I have a problem using FreePBX 14 with Asterisk 15, new installation. There are some extensions that has one way audio after 1 second of the call, I mean, when you answer the call, you can hear the first second of the call, then it goes to only one way audio.
I have check RTP debug and I can see that when the call is answered, there are some packages that sent and received, after a few (10-15) it stop to receiving and only sent.
I have check SIP messages and find out that the call setup is correct and INVITE, Trying, Ringing, ACK, etc., comes and goes as expected.
Call sequence, SIP and RTP debug here: https://pastebin.com/1BALwVZ6
Call sequence is INVITE-Trying-Ringing-200 OK-ACK, and then again INVITE-Trying-200 OK-ACK, after the second 200 OK audio stops.
After the call is answered I see a new INVITE coming from the extensión with the problem, so I create a file /etc/asterisk/pjsip.endpoint_custom_post.conf and write:
(+) type=endpoint direct_media=no
Then check on CLI, ‘pjsip show endpoint 113’ and I see that direct_media is false:
Even though I still see the re-invite on calls.
Please your help, I don’t know what else to check.
PJSIP Extension 113 config, Extension 113 is on a branch office:
 type=endpoint aors=113 auth=113-auth tos_audio=ef tos_video=af41 cos_audio=5 cos_video=4 allow=ulaw,alaw,gsm,g729 context=from-internal callerid=device <113> dtmf_mode=rfc4733 [email protected] mwi_subscribe_replaces_unsolicited=yes aggregate_mwi=yes use_avpf=no rtcp_mux=no bundle=no ice_support=no media_use_received_transport=no trust_id_inbound=yes media_encryption=no timers=yes media_encryption_optimistic=no send_pai=yes rtp_symmetric=yes rewrite_contact=yes force_rport=yes language=es one_touch_recording=on record_on_feature=apprecord record_off_feature=apprecord language=es  type=aor [email protected] max_contacts=1 remove_existing=yes maximum_expiration=7200 qualify_frequency=60 [113-auth] type=auth auth_type=userpass password=super secret password ;) username=113 [113-identify] type=identify endpoint=113 (+) type=endpoint direct_media=no [0.0.0.0-udp] type=transport protocol=udp bind=0.0.0.0:5060 external_media_address=1X2.X04.1X0.14 -> public IP, behind Firewall, no public access, communication between branches are trough VPN, every office has its own firewall external_signaling_address=1X2.X04.1X0.14 allow_reload=no tos=cs3 cos=3 local_net=172.16.10.0/24 -> Local SIP Provider ISP local_net=172.23.1.0/8 -> Server LAN local_net=192.168.0.0/16 -> Remote Branches