On outbound calls I can here the PSTN but not visa versa and call dropped after 32 second

Hello I am new to VoIP and SIP so I thought I would try to learn by doing. Unfortunately I’m ripping my hair out with trying to get this to work. I’ve looked at the forums from my sip provider(Twilio), my router(Unifi USG), and FreePBX I have tried absolutely everything I could find.

On internal calls there is no audio at all.

On outbound calls the soft phone(X-Lite) can hear the PSTN but the PSTN can’t here the soft phone. And on top of that the calls end after 30 seconds, my sip provider(Twilio) says that the “caller”(PBX or soft phone) is ending the call.

Any help is greatly appreciated and if you need more information please ask!

Most probably a NAT issue. Have you configured NAT parameters on FreePBX SIP settings? Have you correctly forwarded UDP and RTP ports on USG and disabled any SIP ALG?

is this the correct configuration for a static ip address?

and this

Always debug the simplest things first:

  1. Get *43 (echo test) to work from a single extension.
  2. Get calls between extensions to work properly.
  3. Get outside calls working.

Assuming that your softphones are on the same LAN subnet as the PBX: In X-Lite, Topology tab, select Use Local IP Address and uncheck Enable ICE. If the extension is chan_sip, set NAT to No (pjsip figures this out automatically).

If you still have trouble, post a log of the simplest call that fails, along with details about what you hear and what you see on the softphone.

BTW, what is about?

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