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I would imagine that you would copy your existing inbound route and then in the copy add a second condition of caller_id_number=^your_sim_number$ and then have it transfer to your extension. I would strongly advise testing this on a non production box. Inbound routes operate in the public...

Is this a newly installed system? or has it been recently updated? Which voice prompts are you using (/usr/share/freeswitch/sounds/en/us/callie/voicemail)?

I do not think there is, if qop (quality of protection), is omitted in http WWW-Authenticate then a browser will fall back to basic protection i.e. plain text passwords (probably OK on an https connection) But for SIP basic protection is not permitted, that is why qop, if missing, should default...

Noted, I can't remember where my original information came from, but I have found this: From RFC8760 (The Session Initiation Protocol (SIP) Digest Access Authentication Scheme) Section 2.6.8: A UAS MUST be able to properly handle a "qop" parameter received in an...

You probably need to double check this, but it is my belief that for SIP qop should always be present in WWW-Authenticate and Proxy-Authenticate headers. If for any reason it is missing qop="auth" should be assumed - Sorry can't remember the RFC right now.

Depending on configuration, recording file locations are normally controlled by the record_path channel variable. Have a look at the user_record dialplan in your dialplan manager. You will probably find the record path being set similar to this...

I tend to avoid the latest and the very new because I am allergic to pain!

Don't use an analogue trunk, it will be a backward step. Don't know about Canada, but they will be phased out in the UK by 2025. I have used Sangoma FXO / FXS cards in the past with DAHDI ,Asterisk and an analogue line - not something I would care to repeat!

It sounds like you are looking for more than the FusionPBX definition of an IVR. If I understand correctly you are wanting to build some sort of customer service application and as such would be beyond the scope of a simple answer in a forum thread. There are many programming languages that...

I'm sorry, but I do not understand your question. You can find documentation on IVRs here: https://docs.fusionpbx.com/en/latest/applications/ivr.html

Welcome, you will find many helpful people here.

The answer above looks good to me. Another pretty good tester site is https://regex101.com/ if you want to experiment. You may also find this post informative: https://www.pbxforums.com/threads/regex-for-one-single-outbound-route-uk.3819/

That would be a great day, if were to happen. Like DigitalDaz, I'm not knocking FusionPBX, it is very good, and I certainly would not be running the platform I have now without it. It does need fresh input, I feel the front end php code has become over complex and convoluted. Simplified code...

Something I forgot to add in my last post... Having an truly open source product with a good community is something I really care about. I have mentioned in other treads that we were badly burnt a few years ago (2016 I think) when the organisation decided to pull the plug on the closed source...

This is the content-length on the INVITE right? Then it is almost certainly the SDP body, look at the body do you see more codecs offered? If the answer is yes, then reduce the number offered, you could even try restricting it to PCMA or PCMU only. I'm only guessing, if you can post the...

I will be interested to see how you get on, I don't feel comfortable with FusionPBX any more. I would probably want to go Carrier + CC and the price is very good, Enterprise being no more than my current FusionPBX membership. However, VitalPBX looks a little too "closed source" to me. It was...

Sounds like you will need to do some logging on the Fusion and maybe some packet catures in order to understand exactly what is going on.

So what you are looking at are the various threads spawned by the FreeSWITCH main process (basically top -H). Freeswitch will spawn more threads as it handles more calls etc. So I don't think what you are seeing is abnormal. Look at the two screenshots below, the first from a relatively idle...

You can modify your *97 dialplan to add set statements for the channel variables:

There are channel variables that you can set to change what is announced in the voicemail application. It has been discussed on this forum several times. If I get time I will find the link(s) for you, but just search the forum, I'm sure you will find them yourself.