Stewart1
(Stewart)
March 18, 2020, 6:04am
2
See
Hello,
Using pjsip on FreePBX 14.0.5.25 (Asterisk 13.24.1) on my landline, I have a misc destination setup to my cell phone. When the deviation is activated and people call my landline, it establishes a connection with my cell that I can answer, but then NO SOUND at all. After a while, it disconnects.
There are some related questions, but on chan-sip (activating fax, adding progressiveband, etc.), so it does not help
Anybody with any idea why that happens and what can be done to cure this ?
…
In your router/firewall, forward the RTP port range (default UDP 10000-20000) to the PBX.
Or, set up some way to send audio from Asterisk prior to the cell being answered.
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