@HypnotiXDMP wrote:

So, we had a power outage at our DC and our generator did not kick on in time, and the server this specific PBX was on had to be powered back up. After powering it back up, I had all kinds of problems, which I believe I was able to resolve most of. The specific issue I am having now I cannot seem to clear up, and am unsure of what it means by it needs to be repaired as I am not even sure how to find the exact line it suggests. Below is the error that pops up, and this is after clicking the “Apply Config” button that will no go away.

exit: 255
Unable to continue. SQLSTATE[HY000]: General error: 1194 Table ‘trunk_dialpatterns’ is marked as crashed and should be repaired in /var/www/html/admin/modules/core/Core.class.php on line 1778
#0 /var/www/html/admin/modules/core/Core.class.php(1778): PDOStatement->execute()
#1 /var/www/html/admin/modules/core/functions.inc.php(4987): FreePBX\modules\Core->getAllTrunkDialRules()
#2 /var/www/html/admin/modules/core/functions.inc.php(1961): core_trunks_list_dialrules()
#3 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): core_do_get_config(‘asterisk’)
#4 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#5 {main}

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@mvogel4949 wrote:

Tomorrow I am going to setup a Dual NIC for the first time. Eth0 will plug into the local network and Eth1 will connect directly to an onsite SIP connection. Obviously in System Admin I will program eth0 and eth1 with the proper subnets and gateways.

Do I need to make changes at the root level to choose which eth is the default? Do I even need to set a default?

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@rnrstar wrote:

FreePBX 13.0.195.17
Asterisk 13.19

I’ve got a Freepbx HA configuration where it’s showing /dev/mapper/vg_ha-slash is going from 50% to 75% in a 12 hour period. I deleted a bunch of httpd error logs and that reduced it to 50% but less than 10 hours later it’s back to 71% except this time the log files are not the culprit. I’ve tried running du -kscx * from the root to see what folders are using a bunch of disk space but nothing shows that it can account for the vanishing disk space.

[root@freepbx-a log]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/mapper/vg_ha-slash
48249720 32481152 13310968 71% /
tmpfs 3975920 61448 3914472 2% /dev/shm
/dev/md0 487588 32689 429303 8% /boot
//192.168.66.21/PBX 9362775284 6574903120 2787872164 71% /mnt/qnap
/dev/drbd2 47449536 9832780 35199784 22% /drbd/mysql
/dev/drbd4 9387332 21264 8882552 1% /drbd/spare
/dev/drbd3 37930460 1449748 34547260 5% /drbd/httpd
/dev/drbd1 85511988 27529636 53631924 34% /drbd/asterisk
[root@freepbx-a log]# cd /
[root@freepbx-a /]# du -kcsx *
361224 backup.tftpboot20180525
6928 bin
30411 boot
284 dev
20 drbd
61552 etc
498960 home
4 installed-version
265164 lib
26456 lib64
16 lost+found
4 media
8 mnt
755444 opt
du: cannot access proc/5495/task/5597/fd/198': No such file or directory du: cannot accessproc/5495/task/5597/fd/333’: No such file or directory
du: cannot access proc/5495/task/5610/fd/198': No such file or directory du: cannot accessproc/5495/task/5610/fd/363’: No such file or directory
du: cannot access proc/5495/task/5610/fd/364': No such file or directory du: cannot accessproc/5495/task/5011/fdinfo/364’: No such file or directory
du: cannot access proc/5495/task/5651/fd/363': No such file or directory du: cannot accessproc/5495/task/5651/fdinfo/198’: No such file or directory
du: cannot access proc/5495/task/26381/fd/365': No such file or directory du: cannot accessproc/5495/task/26381/fdinfo/198’: No such file or directory
du: cannot access proc/5495/task/26382/fd/363': No such file or directory du: cannot accessproc/5495/task/26395/fd/663’: No such file or directory
du: cannot access proc/5495/task/26395/fd/674': No such file or directory du: cannot accessproc/27010/task/27010/fd/4’: No such file or directory
du: cannot access proc/27010/task/27010/fdinfo/4': No such file or directory du: cannot accessproc/27010/fd/4’: No such file or directory
du: cannot access `proc/27010/fdinfo/4’: No such file or directory
0 proc
4 repair
423252 root
12852 sbin
4 selinux
4 srv
0 sys
0 tftpboot
349864 tmp
4 uploads
1699616 usr
1597048 var
6089123 total

Any suggestions as to what to look for?

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@sentinelace wrote:

I am using zulu. Works great for all my internet based apps. I have a local application that I was told I needed the CRM link. I purchased that. My app is not an option, so I assume I use resetAPI ? I did that and still doesn’t work when I click on the number. What am I missing?

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@dannyprecise wrote:

I have a list I need to import in the outbound route.

If I wanted to route any calls that started with 201555 where would I enter that and Do I need any other characters like X N. Do I enter it in the prepend,prefix,pattern match?

Here are a few others that I will be imported once I know where to enter it into the dial pattern
201555
202555
203555
205555
206555
207555
207835
208555
208900
209273
209332
209425
209427
209446
209447
209454
209555
209949
210555
210840
210899
212550
212555
212970
213357
213555

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@xtranetcc wrote:

Hi Community,

FreePBX 14.0.4.1
PBX Firmware: 12.7.5-1807-1.sng7
PBX Service Pack: 1.0.0.0

I have a newly installed server with an IVR setup.

I have a problem today where a hacker has bee exploiting the *2 and ## feature codes.
They call in and dial *2destination# to make out free calls via our Pabx.

I have tried the following from a previous post which does not work. Those feature codes still remain active
although they are disabled in the FreePBX GUI:

I have also tried disabling *2 and ## feature codes on previous installations and when i dial in while the ivr plays and press *2destionation # the call is successfull.

This does not make sense as the feature codes have been disabled.

Please advise.

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@inatechsol wrote:

I did my normal Module Admin update and after the update we have not been able to make any outbound calls. here is the full log message when i dial

[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [4042566505@from-internal:1] Macro(“SIP/290-0000000b”, “user-callerid,LIMIT”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/290-0000000b”, “TOUCH_MONITOR=1541265885.17”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/290-0000000b”, “AMPUSER=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/290-0000000b”, “0?report”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/290-0000000b”, “1?Set(REALCALLERIDNUM=290)”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/290-0000000b”, “AMPUSER=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/290-0000000b”, “0?limit”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/290-0000000b”, “AMPUSERCIDNAME=Maulik Patel”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:8] ExecIf(“SIP/290-0000000b”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:9] GotoIf(“SIP/290-0000000b”, “0?report”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/290-0000000b”, “AMPUSERCID=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:11] Set(“SIP/290-0000000b”, “__DIAL_OPTIONS=Ttr”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:12] Set(“SIP/290-0000000b”, “CALLERID(all)=“Maulik Patel” <290>”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:13] GotoIf(“SIP/290-0000000b”, “0?limit”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“SIP/290-0000000b”, “1?Set(GROUP(concurrency_limit)=290)”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:15] ExecIf(“SIP/290-0000000b”, “0?Set(CHANNEL(language)=)”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:16] NoOp(“SIP/290-0000000b”, “Macro Depth is 1”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:17] GotoIf(“SIP/290-0000000b”, “1?report2:macroerror”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (macro-user-callerid,s,18)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:18] GotoIf(“SIP/290-0000000b”, “1?continue”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (macro-user-callerid,s,37)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/290-0000000b”, “CALLERID(number)=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/290-0000000b”, “CALLERID(name)=Maulik Patel”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:39] GotoIf(“SIP/290-0000000b”, “0?cnum”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:40] Set(“SIP/290-0000000b”, “CDR(cnam)=Maulik Patel”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:41] Set(“SIP/290-0000000b”, “CDR(cnum)=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@macro-user-callerid:42] Set(“SIP/290-0000000b”, “CHANNEL(language)=en”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [4042566505@from-internal:2] Set(“SIP/290-0000000b”, “ROUTEUSER=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [4042566505@from-internal:3] Set(“SIP/290-0000000b”, “ROUTEUSER=290”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [4042566505@from-internal:4] GotoIf(“SIP/290-0000000b”, “1?notblind”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (from-internal,4042566505,7)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [4042566505@from-internal:7] GotoIf(“SIP/290-0000000b”, “1?restrictedroute-cfcd208495d565ef66e7dff9f98764da,4042566505,2:outbound-allroutes,4042566505,2”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (restrictedroute-cfcd208495d565ef66e7dff9f98764da,4042566505,2)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Channel ‘SIP/290-0000000b’ sent to invalid extension: context,exten,priority=restrictedroute-cfcd208495d565ef66e7dff9f98764da,4042566505,2
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [i@restrictedroute-cfcd208495d565ef66e7dff9f98764da:1] Goto(“SIP/290-0000000b”, “bad-number,s,1”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (bad-number,s,1)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [s@bad-number:1] Goto(“SIP/290-0000000b”, “11,1”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx_builtins.c: Goto (bad-number,11,1)
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [11@bad-number:1] ResetCDR(“SIP/290-0000000b”, “”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [11@bad-number:2] NoCDR(“SIP/290-0000000b”, “”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [11@bad-number:3] Progress(“SIP/290-0000000b”, “”) in new stack
[2018-11-03 13:24:45] VERBOSE[11844][C-0000003e] pbx.c: Executing [11@bad-number:4] Wait(“SIP/290-0000000b”, “1”) in new stack
[2018-11-03 13:24:46] VERBOSE[11844][C-0000003e] pbx.c: Executing [11@bad-number:5] Playback(“SIP/290-0000000b”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2018-11-03 13:24:46] VERBOSE[11844][C-0000003e] file.c: <SIP/290-0000000b> Playing ‘silence/1.ulaw’ (language ‘en’)
[2018-11-03 13:24:47] VERBOSE[11844][C-0000003e] file.c: <SIP/290-0000000b> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2018-11-03 13:24:50] VERBOSE[11844][C-0000003e] file.c: <SIP/290-0000000b> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2018-11-03 13:25:13] VERBOSE[2127][C-00000040] netsock2.c: Using SIP RTP TOS bits 184
[2018-11-03 13:25:13] VERBOSE[2127][C-00000040] netsock2.c: Using SIP RTP CoS mark 5
[2018-11-03 13:25:13] VERBOSE[11924][C-00000040] pbx.c: Executing [14042566505@from-internal:1] Macro(“SIP/290-0000000c”, “user-callerid,LIMIT”) in new stack
[2018-11-03 13:25:13] VERBOSE[11924][C-00000040] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/290-0000000c”, “TOUCH_MONITOR=1541265913.18”) in new stack
[2018-11-03 13:25:13] VERBOSE[11924][C-00000040] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/290-0000000c”, “AMPUSER=290”) in new stack
[2018-11-03 13:25:13] VERBOSE[11924][C-00000040] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/290-0000000c”, “0?report”) in new stack

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@mixhali wrote:

Hi all, some background first.

In Australia we are all being forced to use the governments high speed internet service called NBN, if you are establishing new services in a new office for example this is your only option. This caused an issue for me as the new NBN service only supported 4 PSTN lines and I needed 8. So I decided to try switching to a sip based solution and roll out asterisk.

I purchased a trunk from our local provide called Engin. A friend of mine gave me 20 older cisco phones from his office 7961G. I spent a month pulling my hair out with it but managed to finally get it working nicely.

I now have one annoying problem, every so often we try to call someone and the line hangs up after one or two rings. After checking logs I found this which I believe is causing the issue.

[2018-11-04 03:30:47] WARNING[3949][C-00000899]: channel.c:5740 set_format: Unable to find a codec translation path: (g729) -> (ulaw)
[2018-11-04 03:30:47] WARNING[3949][C-00000899]: channel.c:5740 set_format: Unable to find a codec translation path: (ulaw) -> (g729)

I have logged a few topics on this in varying forms and people have suggested just switching off support for g729 or buying licencs and installing the codec and even asking why I need it.

The ansers in order:
tried switching off support on my sip settings and disallow the codec on my extension - -Same issue
I downloaded the codec, Cant install it. See my other thread on this one

As to why I need it, this thread is nswering that question unless someone can tell me how I can circumvent the need in the first place.

I really don’t understand how I can take hundreds a calls a day on the same channel and it works fine however some specific numbers require this codec, why would that be. I would have thought that my sip provider would always talk to me with the same codec. Do you think I can call them and ask them not to use g729, or is it outside of their control? I’m thinking maybe these particular numbers are using sip also and prefer to use g729 and that is just being passed through by my trunk provider.

Appreciate any thoughts.

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@digitalo wrote:

Hi Guys,

I’m going crazy over this, I have a custom context for my trunk which is working fine.
It strips off the country code for calls from my country, removes the plus for international calls and adds one or two zero’s.
However if a call comes in as anonymous is results in 0anonymous.
I tried several approaches but none of them worked for all cases.
Can anyone point me in the right direction here? Tnx!

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@digitalo wrote:

New system, only updated the base and all the modules but zulu won’t start.
Already tried deleting and reinstalling but to no avail.

fwconsole start zulu
Running PBXact startup...
Setting Permissions...
Setting specific permissions...
 22800 [============================]
Finished setting permissions
Unable to run Pre-Asterisk hooks, because Asterisk is already running on PID 22933 and has been running for 7 hours, 37 minutes, 34 seconds
Running Asterisk post from Zulu module

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@steve_pbuk wrote:

Currently my call history goes back to when I first set the server up in January 2018.

Is it possible to enable something that limits how long call history stays in the database?

Thanks

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@dannyprecise wrote:

I have both enabled

Automatic System Updates
.Automatic Module Updates

On the System update tab it shows it hasnt checked for 3 weeks but I have the schedule set to run every Sat. If I manually check it will check, download and update fine if there is one. Same for the module updates as well.

Current System Update Status:

Idle

Refresh page

Last Online Check Status:

3 weeks ago (Complete)

Last System Update:

4 weeks ago (Complete)

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@Alkbert wrote:

Greetings. I would like to know if it’s possible to create an automatic conference room using the Bridge() app. This is what I want to do: when a user XX calls the number YYY, automatically some extensions are called and bridged all: the caller and the callings parties. This will need some code stuff but I’m still thinking how to start. Any ideas?

Also, is Asterisk capable of automatically calling 2 numbers and bridging them together by itself ?

Thanks.

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@dannyprecise wrote:

What is the basic setting to push a caller to a specific VM box after X time or if no agent is avail?

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@Paulbzero wrote:

We have a situation in our office where someone is on call during the early morning hours, and late evenings. During a 24-hour period, the call forwarding would go something like this:
00:00-08:00: Route to Shift 1’s cell phone
08:00-17:00: Route to IVR
17:00-24:00: Route to Shift 2’s cell phone

The problem is the people who are on call for Shift 1 and Shift 2 vary from day to day.
I’m looking for something that will automatically forward calls based on the system time.

I know I can set up time conditions, but am looking for something a little easier for a novice to set up and the fact that different people are on call with different cell numbers makes things more difficult. Maybe set up a calendar of events? Also it would be helpful if we can “pre-load” a number of days in advance.

Hopefully someone has run across this situation before, and has a module/script that would work for this situation.

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@daniel01901 wrote:

Hi,

I’m having issues running the FreePBX Distro Conversion Tool when trying to move our old 32-bit FreePBX install to a new 64-bit distro.

Below is output from the new server.

curl -s https://convert.freepbx.org | bash
Checking that ‘curl’ exists … OK!
Validating sha256 integrity … OK!
Trying to download converter to /tmp/tmp.XR9rZNYwNR … Complete!
Validating download … OK!
Starting FreePBX Converter version release/20171122r1
Testing connectivity to Conversion server…Success!

FreePBX Conversion Wizard

The FreePBX Conversion Wizard needs to be run on two machines, firstly on the
NEW machine, which must be an ACTIVATED FreePBX Distro machine, and then it
must be run on the DONOR machine.
The DONOR machine is the machine that is currently processing calls, and is
the machine that will be migrated to the NEW machine. No changes will be made
to the DONOR machine, and this script will not stop or restart any services
that may cause an outage.
If this is the NEW machine, just push ‘Enter’ to prepare this machine

Enter Conversion ID (leave blank if this is NEW):
Testing FreePBX functionality … Success! (Version 14.0.4.5)
Getting Deployment … 35828561
Getting all Module versions (for conversion) … OK
Reserving a conversion slot … Reserved!

The Conversion process is now ready. Please run the script on the
DONOR node now, and when asked for a slot identifier, please enter
the following ID:

    e4619eab-d65c-4083-a40c-7340d0cd0339

[/] Donor now sending module ‘contactmanager’, table ‘contactmanager_entry_images’ …^CCleaning up…Done!

Below is output from the donor machine.

[root@pbx ~]# curl -s https://convert.freepbx.org | bash
Checking that ‘curl’ exists … OK!
Validating sha256 integrity … OK!
Trying to download converter to /tmp/tmp.kGIne0GRY6 … Complete!
Validating download … OK!
Starting FreePBX Converter version release/20171122r1
Testing connectivity to Conversion server…Success!

FreePBX Conversion Wizard

The FreePBX Conversion Wizard needs to be run on two machines, firstly on the
NEW machine, which must be an ACTIVATED FreePBX Distro machine, and then it
must be run on the DONOR machine.
The DONOR machine is the machine that is currently processing calls, and is
the machine that will be migrated to the NEW machine. No changes will be made
to the DONOR machine, and this script will not stop or restart any services
that may cause an outage.
If this is the NEW machine, just push ‘Enter’ to prepare this machine

Enter Conversion ID (leave blank if this is NEW): e4619eab-d65c-4083-a40c-7340d0cd0339
Testing FreePBX functionality … Success!
Checking Slot ID … OK!
Getting modules to convert … Complete! 114 modules
Dumping astdb … Complete!

— WARNING — WARNING — WARNING — WARNING —

Some directories are large, and may slow down the backup. You can copy these
directories across later, after the conversion is complete. This will speed up
the conversion. Answering 'Y’es will embed them in the backup that is sent to
the new machine

Directory /var/spool/asterisk/backup? (15.39GB) [yN] y
Directory /tftpboot? (774.72MB) [yN] y

Creating encrypted backup for new machine. This may take some time.
Backing up the following:
du: cannot access 9': No such file or directory du: cannot accesst’: No such file or directory
Complete!
Sending backup details to new machine … Complete!
Skipping module accountcodepreserve (Nothing to convert)
Skipping module amd (Nothing to convert)
Module announcement … announcement
Module areminder … areminder areminder_calls areminder_settings areminder_updates
Module arimanager … arimanager
Skipping module asterisk-cli (Nothing to convert)
Skipping module asteriskinfo (Nothing to convert)
Skipping module backup (Nothing to convert)
Skipping module blacklist (Nothing to convert)
Module broadcast … broadcast_campaigns broadcast_campaign_groups broadcast_settings broadcast_groups broadcast_callees broadcast_log
Skipping module builtin (Nothing to convert)
Skipping module bulkhandler (Nothing to convert)
Skipping module calendar (Nothing to convert)
Module callback … callback
Module callerid … callerid_entries
Skipping module callforward (Nothing to convert)
Module calllimit … calllimit calllimit_usage
Module callrecording … callrecording callrecording_module
Skipping module callwaiting (Nothing to convert)
Skipping module campon (Nothing to convert)
Skipping module cdr (Nothing to convert)
Skipping module cel (Nothing to convert)
Skipping module certman (Nothing to convert)
Module cidlookup … cidlookup cidlookup_incoming
Module conferences … meetme
Skipping module conferencespro (Nothing to convert)
Skipping module configedit (Nothing to convert)
Module contactmanager … contactmanager_groups contactmanager_group_entries contactmanager_entry_numbers contactmanager_entry_images contactmanager_entry_userman_imagesERROR
Unable to continue.
Cleaning up…Done!

Does anyone know why it could be failing on contactmanager_entry_userman_images ?

Thanks

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@anscomputer wrote:

Hello,

I have a customer who needs to call the number 1927
But the number ring busy.
I think this is because the freepbx try to call an extension and not an external number.
Someone know a solution to call the 4digits number ?

Best regards,
Luca

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@Tommaso wrote:

Hello! i’m not into everything involved around FreePBX but i managed to set up some soft/phisical phones to dialog internally and to make outbound calls,so now the problem is that i really cant get incoming calls!
if i call the number, PBX say “the number you chose is not in use”

Trough the Asterisk Info page, i see that there a re A LOT of stuffs that definitly should not be like that, first of all, my gateway for POTS line say “Rejected” (TA410 with IP 192.168.0.130)

Here i’ll leave some screenshots, hope that someone can help me and thanks already for your help!

screen.png1920x1040 54.6 KB

screen (2).png1920x1040 59.6 KB

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@PitzKey wrote:

Hello everyone.

We recently started using the Login Feature (Hot Desking) with the Sangoma phones, it works wonderful!

However, per the documentation, you can map an extension to just a template, and once you login from any phone it will pull the config from the template the extension is mapped.

I’m trying to do so, but I get an error “Please select model before proceeding with submission…”

image.png1294x516 34.1 KB

And the same thing happens when I leave “Select Account” blank.

OK, so I added 101, set account 1, model s500. Mac = Blank.

But when I want to add a second extension with no MAC, I get the following error.

image.png1291x585 42.6 KB

I’m on the latest version of EPM and all modules are updated.

Am I missing something? Or this is a bug?

Thanks

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@mvogel4949 wrote:

Using FreePBX14 along with the most recent version of Directory 13.0.19.5. I have a recorded name for my VM but even if I have voicemail greeting chosen in the directory it still tries to spell the name.

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