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Hi all, I have an IVR that executes some API requests. A couple of those may take 3 to 5 seconds to come back with a result (this is expected, not a bug - it is the backend system taking that long - it is is not ideal but it is what it is). I was wondering if there is a way to play music on...

So, I found a workable solution for me. I am posting it here in case it helps anyone else: cmd = 'curl -X POST \"https://myapiserver.com/api/data\" -H \"Content-Type: application/json\" -d \'' .. myJSON .. '\''; local handle = io.popen(cmd) local result = handle:read("*a")...

Hi all, I realize this is a pure freeswitch question but I have not been able to find a good forum for freeswitch. The SignalWire slack channel is just like shouting in the middle of the street hoping someone will look at you. Anyway, sorry, just venting! the actual question is: I am trying...

And it worked! Thanks again @DigitalDaz !

ahh! it can't get easier than that! thanks @DigitalDaz, I will give it a go. I have a follow up question but I will post it in a separate thread.

OK. So, based on this: netstat -an | grep 8021 your mod_event_socket does not seem to be running. I don't have a ton of experience but it seems related to a misconfiguration in your box (e.g. networking, hosts). Try following this thread...

Interesting. I would check if mod_event_socket is running (netstat -an | grep 8021). Also, try fs_cli with explicit IP: fs_cli 127.0.0.1 or fs_cli [your_IP].

Yes, that is good enough.

It seems freeswitch is down. Try: sudo systemctl status freeswitch.

Is freeswitch up (sudo systemctl status freeswitch) ? It should be listening on TCP 5060 and 5080.

Hi all, I have a project where I need my FusionPBX to use an Audiocodes SBC as a gateway for all calls (in/out). The question came when setting up the SIP Trunk against the Audiocodes. As far as a I know in FusionPBX and freeswitch world a SIP Trunk would be a Gateway. In FusionPBX, the...

Well... then something is not set up properly. I would go step by step (even checking the obvious) - Is the Voicemail module running? - Is Voicemail Mail Enabled set to YES in the extension? - Is Voicemail Mail To set to a valid email? - Are all parameters properly set up in the SMTP session of...

Hi all, Probably very basic question here but where can I find a ist of the variables I can use in the email templates? Thanks!

I have only set it up twice but in both cases I simply folllowed the documentation. I did not even need to reload anything. At the beginning I had issues with my SMTP provider but checking the errors on /tmp/mailer-app.log helped me figuring it out. I am using SMTP2GO.

Thanks @DigitalDaz!

Hi all, I was wondering if anyone is willing to give me some advice on resources for a small FusionPBX system I am to set up. I have a customer that needs to handle 25 non compressed concurrent calls. The calls would not be to internal extensions but to an IVR I am setting up for them. I am...

I just realized that if all you want is a delay you can also do: session:execute("playback", "silence_stream://1000"); or <action application="playback" data="silence_stream://1000" /> You would not get the tone, just a 1 second delay. It may fit your need.

Well, as usual while typing out all this I actually found the problem. I was editiig the wrong file. I was editing /usr/share/freeswitch/lang/es/es.xml but I (also?) had to edit /etc/freeswitch/languages/es/es.xml. <include> <language name="es" say-module="es"...

Hi all, I made some progress here: I found out that the languages section in /etc/freeswitch/freeswitch.xml was missing a reference to Spanish. So I added it: <!-- languages section --> <section name="languages" description="Language Management"> <X-PRE-PROCESS cmd="include"...

ahh I understand. How about if you playback an audio file with a tone recording instead of the tone? Something like: <action application="playback" data="fake_ringtone.wav" /> I know, it is not the most elegant or brainy solution but it is simple.