ReInvites like all in-dialog requests are processed in has_totag()/loose_route part. Default kamailio.cfg has almost identical part for this purpose.
Main idea is that an initial request (non in-dialog) hasn't To tag already. It appears only when your request "hits" remote side (called or callee, as you wish) and goes back to you. Simplified, but I hope you got an idea :)
--
sent from myMail for Android
Hi Alex,
Thank you for your reply.
Does below config forward all sip requests coming from the customer to the supplier , and relay sip replies back to customer ?
Where re-invite packets are handled in the below config file?
Thank you
Ali Taher
-----Original Message-----
From: sr-users <sr-users-bounces@lists.kamailio.org> On Behalf Of Alex Balashov
Sent: Tuesday, July 17, 2018 4:25 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Cc: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] SIP re-invite Packets
On Tue, Jul 17, 2018 at 04:21:25PM +0300, Ali Taher wrote:
> Can you please send me a simplified config that suit my case ?
Try this for a main request route:
---
route {
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
if(t_check_trans())
t_relay();
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
# CANCELs / branches.
if(is_method("CANCEL")) {
if(!t_relay_cancel()) {
sl_send_reply("500", "Server Internal Error");
exit;
}
exit;
}
t_check_trans();
if(is_method("OPTIONS")) {
sl_send_reply("200", "OK");
exit;
}
else if(is_method("INVITE")) {
# Route your calls.
exit;
}
sl_send_reply("403", "Method unsupported");
exit;
}
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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