MicroSip and Linphone different behaviour - Opensips + FusionPBX

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Instigata

New Member
Nov 13, 2020
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Hi all,

I had setup an Opensips instance (as Registrar) on an Amazon EC2 instance and FusionPBX (media server) on another instance. I had tested it thoroughly and it seems to work great on the following code (Opensips.cfg):

if (!is_method("INVITE")) {
return;
}
# if the called number begins with "star" (*) then strip it and redirect to freeswitch
# (if it begins with two stars, eg: **, then one will be passed to FS)
if ($rU=~"^\*") {
strip(1);
$du = "sip:Fus.PBX.Int.IP:5090";
route(1);
}
}

I dial the IVR number - everything works ok for MicroSIP (laptop) and Linphone (Android phone).

However, in the following snippet - MicroSIP (call disconnects immediately) and Linphone has no audio (disconnects after 32 seconds). Pcap shows no connection on MicroSIP but on Linphone I can play the audio stream.


if (t_check_status("408|486")) {
$du = "sip:Fus.PBX.Int.IP:5090";
# do not set the missed call flag again
route(1);
}

Appreciate if someone can give me some pointers. Following are Fusionpbx settings:

Vars:
Ext-sip-ip = my public IP
Ext-rtp-ip = my public IP
Internal_sip_port: 5090

Internal SIP profile:
apply-nat-acl: nat.auto

ext-rtp-ip: $${local_ip_v4}
ext-sip-ip: $${local_ip_v4}

local-network-acl: localnet.auto
rtp-ip: $${local_ip_v4}
sip-ip: $${local_ip_v4}

Please let me know if any other information is required.

Thanks again.
 
Last edited:

Instigata

New Member
Nov 13, 2020
15
1
3
48
This was fixed for cell phone connections - (still not working on Wi-Fi) by the following code in failure_route:

rewritehostport("Fus.PBX.Int.IP:5090");
t_relay();
exit;
 
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