I didn’t get the scanner message on any routing at all until about a month ago, towards the very end of March. By that time the systems were up and running for two weeks.

I still can’t figure out why it keeps telling me the number is not in service.

not accepting 5060 from anyone but your providers direct ip(s) will improve security, if you still get probed, then yes, question your VoSP but provide them with a diagnostic pcap.

there was never a mention of analog in this thread, just E1,

A caveat, I know nothing about these devices, but If the device has two ip’s logically they have two trunks, if they had had two ports, I would assume one trunk and two extensions, if two ip’s then routing through that ip specific trunk is ineffective?

Maybe the rules are not correctly defined, I’m not sure that by only setting two different IPs is enough to direct calls through one port or another. In fact that gateway has four T1 ports.

Try this as a starting point

https://support.digium.com/community/s/article/How-do-I-connect-an-AsteriskNOW-system-with-FreePBX-to-a-Digium-gateway

No guessing allowed, (it’s pointless).

@caoquocai . . . please
a) RTFM
and
b) let us know more about what you did.

There is another current thread here that might pertain, I contend that udp can allow spoofed connections from perhaps a 400 pound guy in New Zealand, It is likely that restricting yourself to tcp where possible

Just a challenge to those that have a better understanding . . . .

I created two trunks to two IP addresses of gateway. In gateway also showed two trunks connected from one IP address: 192.168.1.10 (freepbxIP).

when calling, freePBX routes to gateway with string: 6XXX@freepbxIP or 7XXX@freepbxIP without any information about IP of gateway.

is that problem to make the gateway doesn’t know the calls to which trunk in gateway?

I tried to create one more freePBX with IP 192.192.1.13 and configured like below, it worked.

freepbx1 (192.168.1.10) --> trunk 6XXX in gateway --> PSTN1
freepbx2 (192.168.1.13) --> trunk 7XXX in gateway --> PSTN2

Thanks

Did you read all of

yet?

I already read this some times, but it not mentions about one free-PBX to gateway with two trunks.

You will have to be way more explicit as to how you did that, you might want to go to the Digium support site because that is where you bought it, I’m sure you will find other folks with the same hardware and likely useful solutions, here you will find . . . . well , not so many, so we can’t guess.

The PBX is somehow treating the HT503 as NATed, even though it’s not. (Your public IP address should not be appearing in any SIP sent to the HT. Also, the log should show “Transmitting (no NAT) …” for such requests.

Your IP addresses are strange. I assume that the PBX is in South Africa, the router has public IP 197.184.153.199 and your LAN uses 192.41.100.0/24 as ‘private’ addresses, even though those are not RFC1918 addresses. If that’s correct, in Asterisk SIP settings, NAT Settings, Local Networks confirm that you have 192.41.100.0 / 24. If you change this, after Submit and Apply Config, also restart Asterisk (or reboot the server).

However, I doubt that this is causing the busy response. Log into the HT503 and confirm that the status page shows the FXO port shows Hook as idle and DND as no. Temporarily unplug the cord from the Line jack of the HT503 and connect it (the cord from the Panasonic) to a standard analog phone for testing. Confirm that you get a dial tone and can successfully call ext 102.

Finally, I don’t understand why the @112-1 is appearing in the To header (and presumably in the SIP URI). This might explain the busy response. Do you have some trunk settings not shown?

If you still have trouble, please post the log for the call setup including SIP trace with the INVITE. Also, all trunk settings and all non-default settings in the HT503.

Hi,

That 197.184.153.199 IP address you see, is that of my laptop’s ISP. My laptop connects into the LAN via VPN and has Linphone installed.

But on the inside on the LAN, the FXO’s IP address is 192.41.100.31 and FreePBX on 192.41.100.240. There is no firewall between those two. And, my laptop can call one of the Polycom SIP phones on 192.41.100.160, without a problem.

The HT503 shows Busy

FXO13.jpg1366x768 93.9 KB

When I plug a phone into that line, I get a dial tone and I can phone 102, though the dial tone isn’t exactly the same as a normal POTS line’s dial tone. I’m not sure if that plays a role?

Here’s the debug log while making a call to 102:

== Setting global variable ‘SIPDOMAIN’ to ‘192.41.100.240’
== Setting global variable ‘SIPDOMAIN’ to ‘192.41.100.240’
== Setting global variable ‘SIPDOMAIN’ to ‘192.41.100.240’
– Executing [102@from-internal:1] Macro(“PJSIP/114-0000016e”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/114-0000016e”, “TOUCH_MONITOR=1523853069.431”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/114-0000016e”, “AMPUSER=114”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“PJSIP/114-0000016e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“PJSIP/114-0000016e”, “1?Set(REALCALLERIDNUM=114)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/114-0000016e”, “AMPUSER=114”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“PJSIP/114-0000016e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“PJSIP/114-0000016e”, “AMPUSERCIDNAME=Mariska”) in new stack
– Executing [s@macro-user-callerid:8] ExecIf(“PJSIP/114-0000016e”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“PJSIP/114-0000016e”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/114-0000016e”, “AMPUSERCID=114”) in new stack
– Executing [s@macro-user-callerid:11] Set(“PJSIP/114-0000016e”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/114-0000016e”, “CALLERID(all)=“Mariska” <114>”) in new stack
– Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/114-0000016e”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/114-0000016e”, “1?Set(GROUP(concurrency_limit)=114)”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“PJSIP/114-0000016e”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:16] NoOp(“PJSIP/114-0000016e”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“PJSIP/114-0000016e”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/114-0000016e”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,37)
– Executing [s@macro-user-callerid:37] Set(“PJSIP/114-0000016e”, “CALLERID(number)=114”) in new stack
– Executing [s@macro-user-callerid:38] Set(“PJSIP/114-0000016e”, “CALLERID(name)=Mariska”) in new stack
– Executing [s@macro-user-callerid:39] GotoIf(“PJSIP/114-0000016e”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:40] Set(“PJSIP/114-0000016e”, “CDR(cnam)=Mariska”) in new stack
– Executing [s@macro-user-callerid:41] Set(“PJSIP/114-0000016e”, “CDR(cnum)=114”) in new stack
– Executing [s@macro-user-callerid:42] Set(“PJSIP/114-0000016e”, “CHANNEL(language)=en”) in new stack
– Executing [102@from-internal:2] Gosub(“PJSIP/114-0000016e”, “sub-record-check,s,1(out,102,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/114-0000016e”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/114-0000016e”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/114-0000016e”, “NOW=1523853069”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/114-0000016e”, “__DAY=16”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/114-0000016e”, “__MONTH=04”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/114-0000016e”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/114-0000016e”, “__TIMESTR=20180416-063109”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/114-0000016e”, “__FROMEXTEN=114”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/114-0000016e”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/114-0000016e”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/114-0000016e”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/114-0000016e”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/114-0000016e”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/114-0000016e”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/114-0000016e”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“PJSIP/114-0000016e”, “Outbound Recording Check from 114 to 102”) in new stack
– Executing [out@sub-record-check:2] Set(“PJSIP/114-0000016e”, “RECMODE=dontcare”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“PJSIP/114-0000016e”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“PJSIP/114-0000016e”, “recordcheck,1(dontcare,out,102)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/114-0000016e”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/114-0000016e”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/114-0000016e”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“PJSIP/114-0000016e”, “”) in new stack
– Executing [102@from-internal:3] ExecIf(“PJSIP/114-0000016e”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [102@from-internal:4] Set(“PJSIP/114-0000016e”, “MOHCLASS=default”) in new stack
– Executing [102@from-internal:5] ExecIf(“PJSIP/114-0000016e”, “0?Set(TRUNKCIDOVERRIDE=114)”) in new stack
– Executing [102@from-internal:6] Set(“PJSIP/114-0000016e”, “_NODEST=”) in new stack
– Executing [102@from-internal:7] Macro(“PJSIP/114-0000016e”, “dialout-trunk,1,102,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“PJSIP/114-0000016e”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“PJSIP/114-0000016e”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERID(num)=114)”) in new stack
– Executing [s@macro-dialout-trunk:4] GotoIf(“PJSIP/114-0000016e”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“PJSIP/114-0000016e”, “DIAL_NUMBER=102”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“PJSIP/114-0000016e”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“PJSIP/114-0000016e”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“PJSIP/114-0000016e”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“PJSIP/114-0000016e”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,11)
– Executing [s@macro-dialout-trunk:11] GotoIf(“PJSIP/114-0000016e”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:12] Macro(“PJSIP/114-0000016e”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“PJSIP/114-0000016e”, “114”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“PJSIP/114-0000016e”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“PJSIP/114-0000016e”, “off”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] ExecIf(“PJSIP/114-0000016e”, “0?Set(REALCALLERIDNUM=114)”) in new stack
– Executing [s@macro-outbound-callerid:7] GotoIf(“PJSIP/114-0000016e”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,11)
– Executing [s@macro-outbound-callerid:11] Set(“PJSIP/114-0000016e”, “USEROUTCID=“Mariska” <114>”) in new stack
– Executing [s@macro-outbound-callerid:12] Set(“PJSIP/114-0000016e”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“PJSIP/114-0000016e”, “TRUNKOUTCID=114”) in new stack
– Executing [s@macro-outbound-callerid:14] GotoIf(“PJSIP/114-0000016e”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,19)
– Executing [s@macro-outbound-callerid:19] ExecIf(“PJSIP/114-0000016e”, “1?Set(CALLERID(all)=114)”) in new stack
– Executing [s@macro-outbound-callerid:20] ExecIf(“PJSIP/114-0000016e”, “1?Set(CALLERID(all)=“Mariska” <114>)”) in new stack
– Executing [s@macro-outbound-callerid:21] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“PJSIP/114-0000016e”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:24] Set(“PJSIP/114-0000016e”, “CDR(outbound_cnum)=114”) in new stack
– Executing [s@macro-outbound-callerid:25] Set(“PJSIP/114-0000016e”, “CDR(outbound_cnam)=Mariska”) in new stack
– Executing [s@macro-dialout-trunk:13] GosubIf(“PJSIP/114-0000016e”, “1?sub-flp-1,s,1()”) in new stack
– Executing [s@sub-flp-1:1] ExecIf(“PJSIP/114-0000016e”, “1?Return()”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“PJSIP/114-0000016e”, “OUTNUM=102”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“PJSIP/114-0000016e”, “custom=SIP/114-1”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“PJSIP/114-0000016e”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“PJSIP/114-0000016e”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:18] Macro(“PJSIP/114-0000016e”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/114-0000016e”, “”) in new stack
– Executing [s@macro-dialout-trunk:19] GotoIf(“PJSIP/114-0000016e”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:20] Set(“PJSIP/114-0000016e”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“PJSIP/114-0000016e”, “__CRM_DESTINATION=102”) in new stack
– Executing [s@macro-dialout-trunk:22] Set(“PJSIP/114-0000016e”, “__CRM_SOURCE=114”) in new stack
– Executing [s@macro-dialout-trunk:23] AGI(“PJSIP/114-0000016e”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/114-0000016e>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:24] Set(“PJSIP/114-0000016e”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:25] NoOp(“PJSIP/114-0000016e”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:26] GotoIf(“PJSIP/114-0000016e”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/114-0000016e”, “1?Set(CONNECTEDLINE(num,i)=102)”) in new stack
– Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/114-0000016e”, “1?Set(CONNECTEDLINE(name,i)=CID:114)”) in new stack
– Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/114-0000016e”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)114)”) in new stack
– Executing [s@macro-dialout-trunk:30] GotoIf(“PJSIP/114-0000016e”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/114-0000016e”, “SIP/114-1/102@112-1,300,T”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/114-1/102@112-1
– Got SIP response 486 “Busy Here” back from 192.41.100.31:5062
– SIP/114-1-00000040 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/114-0000016e”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
– Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/114-0000016e”, “0?continue,1:s-BUSY,1”) in new stack
– Goto (macro-dialout-trunk,s-BUSY,1)
– Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“PJSIP/114-0000016e”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“PJSIP/114-0000016e”, “busy”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:3] Busy(“PJSIP/114-0000016e”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘PJSIP/114-0000016e’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 102, 7) exited non-zero on ‘PJSIP/114-0000016e’
– Executing [h@from-internal:1] Macro(“PJSIP/114-0000016e”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/114-0000016e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/114-0000016e”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/114-0000016e”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/114-0000016e”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/114-0000016e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/114-0000016e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/114-0000016e’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/114-0000016e’
– PJSIP/114-0000016e Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/114-0000016e”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/114-0000016e”, “HANGUP CAUSE: 17”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/114-0000016e”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/114-0000016e”, “MASTER CHANNEL: 1523853069.431 = 1523853069.431”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/114-0000016e”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/114-0000016e”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/114-0000016e”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/114-0000016e>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/114-0000016e”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/114-0000016e’
– PJSIP/114-0000016e Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

I will attempt to upload the screenshots again, seems all of them didn’t upload.

fxo1.jpg1366x768 104 KB

fxo2.jpg1366x768 96.1 KB

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fxo4.jpg1366x768 93.8 KB

fxo5.jpg1366x768 118 KB

fxo6.jpg1366x768 119 KB

fxo7.jpg1366x768 123 KB

FXO8.jpg1366x768 137 KB

FXO9.jpg1366x768 125 KB

FXO11.jpg1366x768 125 KB

FXO12.jpg1366x768 127 KB

The non-standard dial tone should not be a problem.

I’m guessing that because you don’t have the FXS port configured, the lack of registration is kicking the device into ‘life line’ mode and disconnecting the FXO port. In the HT Advanced Settings, try setting Life Line Mode to Always Disconnected and see whether the FXO port now shows idle (and if so, whether you can call). If not, try setting up an extension and registering the FXS port.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Hai

whether mysql 5.7 will be support ?

root@pbx:~ $ fwconsole ma list |grep motif
| motif | 12.0.4.2 | Enabled | GPLv3+ |
WARNING: Always run Incredible PBX behind a secure hardware-based >firewall.
root@pbx:~ $

Looks like 12.0.4.2 is what it is.

How would I go about updating this? (sorry, beyond new to FreePBX…)

Thx.

Why is that? This external server is only the storage for saved calls. Even if it goes down how will it cause whole calling system go down on the other machine?

Are all modules updated?

fwconsole ma upgradeall

You’re saying the fix is in CORE 14.0.5.16 but do we know when the bug was introduced? I.e. did it even exist in 13 (which I’m on)?

I’m running in a VPS on Vultr. Grabbed a TCPDUMP as you described (thank you!) Of course NOW it IS ringing so I’ll have to wait till it’s failing again. I made some changes last night to followme and stuff last night but I think it was intermittent before that too.

At least now I’m armed for when it does happen so thanks again @wmjackson

J.