I am using Freepbx with Asterisk version 13 for connecting to the SIP Trunk. The trunk is registered and inbound routes and outbound routes are configured and working properly. On checking both inbound and outbound calls, the sound is not audible to both ends. I have attached the log of the call. Please help out here as I am very new to FreePBX.
I am using chan_sip in port 5060. The providers had provided the configuration to allow alaw, ulaw, g729 and g722. In the outbound configuration, I have following codecs allowed.
I have disabled the firewalld service and iptables in the Linux server. Also, I noticed call being automatically disconnecting after 31 seconds:
NOTICE: chan_sip.c:29644 check_rtp_timeout: Disconnecting call ‘SIP/NTC-00000063’ for lack of RTP activity in 31 seconds
Pastebin link: https:// pastebin .com/eJuvCUZx (Cannot post link as new user!)