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I noticed that this client is using INFO for DTMF. You may want to add something like this to your dialplan in order for DTMF to be processed. <condition field="${sip_user_agent}" expression="SIP.js"> <action application="set" data="dtmf_type=info"/> </condition>

When using JS SIP your "phone" is your browser. That being the case Chrome, Firefox, etc would need to support G729 or GSM. This is a useful article that shows the codecs that different browsers support.

On the disa dialplan set add this: Action, Set, caller_id_number=xxxxxxxxxx

FS db in BDR cluster with 300 users should be fine. If you are using UDP for SIP I would set my registration expires to something like 120 sec or higher and use keepalives. You've got a environment stood up. Run some tests with SIPP or StarTrinity SIP Tester. Something to keep in mind...even...

Probably because of scaling issues. BDR is slow compared to something like sqlite in a RAM drive.

Sorry this is short b/c I'm running late, but I believe it has to do with the context. Check the settings of the sofia profile to which you are connecting.

I was being a little snarky when I wrote that post. CPU is typically not a bottleneck. My previous posts encouraged people to use load testing tools. Test all that stuff you mentioned.

Depending on the day of week, mood, alcohol caffeine levels, and direction of the wind, I'll either use a db query like the one above or use the Bulk Account Settings app for changes like this. My wife won't appreciate this antidote so I'll share it here. I was feeling frisky yesterday so I...

Interesting. It still works for me. I'm on a newish version of Master branch. Perhaps grab the query at the "SELECT call_recording_uuid, call_recording_path, call_recording_name..." query and try that directly on the database.

Real world: I have run 6500+ registrations on one server without issue using around 5% CPU. I still think you are asking the wrong questions.

Guys. It's time to put on your VoIP Engineer pants. There is a reason that every time the "how many ______ can my system handle" question gets asks that you don't get a specific answer. Here's where I would start... Each sofia sip profile is single threaded. That's the weakest link. To me I...

We have a FusionPBX environment with 20,000+ extensions. If you are facilitating registered endpoints (phones) your number one bottleneck is mod_sofia. All I can say is get to know sipp (the testing app) so you can test the limits of sofia's handling of SIP packets. You can grow your environment...

I found this post. I'm curious if this would work for a BDR replacement. http://eradman.com/posts/active-active-pglogical.html

You can stick something like this in your dialplan. <condition field="${call_direction}" expression="^inbound$"> <action application="set" data="sip_h_Alert-Info=<http://127.0.0.1/Bellcore-dr2>"/> </condition>

I used that powerpbx guide as well for OpenSIPS, however I went with Debian 9 b/c of issues I ran into with OpenSSL. I need to go back and see if Debian 10 will work with my updated config, but I know that 9 works. I have Team working with OpenSIPS & Fusion and didn't need RTPengine. It...

I figured out the issue with Teams transfer and hold/resume issue. This discussion provided some guidance. Ultimately I had to strip out the REPLY method from the Allowed Methods fields in all SIP messages coming from FreeSWITCH. For simplicity I just matched what Teams was sending me...

Sorry, I pulled a bunch of stuff off my repo due to security concerns raised about some of the code. Put the lua script in scripts/emergency_notify Put the tpl files in scripts/emergency_notify/resources/templates/en/us Here's the LUA: index.lua -- Intructions: -- Simply add an action...

@gflow, I was able to get it working the same as you with OpenSIPS. I ran into the same issues with putting calls on hold and resuming from Teams. It works intermittently. Sometimes you are able to resume right away and other times it might take 30 seconds to resume after you push the button...

4.4 has token security. Perhaps comment out the lines in https://github.com/konradSC/Fusionpbx-Public/blob/master/Apps/webphone/webphone.php with the word "token" to make it 4.4 compatible. <Disclaimer>Master branch (4.5) has a lot of security improvements so from my understanding that is the...

I've only tested in Master, but perhaps it might work in 4.4.