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I think your testing method is not right. You are worried about CPU, however you need to be looking at your SIP retransmissions. Those are very high. From my understanding each SIP Profile runs as a single mod_sofia thread. Slowly turn up your Calls Per Second rate in SIPP and look for...

Did you add the new stuff in Default Settings?

Here's the How To. https://docs.fusionpbx.com/en/latest/applications/voicemail_transcription.html?highlight=watson Note that some of the Default Settings changed slightly. "api_key" is now "watson_key" "transcription_server" is now "watson_url"

This is the first thing that came to mind... Start with a dialplan step that matches the Destination & Time Condition. If the conditions are true, then you run a lua script when the call is terminated. You would use the "api_hangup_hook" to hook the script to the end of the call...

https://github.com/fusionpbx/fusionpbx/pull/4404

I fixed the error by adding another gsub line in send_email.lua. We swap out a percent sign with an asterisk. if (transcription ~= nil) then transcription = transcription:gsub("%%", "*"); body = body:gsub("${message_text}"...

Needs to be committed before it gets added to fusion docs.

Makes sense. I'll try to replicate and come up with a work around. It might be a day or two though..

I saw that I gave @Dan credit. It was @bcmike. Nice work!!

It seems that narrowband works for G722 calls as well, so I'll probably leave this in my default settings for now... Sub: transcription_server type: text Value: https://stream.watsonplatform.net/speech-to-text/api/v1/recognize?model=en-US_NarrowbandModel

I can't take credit for the Azure stuff. I think I just wrote some bug fixes :) Good to know about the G722 vs G711. Let me see if we can include some codec detection so this doesn't break.

@Dan Nice Work!! This has been on my To Do list for a while :) I took the liberty and made some updates to your code. First, I had to use IBM's narrowband model. It could be related to using PCMU as the codec. That required me to set this in Default Settings... Sub: transcription_server...

ignoreip = 127.0.0.1 (space)10.10.10.10 1.1.1.1

You need a space in there and not just a carriage return. ignoreip = 127.0.0.1 10.10.10.10 1.1.1.1

From my research it seems that the best option right now is going to be IBM's transcription. That will probably require a bigger rewrite though as that API is more of 'dry cleaner' model. You drop off the clothes, get a ticket, and then come back with your ticket to pick them up. Right now the...

Yes. That is how I list them.

At this point we have almost 8,000 Destinations on our platform. It's easy to setup a destination that routes to an extension and then later delete the extension without updating the destination. The destination is now set to transfer to an invalid number. I wanted a way to audit our system to...

We are talking FreeSWITCH speed here...Using the file cache method, the dialplan is only pulled from the database when there are not dialplan xml files present on the file system. Go look in the file cache directory on your server. You'll see a bunch xml files there. Clear cache then look again...

Yes, they work fine. Nothing special to setup. Configure the SIP Domain, SIP User, SIP auth user, SIP auth password. Just like a phone.

Looks like IBM Watson might be a viable alternative. Their synchronous HTTP service can take up to 100 meg files. https://cloud.ibm.com/docs/services/speech-to-text?topic=speech-to-text-http The asynchronous interface can take up to 1 gig files. That wouldn't be as realtime as synchronous...