Hello, I’m hosting FreePBX on AWS for a lab. I’m trying to make a call from one UCP phone to another. When the call starts, the destination UCP phone rings, but then there is no audio.
Here is an excerpt from Asterisk logs. It seems complaining about STUN requests (I do not need it and I didn’t set one), but I’m not sure if it’s the root cause:
12454 [2023-06-15 11:37:55] VERBOSE[5799] netsock2.c: Using SIP RTP Audio TOS bits 184
12455 [2023-06-15 11:37:55] VERBOSE[5799] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
12456 [2023-06-15 11:37:55] VERBOSE[5799] netsock2.c: Using SIP RTP Audio CoS mark 5
12457 [2023-06-15 11:37:55] VERBOSE[5799] res_rtp_asterisk.c: DTLS ECDH initialized (automatic), faster PFS enabled
12458 [2023-06-15 11:37:55] VERBOSE[14768][C-00000013] app_dial.c: PJSIP/99201-0000001b is ringing
12459 [2023-06-15 11:37:58] ERROR[5799] pjproject: icess0x7fef781d7bc8 ......Error sending STUN request: Network is unreachable
12460 [2023-06-15 11:37:58] VERBOSE[14768][C-00000013] app_dial.c: PJSIP/99201-0000001b answered PJSIP/99200-0000001a
12461 [2023-06-15 11:37:58] ERROR[5799] pjproject: icess0x7fef781239b8 ...Error sending STUN request: Network is unreachable
12462 [2023-06-15 11:37:58] VERBOSE[14820][C-00000013] bridge_channel.c: Channel PJSIP/99201-0000001b joined 'simple_bridge' basic-bridge <4ed5686d-17e8-460b-bbbe-8af76b433ddd>
12463 [2023-06-15 11:37:58] VERBOSE[14768][C-00000013] bridge_channel.c: Channel PJSIP/99200-0000001a joined 'simple_bridge' basic-bridge <4ed5686d-17e8-460b-bbbe-8af76b433ddd>
12464 [2023-06-15 11:38:25] NOTICE[14500] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/99201-0000001b' for lack of audio RTP activity in 30 seconds
How can I set things up properly? Please, note that SIP apps (e.g, Zoiper) work fine instead.