Hi all,
I’m debugging a new FreePBX+Asterisk system i’ve just built, and i’m trying to figure out why my calls aren’t connecting. I’ve followed tutorials to set up inbound and outbound routes, and I can see calls attempted, and sometimes get a short connection but then an immediate call drop (with no audio either side). This is the same for internal extension dialling, and external calls (routed out via the SIP trunk). The SIP trunk provider has confirmed they are seeing the activity on their end
It’s possible a firewall configuration thing, but i’ve tried all sorts to get it working to no avail.
There’s always a mention of “no reply to our critical packet” in the logs when this happens, so it’s looking like a firewall thing. Any ideas what I could be doing wrong?
[2018-02-01 09:24:00] VERBOSE[23392][C-0000007a] app_dial.c: Connected line update to SIP/1002-00000065 prevented.
[2018-02-01 09:24:00] VERBOSE[23392][C-0000007a] app_dial.c: SIP/1001-00000066 answered SIP/1002-00000065
[2018-02-01 09:24:00] VERBOSE[23393][C-0000007a] bridge_channel.c: Channel SIP/1001-00000066 joined ‘simple_bridge’ basic-bridge <28c00e62-c888-4133-b9c8-50245c1f8a52>
[2018-02-01 09:24:00] VERBOSE[23392][C-0000007a] bridge_channel.c: Channel SIP/1002-00000065 joined ‘simple_bridge’ basic-bridge <28c00e62-c888-4133-b9c8-50245c1f8a52>
[2018-02-01 09:24:06] WARNING[2619] chan_sip.c: Retransmission timeout reached on transmission e4OqmDGtOzJjvEqetSb4VA… for seqno 2 (Critical Response) – See REDACTED
Packet timed out after 6399ms with no response
[2018-02-01 09:24:06] WARNING[2619] chan_sip.c: Hanging up call e4OqmDGtOzJjvEqetSb4VA… - no reply to our critical packet (see REDACTED).
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] bridge_channel.c: Channel SIP/1002-00000065 left ‘simple_bridge’ basic-bridge <28c00e62-c888-4133-b9c8-50245c1f8a52>
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] app_macro.c: Spawn extension (macro-dial-one, s, 53) exited non-zero on ‘SIP/1002-00000065’ in macro ‘dial-one’
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] app_macro.c: Spawn extension (macro-exten-vm, s, 20) exited non-zero on ‘SIP/1002-00000065’ in macro ‘exten-vm’
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Spawn extension (ext-local, 1001, 2) exited non-zero on ‘SIP/1002-00000065’
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [h@ext-local:1] Macro(“SIP/1002-00000065”, “hangupcall,”) in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/1002-00000065”, “1?theend”) in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-02-01 09:24:06] VERBOSE[23393][C-0000007a] bridge_channel.c: Channel SIP/1001-00000066 left ‘simple_bridge’ basic-bridge <28c00e62-c888-4133-b9c8-50245c1f8a52>
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/1002-00000065”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/1002-00000065”, "SIP/1001-00000066 monior file= ") in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [s@macro-hangupcall:5] AGI(“SIP/1002-00000065”, “attendedtransfer-rec-restart.php,SIP/1001-00000066,”) in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] res_agi.c: <SIP/1002-00000065>AGI Script attendedtransfer-rec-restart.php completed, returning 0
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Executing [s@macro-hangupcall:6] Hangup(“SIP/1002-00000065”, “”) in new stack
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] app_macro.c: Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/1002-00000065’ in macro ‘hangupcall’
[2018-02-01 09:24:06] VERBOSE[23392][C-0000007a] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on ‘SIP/1002-00000065’
Thanks in advance,
Adam