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Usually right near the top, order 10 or 15. It's set globally and in the domain to capture internal and external calls. https://github.com/fusionpbx/fusionpbx/blob/master/app/dialplans/resources/switch/conf/dialplan/010_caller_details.xml...

Forgive me if I misunderstood, but why can't you use the existing 'external' profile for this 'SIP Trunk'? The internal and external don't refer to IP networking, this is about which context the INVITES will be in. External is for trunks who are authenticated via IP ACL, Internal is for...

The whole page probably needs rewriting if it's Ubuntu.

No problem, I will say I don't know exactly if that's what the SIP History is for, so it might get weird in the future. I really think it should be sip_to_user. Oh well! If it works, it works.

So you want to make sure your inbound routes make use of the ${sip_to_user} field to route to the extensions then. Some providers put it in the VIA, some in the SIP To.

It's the variable: ${sip_history_info} However, from reading about it, it seems to be used mainly in LUA scripts to format what the diversion history looks like. Are you sure that the called number is not in the SIP To header?

Try TCP. A lot of ISPs mess with UDP 5060. Alternatively: Try setting up TLS TCP. Encryption means no tampering with SIP packets! OpenVPN to the server Duplicating the internal profile to run on a different port altogether.

Meticulously go through the scripts, and duplicate what's there, swapping out CentOS specific alternatives where the packages aren't available, or file locations are slightly different. Mark is updating it, so there is some activity. I'm sure he would be very happy of some pull requests adding...

Reload XML from the SIP Status page usually, though 'it depends'. Call centre stuff might need the module restarting? Make sure you are backing up each time!

Long and short of it is your SIP provider. ${sip_req_user} is useless for me, because our SIP trunks send INVITES with the URI in the form of: INVITE sip:gw+aa21fa11-1803-475f-ad01-8e28312b44d0@[PBXIPAddress]:5080;transport=udp;gw=aa21fa11-1803-475f-ad01-8e28312b44d0 SIP/2.0 Whereas the...

Github usually. Mark J Crane submitted a change 11 days ago adding {sip_req_user} to the master branch for inbound routes: https://github.com/fusionpbx/fusionpbx/commit/cb96c4a848dfb8766985b6aab46fcb8c97f87be7 And his comment appears to match your problem:

When you create a destination, it uses the SIP TO User as the destination condition. Now if you create a destination for 0564xxx and point it to where you need to, I believe it might work.

Ah, right. Yeh, I forget what it's like on out of the box installs but I have to ensure that the SIP TO User is treated as the destination number. I've got this default setting which I believe is used for new inbound routes. I can't remember if this is changed either, but the caller-details...

I would personally recommend building a new FusionPBX, backing up the database and restoring it on a new one. I would say 'update FreeSWITCH' but it's far too cumbersome these days as it builds from source and you have to match exactly what the installer does. I'm seeing good stability on...

They just probably added inbound destinations to the appropriate places, or have misunderstood the requirements?

The agent_status index.lua script doesn't appear to have the functionality to toggle the On Break status. I have put in a feature request myself as I may be doing a big deployment of the call centre soon, and I would ideally like to have this. My LUA skills are terrible with freeswitch however.

I'd say stick a call flow in front of extension 100 and 101 surely? +441212121212 ---> Call Flow 1 --> Extension 100 --> Call Flow 2 --> Destination 1 | | | ---->...

One thing I mucked up recently was defining the autoprovisioning URL from FusionPBX....but incorrectly. It doesn't do any transformations on the field at all, so you cannot use it to dynamically set the URL. Provisions, then never again. One thing I would do: 1. Packet capture from yealink...

I have to agree here. Really the only one on the page should be Debian, and then a link to the github to show the 'unsupported' alternative options. FreeSWITCH used to be a lot more open about other platforms and the Fusion side is 'just' PHP so theoretically good for almost any distro. Debian...

You talking about chaining them? Inbound Call ----> Call Flow 1 -----> Call Flow 2 --> Destination 2 | | V V Destination 1 Voicemail