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Inbound fax issues

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@SilkBC wrote:

Hello,

I am running FreePBX v13.0.197.14 (firmware 10.13.66-22) and Asterisk v13.23.1. Up until now, inbound fax-to-email has been working, but my client has reported to me that they have had a couple of their clients try to fax documents and either the entire fax does not come through (only some pages), or it fails.

My client does seem to mostly get faxes OK, but they are only a few pages. The issues seem to occur wih faxes that consist ofseveralpages.

I have been able to reproduce the problem with both a physical fax and a email-to-fax gateway that another client of mine uses. In my testing, I have tried two different multi-page documents: one consisting of 16 pages and the other 9 pages.

Regardless of which document or fax method I try, the whole fax does not go through. The fax is indicated as failed, but a couple of pages of my fax do still make it through, but it is random: sometime it will be the first two or three, but in one case, only page 3 made it through.

I have included a log of my last fax attempt below, which is from a physcial fax machine (IP addresss of my client’s PBX changed). I don’t see anything overtly wrong, but I am not an expert at this either.

Not sure if this points ot a problem with FreePBX, or perhaps the SIP trunk provider. I have opened a ticket with them too, but am just trying to cover all the bases I can.

The only real possible issue I see in the call trace is “WARNING T.30 ECM carrier not found”.

Here is the call trace:

[2019-12-10 16:33:48] VERBOSE[14303][C-00016876] netsock2.c: Using SIP RTP TOS bits 184
[2019-12-10 16:33:48] VERBOSE[14303][C-00016876] netsock2.c: Using SIP RTP CoS mark 5
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk-sip-ThinkTel-SIP-YEG1:1] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “GROUP()=OUT_2”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk-sip-ThinkTel-SIP-YEG1:2] Goto(“SIP/ThinkTel-SIP-YEG1-0001d781”, “from-trunk,6046838032,1”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (from-trunk,6046838032,1)
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:1] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__DIRECTION=INBOUND”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:2] Gosub(“SIP/ThinkTel-SIP-YEG1-0001d781”, “sub-record-check,s,1(in,6046838032,dontcare)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:1] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?initialized”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:2] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__REC_STATUS=INITIALIZED”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:3] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “NOW=1576024428”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:4] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__DAY=10”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:5] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__MONTH=12”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:6] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__YEAR=2019”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:7] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__TIMESTR=20191210-163348”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:8] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__FROMEXTEN=unknown”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:9] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__MON_FMT=wav”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:10] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “Recordings initialized”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:11] ExecIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?Set(ARG3=dontcare)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:12] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “REC_POLICY_MODE_SAVE=”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:13] ExecIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?Set(REC_STATUS=NO)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:14] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “2?checkaction”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (sub-record-check,s,17)
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@sub-record-check:17] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “1?sub-record-check,in,1”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (sub-record-check,in,1)
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [in@sub-record-check:1] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “Inbound Recording Check to 6046838032”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [in@sub-record-check:2] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FROMEXTEN=unknown”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [in@sub-record-check:3] ExecIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “10?Set(FROMEXTEN=7783401109)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [in@sub-record-check:4] Gosub(“SIP/ThinkTel-SIP-YEG1-0001d781”, “recordcheck,1(dontcare,in,6046838032)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “Starting recording check against dontcare”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“SIP/ThinkTel-SIP-YEG1-0001d781”, “dontcare”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [recordcheck@sub-record-check:3] Return(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [in@sub-record-check:5] Return(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:3] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__FROM_DID=6046838032”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:4] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “returnhere=1”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:5] Gosub(“SIP/ThinkTel-SIP-YEG1-0001d781”, “app-blacklist-check,s,1()”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@app-blacklist-check:1] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?blacklisted”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@app-blacklist-check:2] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “CALLED_BLACKLIST=1”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [s@app-blacklist-check:3] Return(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:6] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “CDR(did)=6046838032”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:7] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:8] ExecIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0 ?Set(CALLERID(name)=7783401109)”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:9] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__MOHCLASS=”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:10] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__REVERSAL_REJECT=FALSE”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:11] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “1?post-reverse-charge”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (from-trunk,6046838032,13)
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:13] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:14] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:15] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:16] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:17] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:18] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “CallerID Entry Point”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:19] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAX_DEST=ext-fax^222^1”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:20] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAXOPT(faxdetect)=yes”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:21] Answer(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:48] VERBOSE[8996][C-00016876] pbx.c: Executing [6046838032@from-trunk:22] Wait(“SIP/ThinkTel-SIP-YEG1-0001d781”, “7”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] res_fax.c: Redirecting ‘SIP/ThinkTel-SIP-YEG1-0001d781’ to fax extension due to CNG detection
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Spawn extension (from-trunk, fax, 1) exited non-zero on ‘SIP/ThinkTel-SIP-YEG1-0001d781’
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [fax@from-trunk:1] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “__DIRECTION=INBOUND”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [fax@from-trunk:2] Goto(“SIP/ThinkTel-SIP-YEG1-0001d781”, “ext-fax,222,1”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (ext-fax,222,1)
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [222@ext-fax:1] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAX_FOR=FAX Metrix (222)”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [222@ext-fax:2] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “Receiving Fax for: FAX Metrix (222), From: “PSD INC” <7783401109>”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [222@ext-fax:3] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAX_RX_USER=222”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [222@ext-fax:4] Set(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAX_RX_EMAIL_LEN=14”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [222@ext-fax:5] Goto(“SIP/ThinkTel-SIP-YEG1-0001d781”, “s,receivefax”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (ext-fax,s,3)
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [s@ext-fax:3] StopPlayTones(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] pbx.c: Executing [s@ext-fax:4] ReceiveFAX(“SIP/ThinkTel-SIP-YEG1-0001d781”, “/var/spool/asterisk/fax/1576024428.125964.tif,f”) in new stack
[2019-12-10 16:33:50] VERBOSE[8996][C-00016876] res_fax.c: Channel ‘SIP/ThinkTel-SIP-YEG1-0001d781’ receiving FAX ‘/var/spool/asterisk/fax/1576024428.125964.tif’
[2019-12-10 16:34:05] WARNING[14303][C-00016876] chan_sip.c: Failed to initialize UDPTL, declining image stream
[2019-12-10 16:34:05] WARNING[14303][C-00016876] chan_sip.c: Failing due to no acceptable offer found
[2019-12-10 16:34:11] WARNING[8996][C-00016876] res_fax_spandsp.c: WARNING T.30 ECM carrier not found
[2019-12-10 16:34:12] WARNING[8996][C-00016876] res_fax_spandsp.c: WARNING T.30 ECM carrier not found
[2019-12-10 16:34:12] WARNING[8996][C-00016876] res_fax_spandsp.c: WARNING T.30 ECM carrier not found
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Spawn extension (ext-fax, s, 4) exited non-zero on ‘SIP/ThinkTel-SIP-YEG1-0001d781’
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [h@ext-fax:1] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “1?failed”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (ext-fax,h,104)
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [h@ext-fax:104] NoOp(“SIP/ThinkTel-SIP-YEG1-0001d781”, “FAX FAILED for: FAX Metrix (222) , From: “PSD INC” <7783401109>”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [h@ext-fax:105] Macro(“SIP/ThinkTel-SIP-YEG1-0001d781”, “hangupcall,”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “1?theend”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/ThinkTel-SIP-YEG1-0001d781”, “0?Set(CDR(recordingfile)=)”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/ThinkTel-SIP-YEG1-0001d781”, “”) in new stack
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/ThinkTel-SIP-YEG1-0001d781’ in macro ‘hangupcall’
[2019-12-10 16:35:07] VERBOSE[8996][C-00016876] pbx.c: Spawn extension (ext-fax, h, 105) exited non-zero on ‘SIP/ThinkTel-SIP-YEG1-0001d781’

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Switchvox vs PBXact which one to choose

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@nar6du14 wrote:

I everybody,

I have a particular need for an UC system able supporting push notifications for battery saving of mobile. I know that PBXact and Switchvox support this feature but which is working? I tested zulu uc but it seems to not work properly so I suspect it to not work on PBXact. I bought a valid certificate from comodo and installed it on freepbx but zulu on mobile says it is invalid…so is switchvox better at this time?

B.R

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Can't apply config after power fail

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@abcym15 wrote:

One of our FreePBX Distro servers (14.0.13.15) had a hard reboot yesterday during a power fail. Now I can’t Apply Config, we get the following error:

exit: 1
Unable to continue. SQLSTATE[42S01]: Base table or view already exists: 1050 Table ‘asterisk.kvstore_FreePBX_Media’ already exists in >/var/www/html/admin/libraries/BMO/Database.class.php on line 212
#0 [internal function]: PDO->query(‘CREATE TABLE IF…’)
#1 /var/www/html/admin/libraries/BMO/Database.class.php(212): call_user_func_array(‘parent::query’, Array)
#2 /var/www/html/admin/libraries/BMO/DB_Helper.class.php(128): FreePBX\Database->query(‘CREATE TABLE IF…’)
#3 /var/www/html/admin/libraries/BMO/DB_Helper.class.php(55): FreePBX\DB_Helper::createTable(‘kvstore_FreePBX…’)
#4 /var/www/html/admin/libraries/BMO/DB_Helper.class.php(155): FreePBX\DB_Helper::checkDatabase(Object(FreePBX\Media))
#5 /var/www/html/admin/libraries/BMO/Media.class.php(80): FreePBX\DB_Helper->getConfig(‘mediamissingmes…’)
#6 /var/lib/asterisk/bin/retrieve_conf(77): FreePBX\Media->getSupportedHTML5Formats()
#7 {main}

I have tried a reboot and this hasn’t solved the problem.

Do I need to somehow repair the database?

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FreePBX for Small Hotel, Internal Calls Only, WIFI Phones

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@xwebnetwork wrote:

Been doing some research and just want to get some input regarding a FreePBX setup for a 16 room hotel. Right now, we have a Comdial system and over the years, before purchasing the place, wires have been run, and re-run, and left behind so in an effort to clean up all of the crawl spaces and what not, the phone system immediately comes to mind. Ideally, I’d like to scrap the whole Comdial system in favor of a wireless solution so I have been looking into FreePBX as an option. I’ve got solid WIFI setup throughout the place and have plenty of remote server resources that I can use to deploy a VM with the PBX software. My question to those of you whom are well versed, do you you think this is a viable solution? From what I have read, seems like this would be perfect. Anyways, just to recap:

  • Scrap old wired, digital Comdial system
  • Deploy remote VM with FreePBX server
  • Purchase some inexpensive WIFI SIP phones and provision 1 per room with room # as extension (ie, 101, 102…)
  • The room phones will only be able to dial internal extensions, primarily 0 just to get to the front office
  • The primary office phone will be the ONLY phone to have multiple lines, internal and external, as it will also connect to the primary public facing numbers

Anyways, thank you in advance!

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FreePBX 15 v2?

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@Gordon0193 wrote:

Will a second release of of FreePBX 15 be coming out? The “1910” version seems kind of incomplete in regards to a land and go a least from a new install perspective.

Thanks!!

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I can’t remove the FreePBX modules that are unnecessary for me

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@plastilin wrote:

There are several modules that I do not plan to use in a small project. I delete them, and after a while they all recover again on their own. This has already been 3 times. How can you really remove unnecessary modules?

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Copying old HDD to new HDD via USB

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@johnjces wrote:

Having followed along on a thread regarding an HDD failure, I have a system with FreePBX 15 distro installed and working well. I forgot to replace the old HDD with a new solid state HDD and my options are,

Install New and restore backup requesting a new Zend reset or

Copy the old drive to the new drive and hopefully make the new drive bootable. One big issue I have, and it may not be an issue, is that my FreePBX box, as well as anything else I have, only has one SATA port. Go figure… well, I have a dell with two SATA ports but I simply don’t have the hard drive to hard drive type cable this connector on the MB supplies. Looks just like the rear of a SATA drive. So my only alternative is hooking up the SSHD via USB and an adapter.

I tried Clonezilla, had it copy the MBR and etc, but the new SSHD just booted to a Grub prompt and I have no experience with GRUB or what to enter to boot the box or make it work/boot permanently.

SO… my question is; what cloning software will work with a HDD/SSDD via USB and without further messing around to allow the new drive to boot as it should. OR, has anyone used Clonezilla and had a new drive boot normally?

Or go thru the time to reinstall on the new drive and do a restore.

Thanks so much!

John

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Gcsfuse & freepbx (FreePBX is unable to write to folder mounted with gcsfuse and the sync stops)

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@tbagalini wrote:

I am running freepbx Centos on a GCE VM. Syncing stops when FreePBX/Asterisk tries to write and create a folder in the mounted folder and FreePBX is unable to write / create folder in the folder. Everything works again when the system is rebooted because gcsfuse is no longer mounted.

Gcsfuse works when: .

-create a folder /var/spool/asterisk/cloud.
-mount it using gcsfuse --implicit-dirs siphqrecordings /var/spool/asterisk/cloud.
-create a sample text file nano test.txt.
-it syncs up to my bucket just fine

It completely stops working when:

-FreePBX tries to place a file and folder in the directory (a recorded call), the folder and file never gets written. I have unmount/reboot the server and then freepbx is able to use the folder and write files and directories to it. Even if I try to create a manual file using nano it never sync up.

FreePBX is unable to write to folder mounted with gcsfuse and the sync stops even for manually created files.

Thank you!

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Outbound call reporting

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@dancarter1220 wrote:

Hi, I have a PBXact with call centre and wallboard commercial modules.

I wanted to have a live wallboard with call stats for agents, free / busy. Number of calls made. Average call times etc.
Here is the issue.

I want to do this for outbound calls not inbound Queues.

It would seem it is only possible for inbound???

If it is not possible with the modules (Paid commercial) from Sangoma is there a 3rd party module I can use for this.
I find it hard to believe that with all the features available this is not.

I would appreciate any help or guidance.

Thanks

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Phone server for remote extensions

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@Red wrote:

If i was to place a phone server in the DMZ, would i do it at the modem level or at the pfsense firewall level. We cant use PBX VPN because we use Grandstream phones. The Sangoma phones we have aren’t working with the VPN and EPM like described in documentation, and we don’t have 300 bucks for hours of looking by Sangoma. We need a solution for our remote phones that, now, haven’t worked in over a month. I can set up internal phone systems blind folded, but setting up a hosted phone solution, has eluded me.
HELP PLEASE.

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Inbound route to extension fail

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@superdigi wrote:

– Executing [0123456789@from-pstn:1] Set (" SIP/future-0000000b ", " __FROM_DID= 0123456789 ") in new stack

– Executing [0123456789@from-pstn:2] Gosub (" SIP/future-0000000b ", " app-blacklist-check,s,1() ") in new stack

– Executing [s@app-blacklist-check:1] GotoIf (" SIP/future-0000000b ", " 0?blacklisted ") in new stack

– Executing [s@app-blacklist-check:2] Set (" SIP/future-0000000b ", " CALLED_BLACKLIST=1 ") in new stack

– Executing [s@app-blacklist-check:3] Return (" SIP/future-0000000b ", “”) in new stack

– Executing [0123456789@from-pstn:3] Set (" SIP/future-0000000b ", " CDR(did)= 0123456789 ") in new stack

– Executing [0123456789@from-pstn:4] ExecIf (" SIP/future-0000000b ", " 1 ?Set(CALLERID(name)=+60164319019) ") in new stack

– Executing [0123456789@from-pstn:5] Set (" SIP/future-0000000b ", " CHANNEL(musicclass)=default ") in new stack

– Executing [0123456789@from-pstn:6] Set (" SIP/future-0000000b ", " __MOHCLASS=default ") in new stack

– Executing [0123456789@from-pstn:7] Set (" SIP/future-0000000b ", " __CALLINGPRES_SV=allowed_not_screened ") in new stack

– Executing [0123456789@from-pstn:8] Set (" SIP/future-0000000b ", " CALLERPRES()=allowed_not_screened ") in new stack

– Executing [0123456789@from-pstn:9] Goto (" SIP/future-0000000b ", " from-did-direct,8888,1 ") in new stack

– Goto (from-did-direct,8888,1)

– Executing [8888@from-did-direct:1] Set (" SIP/future-0000000b ", " __RINGTIMER=15 ") in new stack

– Executing [8888@from-did-direct:2] Macro (" SIP/future-0000000b ", " exten-vm,novm,8888,0,0,0 ") in new stack

– Executing [s@macro-exten-vm:1] Macro (" SIP/future-0000000b ", " user-callerid, ") in new stack

– Executing [s@macro-user-callerid:1] Set (" SIP/future-0000000b ", " TOUCH_MONITOR=1576321224.11 ") in new stack

– Executing [s@macro-user-callerid:2] Set (" SIP/future-0000000b ", " AMPUSER=+60164319019 ") in new stack

– Executing [s@macro-user-callerid:3] GotoIf (" SIP/future-0000000b ", " 0?report ") in new stack

– Executing [s@macro-user-callerid:4] ExecIf (" SIP/future-0000000b ", " 1?Set(REALCALLERIDNUM=+60164319019) ") in new stack

– Executing [s@macro-user-callerid:5] Set (" SIP/future-0000000b ", " AMPUSER= ") in new stack

– Executing [s@macro-user-callerid:6] GotoIf (" SIP/future-0000000b ", " 0?limit ") in new stack

– Executing [s@macro-user-callerid:7] Set (" SIP/future-0000000b ", " AMPUSERCIDNAME= ") in new stack

– Executing [s@macro-user-callerid:8] GotoIf (" SIP/future-0000000b ", " 1?report ") in new stack

– Goto (macro-user-callerid,s,15)

– Executing [s@macro-user-callerid:15] GotoIf (" SIP/future-0000000b ", " 0?continue ") in new stack

– Executing [s@macro-user-callerid:16] Set (" SIP/future-0000000b ", " __TTL=64 ") in new stack

– Executing [s@macro-user-callerid:17] GotoIf (" SIP/future-0000000b ", " 1?continue ") in new stack

– Goto (macro-user-callerid,s,28)

– Executing [s@macro-user-callerid:28] Set (" SIP/future-0000000b ", " CALLERID(number)=+60164319019 ") in new stack

– Executing [s@macro-user-callerid:29] Set (" SIP/future-0000000b ", " CALLERID(name)=+60164319019 ") in new stack

– Executing [s@macro-user-callerid:30] Set (" SIP/future-0000000b ", " CDR(cnum)=+60164319019 ") in new stack

– Executing [s@macro-user-callerid:31] Set (" SIP/future-0000000b ", " CDR(cnam)=+60164319019 ") in new stack

– Executing [s@macro-user-callerid:32] Set (" SIP/future-0000000b ", " CHANNEL(language)=en ") in new stack

– Executing [s@macro-exten-vm:2] Set (" SIP/future-0000000b ", " RingGroupMethod=none ") in new stack

– Executing [s@macro-exten-vm:3] Set (" SIP/future-0000000b ", " __EXTTOCALL=8888 ") in new stack

– Executing [s@macro-exten-vm:4] Set (" SIP/future-0000000b ", " __PICKUPMARK=8888 ") in new stack

– Executing [s@macro-exten-vm:5] Set (" SIP/future-0000000b ", " RT= ") in new stack

– Executing [s@macro-exten-vm:6] Gosub (" SIP/future-0000000b ", " sub-record-check,s,1(exten,8888,) ") in new stack

– Executing [s@sub-record-check:1] Set (" SIP/future-0000000b ", " REC_POLICY_MODE_SAVE= ") in new stack

– Executing [s@sub-record-check:2] GotoIf (" SIP/future-0000000b ", " 1?check ") in new stack

– Goto (sub-record-check,s,7)

– Executing [s@sub-record-check:7] Set (" SIP/future-0000000b ", " __MON_FMT=wav ") in new stack

– Executing [s@sub-record-check:8] GotoIf (" SIP/future-0000000b ", " 1?next ") in new stack

– Goto (sub-record-check,s,11)

– Executing [s@sub-record-check:11] ExecIf (" SIP/future-0000000b ", " 0?Return() ") in new stack

– Executing [s@sub-record-check:12] ExecIf (" SIP/future-0000000b ", " 0?Set(__REC_POLICY_MODE=) ") in new stack

– Executing [s@sub-record-check:13] GotoIf (" SIP/future-0000000b ", " 0?exten,1 ") in new stack

– Executing [s@sub-record-check:14] Set (" SIP/future-0000000b ", " __REC_STATUS=INITIALIZED ") in new stack

– Executing [s@sub-record-check:15] Set (" SIP/future-0000000b ", " NOW=1576321224 ") in new stack

– Executing [s@sub-record-check:16] Set (" SIP/future-0000000b ", " __DAY=14 ") in new stack

– Executing [s@sub-record-check:17] Set (" SIP/future-0000000b ", " __MONTH=12 ") in new stack

– Executing [s@sub-record-check:18] Set (" SIP/future-0000000b ", " __YEAR=2019 ") in new stack

– Executing [s@sub-record-check:19] Set (" SIP/future-0000000b ", " __TIMESTR=20191214-190024 ") in new stack

– Executing [s@sub-record-check:20] Set (" SIP/future-0000000b ", " __FROMEXTEN=+60164319019 ") in new stack

– Executing [s@sub-record-check:21] Set (" SIP/future-0000000b ", " __CALLFILENAME=exten-8888-+60164319019-20191214-190024-1576321224.11 ") in new stack

– Executing [s@sub-record-check:22] Goto (" SIP/future-0000000b ", " exten,1 ") in new stack

– Goto (sub-record-check,exten,1)

– Executing [exten@sub-record-check:1] GotoIf (" SIP/future-0000000b ", " 0?callee ") in new stack

– Executing [exten@sub-record-check:2] Set (" SIP/future-0000000b ", " __REC_POLICY_MODE=always ") in new stack

– Executing [exten@sub-record-check:3] GotoIf (" SIP/future-0000000b ", " 0?caller ") in new stack

– Executing [exten@sub-record-check:4] GotoIf (" SIP/future-0000000b ", " 1?callee ") in new stack

– Goto (sub-record-check,exten,8)

– Executing [exten@sub-record-check:8] GosubIf (" SIP/future-0000000b ", " 1?record,1(exten,8888,+60164219012) ") in new stack

– Executing [record@sub-record-check:1] Set (" SIP/future-0000000b ", " AUDIOHOOK_INHERIT(MixMonitor)=yes ") in new stack

– Executing [record@sub-record-check:2] MixMonitor (" SIP/future-0000000b ", " 2019/12/14/exten-8888-+60164319019-20191214-190024-1576321224.11.wav, ") in new stack

== Begin MixMonitor Recording SIP/future-0000000b

– Executing [record@sub-record-check:3] Set (" SIP/future-0000000b ", " __REC_STATUS=RECORDING ") in new stack

– Executing [record@sub-record-check:4] Set (" SIP/future-0000000b ", " CDR(recordingfile)=exten-8888-+60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [record@sub-record-check:5] Return (" SIP/future-0000000b ", “”) in new stack

– Executing [exten@sub-record-check:9] Return (" SIP/future-0000000b ", “”) in new stack

– Executing [s@macro-exten-vm:7] Macro (" SIP/future-0000000b ", " dial-one,8888 ") in new stack

– Executing [s@macro-dial-one:1] Set (" SIP/future-0000000b ", " DEXTEN=8888 ") in new stack

– Executing [s@macro-dial-one:2] Set (" SIP/future-0000000b ", " DIALSTATUS_CW= ") in new stack

– Executing [s@macro-dial-one:3] GosubIf (" SIP/future-0000000b ", " 0?screen,1() ") in new stack

– Executing [s@macro-dial-one:4] GosubIf (" SIP/future-0000000b ", " 0?cf,1() ") in new stack

– Executing [s@macro-dial-one:5] GotoIf (" SIP/future-0000000b ", " 1?skip1 ") in new stack

– Goto (macro-dial-one,s,8)

– Executing [s@macro-dial-one:8] GotoIf (" SIP/future-0000000b ", " 0?nodial ") in new stack

– Executing [s@macro-dial-one:9] GotoIf (" SIP/future-0000000b ", " 0?continue ") in new stack

– Executing [s@macro-dial-one:10] Set (" SIP/future-0000000b ", " EXTHASCW=ENABLED ") in new stack

– Executing [s@macro-dial-one:11] GotoIf (" SIP/future-0000000b ", " 0?next1:cwinusebusy ") in new stack

– Goto (macro-dial-one,s,23)

– Executing [s@macro-dial-one:23] GotoIf (" SIP/future-0000000b ", " 1?next3:continue ") in new stack

– Goto (macro-dial-one,s,24)

– Executing [s@macro-dial-one:24] ExecIf (" SIP/future-0000000b ", " 0?Set(DIALSTATUS_CW=BUSY) ") in new stack

– Executing [h@from-did-direct:1] Macro (" SIP/future-0000000b ", " hangupcall, ") in new stack

– Executing [s@macro-hangupcall:1] GotoIf (" SIP/future-0000000b ", " 0?endmixmoncheck ") in new stack

– Executing [s@macro-hangupcall:2] Set (" SIP/future-0000000b ", " MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/exten-8888-+60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:3] System (" SIP/future-0000000b ", " test -e /var/spool/asterisk/monitor/exten-8888-+ 60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:4] NoOp (" SIP/future-0000000b ", " SYSTEMSTATUS = APPERROR ") in new stack

– Executing [s@macro-hangupcall:5] GotoIf (" SIP/future-0000000b ", " 0?finalcdr ") in new stack

– Executing [s@macro-hangupcall:6] Set (" SIP/future-0000000b ", " MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/2019/12/14/exten-8888-+60164219012-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:7] System (" SIP/future-0000000b ", " test -e /var/spool/asterisk/monitor/2019/12/14/exten-8888-+ 60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:8] NoOp (" SIP/future-0000000b ", " SYSTEMSTATUS = SUCCESS ") in new stack

– Executing [s@macro-hangupcall:9] GotoIf (" SIP/future-0000000b ", " 1?mmtestsuccess1 ") in new stack

– Goto (macro-hangupcall,s,11)

– Executing [s@macro-hangupcall:11] Set (" SIP/future-0000000b ", " CDR(recordingfile)=/var/spool/asterisk/monitor/2019/12/14/exten-8888-+ 60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:12] Goto (" SIP/future-0000000b ", " finalcdr ") in new stack

– Goto (macro-hangupcall,s,63)

– Executing [s@macro-hangupcall:63] NoOp (" SIP/future-0000000b ", " CDR recordingfile set to: /var/spool/asterisk/monitor/2019/12/14/exten-8888-+60164319019-20191214-190024-1576321224.11.wav ") in new stack

– Executing [s@macro-hangupcall:64] GotoIf (" SIP/future-0000000b ", " 1?skiprg ") in new stack

– Goto (macro-hangupcall,s,67)

– Executing [s@macro-hangupcall:67] GotoIf (" SIP/future-0000000b ", " 1?skipblkvm ") in new stack

– Goto (macro-hangupcall,s,70)

– Executing [s@macro-hangupcall:70] GotoIf (" SIP/future-0000000b ", " 1?theend ") in new stack

– Goto (macro-hangupcall,s,72)

– Executing [s@macro-hangupcall:72] AGI (" SIP/future-0000000b ", " hangup.agi ") in new stack

– Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi

– <SIP/future-0000000b>AGI Script hangup.agi completed, returning 0

– Executing [s@macro-hangupcall:73] Hangup (" SIP/future-0000000b ", “”) in new stack

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Gcsfuse & freepbx (FreePBX is unable to write to folder mounted with gcsfuse and the sync stops)

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@tbagalini wrote:

I am running freepbx Centos on a GCE VM. Syncing stops when FreePBX/Asterisk tries to write and create a folder in the mounted folder and FreePBX is unable to write / create folder in the folder. Everything works again when the system is rebooted because gcsfuse is no longer mounted.

Gcsfuse works when: .

-create a folder /var/spool/asterisk/cloud.
-mount it using gcsfuse --implicit-dirs siphqrecordings /var/spool/asterisk/cloud.
-create a sample text file nano test.txt.
-it syncs up to my bucket just fine

It completely stops working when:

-FreePBX tries to place a file and folder in the directory (a recorded call), the folder and file never gets written. I have unmount/reboot the server and then freepbx is able to use the folder and write files and directories to it. Even if I try to create a manual file using nano it never sync up.

FreePBX is unable to write to folder mounted with gcsfuse and the sync stops even for manually created files.

Thank you!

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Participants: 4

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Best way to broadcast a call in real time to a large amount of people (1000+)

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@bksales wrote:

This is a new one for me. They want to do their annual meeting virtually and want to use the PBX to have everyone be able to call in and listen while muted. Not sure what the practical and theoretical limits on the PBX are or what the best solution would be. I know we used the broadcast module last week to do 90 concurrent calls and that seemed to work fine, but I’m nervous about trying 1000 concurrent calls let alone 1000 members of a conference bridge.

Any ideas would be appreciated.

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Reload failed because retrieve_conf encountered an error: 139

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@waseemly wrote:

Hi

For some strange reason i have started to get the below error. It happens when i try and do apply config in freepbx frontend

I did delete some files in tmp the other day but to try and fix it i did create a new cron.error in tmp and set to ownership and grp to asterisk, still no joy

   Unable to continue. touch(): Unable to create file /tmp/cron.error because No such file or directory in /var/www/html/admin/libraries/BMO/Cron.class.php on line 279
    #0 [internal function]: Whoops\Run->handleError(2, 'touch(): Unable...', '/var/www/html/a...', 279, Array)
    #1 /var/www/html/admin/libraries/BMO/Cron.class.php(279): touch('/tmp/cron.error')
    #2 /var/www/html/admin/libraries/BMO/Cron.class.php(164): FreePBX\Cron->installCrontab(Array)
    #3 /var/www/html/admin/libraries/Builtin/UpdateManager.php(231): FreePBX\Cron->remove('11 4 * * * [ -e...')
    #4 /var/lib/asterisk/bin/retrieve_conf(1012): FreePBX\Builtin\UpdateManager->updateCrontab()
    #5 {main}

Also in asterisk CLI i get the following when doing apply config in freepbx

- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected

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Direct Dial To EXT Ringing Infinity!

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@toan_cr wrote:

I have trouble in case the customer(caller) call to Inbound then connecnt to IVR, the customer will connect directly to Ext(callee) they want to talk with Direct Dial Method Optiont in IVR.

Trouble happening when the Ext(callee) avalible and don’t pickup the phone call form Direct Dial of customer, the call will ring Ext ultil the customer(caller) hangup the phone. In this case if the customer don’t hangup the call the call will still open.
I think the Ringtime Default must be effect in this case to have the stop point for the call.
The Info of my system:
Asterisk Version: 13.22.0
FreePBX 14.0.13.12
FreePBX Framework 14.0.13.1
Core 14.0.28.19
IVR 14.0.9.4

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Gcsfuse & freepbx (FreePBX is unable to write to folder mounted with gcsfuse and the sync stops)

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@tbagalini wrote:

I am running freepbx Centos on a GCE VM. Syncing stops when FreePBX/Asterisk tries to write and create a folder in the mounted folder and FreePBX is unable to write / create folder in the folder. Everything works again when the system is rebooted because gcsfuse is no longer mounted.

Gcsfuse works when: .

-create a folder /var/spool/asterisk/cloud.
-mount it using gcsfuse --implicit-dirs siphqrecordings /var/spool/asterisk/cloud.
-create a sample text file nano test.txt.
-it syncs up to my bucket just fine

It completely stops working when:

-FreePBX tries to place a file and folder in the directory (a recorded call), the folder and file never gets written. I have unmount/reboot the server and then freepbx is able to use the folder and write files and directories to it. Even if I try to create a manual file using nano it never sync up.

FreePBX is unable to write to folder mounted with gcsfuse and the sync stops even for manually created files.

Thank you!

Posts: 11

Participants: 4

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FreePBX support for newer Cisco phones

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@cyr0nk0r wrote:

We are running the latest version of FreePBX.
We are looking to replace our older cisco small business phones with newer 8800 series 3PCC versions.
I am baffled at the lack of support for any modern Cisco phones on FreePBX.

Is there any updates on when 8800 series 3PCC or even 7800 series phones will be supported?

I mean, if FreePBX wants to exist in a market beyond just 10 person offices, why has there been such neglect on testing and certifying new templates for modern cisco phones? Especially since 3PCC versions have been out for several years.

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Gcsfuse & freepbx (FreePBX is unable to write to folder mounted with gcsfuse and the sync stops)

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@tbagalini wrote:

I am running freepbx Centos on a GCE VM. Syncing stops when FreePBX/Asterisk tries to write and create a folder in the mounted folder and FreePBX is unable to write / create folder in the folder. Everything works again when the system is rebooted because gcsfuse is no longer mounted.

Gcsfuse works when: .

-create a folder /var/spool/asterisk/cloud.
-mount it using gcsfuse --implicit-dirs siphqrecordings /var/spool/asterisk/cloud.
-create a sample text file nano test.txt.
-it syncs up to my bucket just fine

It completely stops working when:

-FreePBX tries to place a file and folder in the directory (a recorded call), the folder and file never gets written. I have unmount/reboot the server and then freepbx is able to use the folder and write files and directories to it. Even if I try to create a manual file using nano it never sync up.

FreePBX is unable to write to folder mounted with gcsfuse and the sync stops even for manually created files.

Thank you!

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Participants: 4

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FreePBX baremetal backup options

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@TEEP wrote:

Hello Back agian, my prevoius topic was closed, This time I was sick myself.
The system is running ok, despite the erors on statup.

Since this new system is running on bare metal i need some backup from the whole system AKA diskcopy.
I first tested clonezilla, many errors with the lvm disks. So first put this option aside for now
The tested with mongo http://www.mondorescue.org/ but could not find anything to install on centos 7 that works.
I also found Rear relax backup http://relax-and-recover.org/ , this looks wel maintned.
To test i changed the disk in my pbx server and installed fresh Freepbx iso

To test the rear backup i followed this howto https://www.linuxtechi.com/install-rear-migration-recovery-tool-centos-7-rhel-7/
First tip, first test the nsf mount to your nsf disk works with (after sudo systemctl restart nfs howto)

showmount -e <Ip or fdn of your NSF server>

After the first run i got an error :
WARNING: /usr/lib/grub/x86_64-efi/moddep.lst not found, grub2-mkimage will likel y fail. Please install the grub2-efi-x64-modules package to fix this.
ERROR: Error occurred during grub2-mkimage of BOOTX64.efi

After some research i found maybe solution

 sudo yum reinstall grub2-efi grub2-efi-modules  shim

still failure on creating iso backup

 sudo yum install grub2-efi-modules

Yes, it works on my NSF share created iso and tar file.
Tormorrow i will wipe de harddisk and try the backup.

So now a small question, is Rear relax good with freepbx ? Or are there better solutions

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Script for contacts on Yealink no longer works on FreePBX 15 (worked on FreePBX14)

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@chrischevy wrote:

Here is a script that I’ve been using for a while to populate the remote address book on yealink phones. I save it as contacts.php in the /var/html/www/ directory:

<?php
/*
The purpose of this file is to read all the Contact Manager entries for the specified group
and then output them in a Yealink Remote Address Book formatted XML syntax.
Instructions on how to use can be found here:
https://mangolassi.it/topic/18647/freepbx-contact-manager-to-yealink-address-book
*/
// Edit this varibale to match the name of hte group in Contact Manager
$contact_manager_group = "Contacts";
header("Content-Type: text/xml");
// get the MySQL/MariaDB login information from the amportal configuration file.
define("AMP_CONF", "/etc/amportal.conf");
$file = file(AMP_CONF);
if (is_array($file)) {
    foreach ($file as $line) {
        if (preg_match("/^\s*([a-zA-Z0-9_]+)=([a-zA-Z0-9 .&-@=_!<>\"\']+)\s*$/",$line,$matches)) {
            $amp_conf[ $matches[1] ] = $matches[2];
        }
    }
}
require_once('DB.php'); //PEAR must be installed
$db_user = $amp_conf["AMPDBUSER"];
$db_pass = $amp_conf["AMPDBPASS"];
$db_host = $amp_conf["AMPDBHOST"];
$db_name = $amp_conf["AMPDBNAME"];
$datasource = 'mysql://'.$db_user.':'.$db_pass.'@'.$db_host.'/'.$db_name;
$db = DB::connect($datasource); // attempt connection
$type="getAll";
// This pulls every number in contact maanger that is part of the group specified by $contact_manager_group
$results = $db->$type("SELECT cen.number, cge.displayname FROM contactmanager_group_entries AS cge LEFT JOIN contactmanager_entry_numbers AS cen ON cen.entryid = cge.id WHERE cge.groupid = (SELECT cg.id FROM contactmanager_groups AS cg WHERE cg.name = '$contact_manager_group');", null);
//dump the result into an array.
foreach($results as $result){
    $extensions[] = array($result[0],$result[1]);
}
// output the XML header info
echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n";
// Output the XML root. This tag must be in the format XXXIPPhoneDirectory
// You may change the word Company below, but no other part of the root tag.
echo "<CompanyIPPhoneDirectory  clearlight=\"true\">\n";
$index = 0;
if (isset($extensions)) {
    // Loop through the results and output them correctly.
    // Spacing is setup below in case you wish to look at the result in a browser.
    foreach ($extensions as $key=>$extension) {
        $index= $index + 1;
        echo "    <DirectoryEntry>\n";
        echo "        <Name>$extension[1]</Name>\n";
        echo "        <Telephone>$extension[0]</Telephone>\n";
        echo "    </DirectoryEntry>\n";
    }
}
// Output the closing tag of the root. If you changed it above, make sure you change it here.
echo "</CompanyIPPhoneDirectory>\n";
?>

It was working fine until I tried to use it on FreePBX 15 for the first time. The result is the same on 2 different system, an XML page with the same data on both systems. The following entries are done on some lines “B” D" “E” “S” “Array” “Array”

In order for the script to work, I have to install PHP-Pear by using the command “yum install -y php-pear-DB”

I’m trying to find out what is wrong with the script but I can’t figure it out (I’m not a very good programmer). Also, I think that the script works as intented since there is an XML array created when I browse the contacts.php page manually. The data “B” D" “E” “S” “Array” “Array” must be coming from somewhere

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