Hi Antonis,
I guess you should move the enagaing of the rtpengine (for the INVITE
time) in the branch route, so it can do different SDP settings according
to the destination of that branch.
How you do it now will set the SDP in INVITE only in relation to the
first branch (so WAN) and reuse it for the next branches too.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 9/27/21 3:48 AM, Antonis Psaras wrote:
Hello Team
I have an OpenSIPs with multi home (WAN / LAN) connected to an
Asterisk (LAN). The problem I have is the following.
A call is coming from Asterisk to OpenSIPs LAN interface for a user
registered on OpenSIPs. That user has call forwarding enabled and
OpenSIPs receives a 302.
That request is handled as follows
failure_route[failure]
{
if (t_check_status("(301)|(302)"))
{
get_redirects("1:1");
uac_replace_from("","$tu");
uac_replace_to("","$ru");
if (!ds_select_dst("1", "0"))
{
send_reply("500","Unable to route");
exit;
}
t_relay();
}
rtpengine_delete();
}
A new INVITE is generated from OpenSIPs towards Asterisk but SDP
negotiated is the initial (OpenSIPs to Client) with the WAN IP.
Is there a way to correct that?
Thank you in advance for your support.
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