Hi everyone, I have configured the server with FreePBX and 2 Yealink phones, but when they call each other the call works but when I try to answer the audio does not go through and the two phones are silent … what can I do
Audio (RTP) uses ports 10,000-20,000 by default. Is there any firewalls between the phones and or PBX? Set SIP debug on and possibly do a wireshark if you are not sure where the block could be.
In Asterisk SIP Settings, confirm that Local Networks and External Address are correctly set. After changing these, Submit and Apply Config, you must restart Asterisk.
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