Recent content by Instigata

Hi all. Clearly no expert :) I am trying to route SIP calls via Linphone/MicroSIP -> Opensips/RTPEngine -> Freeswitch -> PSTN. All webrtc calls work fine. SIP calls directly to PSTN also work fine. Phones registered directly on Freeswitch are ok too. However, SIP calls via Freeswitch to PSTN...

Hi all, I have a call originating from a browser (WebRTC) that's registered on a signaling server. The call is routed to Realtime API via Freeswitch. (All on AWS, so nightmare with private and public IP addresses). So we can't seem to get an audio. Don't want to make changes to the browser...

Thanks all. I was thinking along the lines of integrating with support tickets. @whut did you ever get to integrate with OS tickets? @Adrian Last did you finish up testing the scripts please?

Hi all. I was wondering if there's any modules available to enable OpenAi/GPT voice mode on freeswitch/fusionpbx. I only found this thread from a while ago: https://www.pbxforums.com/threads/fusionpbx-to-gpt4.6957/ It should be easier now. Apologies, if this has been discussed before, and...

From the extensive reading so far. I think I need to modify the Outbound Route and change one of the "action" tags to call ext 123 (and not ext. 500). Any ideas how? EDIT: Changed the bridge and callee_id details.

Dear all. I have an Opensips 3.0 + Fusion PBX setup. I would like calls received on Ext. 500 on Box A to be transferred to ext. 123 on Box B. I have successfully configured a trunk to Box B by setting up a Gateway and the status is REGED. I also added 123 in the Ext. field. In the dialplan...

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;

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; } #...

For item 1, I had to add the following: menu-exec-app:transfer *9196 XML XXX.XXX.XXX.XX

Hi all. I have setup Fusionpbx and working on the IVR. Two queries pls: 1. I would like one of the options in the IVR to transfer to one of the "Feature Codes" (*9196) on option 1 for example. I have added *9196 XML XXX.XXX.XXX.XX in front of Option 1. But I keep hearing invalid input error...

This seems to have been 2 issues. FusionPBX/Freeswitch would use the local IP (I believe "auto-nat" wasn't working), and also the opensips config wasn't correctly handling SDPs. Long and interesting journey.

Hi all. Lots of views, but no answers - I may have been wrong about this being an obvious problem. A little more context. The test I had done on the same instance was with Opensips + Freeswitch, following the OpensipsFreeswitch integration guide. I had to add opensips to the "public" profile...

Hi all, I had tested an Opensips (registrar) + Freeswitch (media) setup on an Amazon instance and all seemed to be working fine. However, for production I wish to have Opensips and Freeswitch on separate instances. I have managed to setup the ACLs/IPs correctly (famous last words). From the...

Herein lies the mystery - or did I misunderstand your comment :)

Hi all. Apologies if this is in the wrong place, been trying to integrate Opensips with Freeswitch by following this old guide: https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration I think I have managed to get the other stuff working alright. However, it seems that...