Using freepbx 14, I have installed two extensions (201 and 202). Each phone displays its own extension. I get a dial tone on both of them. When 202 calls 201, the phone rings, but upon answering, there is no audio in either direction. When 201 calls 202, a call timer begins, but 202 does not ring. When I call *97 (voicemail) from either extension, a call timer starts (showing My Voicemail), but there is no audio, and I am not prompted to enter my password.
Check Settings --> Asterisk SIP Settings and make sure your WAN and Lan IP schemes are correct. Also verify Chan SIP WAN IP is correct
Are the phones and the pbx on the same network?
Thanks for your reply. I think you are on the right track.
The PBX is in my DMZ, and the phones are on my local network, so the router might be blocking some packets. What TCP and UDP ports do I need to allow through my router?
Yo should allow the UDP port that your PBX is listening on, either 5060, 5160 or both if both sip drivers are enabled and used, and the UDP port range that your RTP is set to, by default 10000-20000.
You also need to correctly configure the sip general settings on FreePBX.
Thanks. Got it working.
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