Hi all
I’m running an old version installed nearly 2 yrs ago (FreePBX-Distro-Net-64bit-1.811.210.57). I honestly have not payed much attention to it because it has worked fine and I take the view that if it isn’t broken, don’t fix it. There have been no unusual calls/charges from my VoSP, however today I noticed msgs like this repeating on the asterisk cli:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [660011972592511307@from-sip-external:1] NoOp("SIP/x.x.x.x-0014f520", "Received incoming SIP connection from unknown peer to 660011972592511307") in new stack
-- Executing [660011972592511307@from-sip-external:2] Set("SIP/x.x.x.x-0014f520", "DID=660011972592511307") in new stack
-- Executing [660011972592511307@from-sip-external:3] Goto("SIP/x.x.x.x-0014f520", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/x.x.x.x-0014f520", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/x.x.x.x-0014f520", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2014-06-10 19:33:48.196 EDT.
-- Executing [s@from-sip-external:6] Answer("SIP/x.x.x.x-0014f520", "") in new stack
-- Executing [s@from-sip-external:7] Wait("SIP/x.x.x.x-0014f520", "2") in new stack
-- Executing [s@from-sip-external:8] Playback("SIP/x.x.x.x-0014f520", "ss-noservice") in new stack
-- <SIP/x.x.x.x-0014f520> Playing 'ss-noservice.ulaw' (language 'en')
-- Executing [s@from-sip-external:9] PlayTones("SIP/x.x.x.x-0014f520", "congestion") in new stack
-- Executing [s@from-sip-external:10] Congestion("SIP/x.x.x.x-0014f520", "5") in new stack
== Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/x.x.x.x-0014f520'
-- Executing [h@from-sip-external:1] Hangup("SIP/x.x.x.x-0014f520", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/x.x.x.x-0014f520'
[2014-06-10 19:34:05] WARNING[-1]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ff698c29f5b0bb082d2e4742cd276d09 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
x.x.x.x is my static IP from my ISP
and every once in a while this
XtrComm*CLI> core show channels verbose
Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgedTo
SIP/x.x.x.x-00 from-sip-external s 8 Up Playback ss-noservice 2014 00:00:06 (None)
[2014-06-10 19:37:34] NOTICE[-1]: chan_sip.c:24929 handle_request_register: Registration from '"735" <sip:[email protected]:5060>' failed for '50.23.115.116:5111' - No matching peer found
[2014-06-10 19:37:35] NOTICE[-1]: chan_sip.c:24929 handle_request_register: Registration from '"735" <sip:[email protected]:5060>' failed for '50.23.115.116:5111' - No matching peer found
The box sits behind a DD-WRT v24-sp2 (8/07/10) std flashed router.
What can I do to stop these actions?
Of course there are 62 online modules available for upgrade according to FPBX status, but I don’t really know how to perform the upgrades, and I worry they may “break” something during the upgrade, which is part of the reason I try to leave well enough alone.
Thanks in advance.