6. The new deployment is installed on Ubuntu
24.04 from the OpenSIPS repos. These servers are our inbound
hand-off from our upstream supplier, they take a call, inspect the
called number element against the database, and then pass it on to
the target system with the internal reference or decline the call.
I've never been a programmer and am much more familiar with
Asterisk, having inherited the platform & 'maintained' these hosts
for about ten years I'm far from fluent in the configuration
syntax but also much less terrified than I was when I first opened
the opensips.cfg file. Having said that, I've updated most of the
configuration and database backend to successfully accept, process
& pass on a call to our core voice platform with 2-way audio.
Unfortunately, when a call is terminated by either the called or
calling party the BYE signal is not passed to the other party
leaving the call hanging until they also hang up or an
intermediate system decides the call is in a stale state & drops it.
From what I can tell, this is the v1.7 code which would capture
the session related traffic and associate a BYE with a particular
call.
I think the problem is related to my attempt to convert the
depreciated allow_source_address_group() function:
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