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.
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: