New FreePBX Installation
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@eddiejennings what version of FreeFBX? Asterisk?
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@aaronstuder FreePBX 14 and Asterisk 13 I believe. Whatever the recommended thing was when I installed it.
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@eddiejennings Thanks! I’ll have go try again in the morning.
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@aaronstuder Assuming I have a stable Internet connection, I'll be checking ML throughout the day, so if you have problems, post here.
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@eddiejennings Will do, Thanks
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@aaronstuder said in New FreePBX Installation:
So I setup a fresh FreePBX 14 system.
Got Zoiper setup on Linux and iOS - Check
I am using Twilio so I got outgoing calling working - Check.
Incoming calling would not work...
Found this: http://hwdevelopment.com/blog/27-freepbx-13-asterisk-11-twilio-elastic-sip-trunk-setup
So I change the settings as referenced above and Twilio started to work for incoming calls.
However both Zoiper clients stopped working
So should I be using chan_pjsip or chan_sip for trunks? What about Phones, and Softphones? Both? On what port?
So should I be using chan_pjsip or chan_sip for trunks? What about Phones, and Softphones? Both? On what port?
I used AIX for everything, and didnt come close to any setting for SIP.
Just created AIX extension and used port 4569 I even changed that in the extension page to another port and it works, calls from LAN and calls from WAN.
In Zoiper there is an option to configure an IAX account, the reason I choose IAX cause it does everything in 1 port and options seemed more easier than SIP
But video didnt work, only audio + call recording .
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@emad-r in my limited experience in VoIP, I've only used PJSIP. I don't know enough about IAX to give you an assessment of the two.
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Trying again now...
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@Emad-R It is IAX2 not AIX.
IAX2 is Inter-Asterisk Exchange and is not a standard for everything. Very little support can be found for IAX2 in general.
IAX2 is nice because it only uses one port (UDP 4569) for signaling and audio, but again, not a heavily used protocol.
The SIP protocol as implemented in Asterisk was showing its age and limiting options in the modern world. That is why Asterisk developers came up with PJSIP, to extend funcitonality while also retaining basic compatibility.
Asterisk (not FreePBX) recommends that you should be using PJSIP by default now.
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@aaronstuder there is no significant difference between using SIP or PJSIP for a trunk connection. PJSIP attempts different signaling, but will still signal to match SIP if needed.
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I pulled a @jaredbusch and justed used voip.ms
