Hi, I am trying to troubleshoot an Anveo-direct Asterisk setup.
My calls do connect, and are answered by Asterisk. But when playing back an audio file in my dialplan, no sound is sent to the caller. I have tested the same dialplan with my SIP URI, and it works fine.
Based on the following two error messages, my feeling is it is a firewall or signalling issue:
DEBUG[C-00000001]: chan_sip.c:5847 do_setnat: Setting NAT on RTP to Off DEBUG[C-00000001]: chan_sip.c:10376 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
WARNING: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission Q17T3O550D4AD0NOMMBKPN8M3G@220.127.116.11-b2b_1 for seqno 200 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response
But I have opened the requisite ports, as seen here on my router:
To better illustrate my issue, I am attaching here my debug log in Asterisk, my SIP trace in Anveo, my extensions.conf, and my pjsip.conf:
Based on this debug info, or other individual’s previous experience with Anveo-direct, could anyone please provide me with suggestions on moving forward? Many thanks !