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    Call forward to another server/extension.

    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.
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    Call forward to another server/extension.

    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...
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    MicroSip and Linphone different behaviour - Opensips + FusionPBX

    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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    MicroSip and Linphone different behaviour - Opensips + FusionPBX

    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; } #...
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    IVR to Feature Code + Timeout Action

    For item 1, I had to add the following: menu-exec-app:transfer *9196 XML XXX.XXX.XXX.XX
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    IVR to Feature Code + Timeout Action

    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...
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    Opensips + FusionPBX audio sent to wrong IP and wrong port (192.168.1.7:7078)

    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.
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    Opensips + FusionPBX audio sent to wrong IP and wrong port (192.168.1.7:7078)

    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...
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    Opensips + FusionPBX audio sent to wrong IP and wrong port (192.168.1.7:7078)

    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...
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    Opensips 3.0 + Freeswitch Integration

    Herein lies the mystery - or did I misunderstand your comment :)
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    Opensips 3.0 + Freeswitch Integration

    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...
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    Installing OpenSIPS & Kamailio is....

    Hi again Rob. Thanks. This is a test config file - we are not using MS-Teams at all. If you can fix for Kamailio, we can try and modify for Opensips. Thanks again.
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    Installing OpenSIPS & Kamailio is....

    Thanks for the offer. I have setup RTPEngine. However, I am getting no-audio and one-way audio calls. Attached is the opensips.cfg - let me know what I have done wrong pls :) Thanks for your help.
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    Installing OpenSIPS & Kamailio is....

    Alright - thanks a lot. Would anyone have a working config file for Opensips with RTPEngine please? Thanks again.
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    Installing OpenSIPS & Kamailio is....

    Hi all. First post here. Trying to setup an RTP Proxy and kind of in the same boat. I have used the powerpbx guide to setup Opensips on Amazon EC2, 2 questions: 1. Is it necessary to setup RTPEngine first? 2. Any suggested configuration for OpenSips to work with RTP? Best wishes and thanks...