@no_lifer wrote:

Hello,

I know much people already made a topic about it but I think I’m a bit different than the other people and searching for a day didn’t really help me to fix the issue so far.

Anyways my settings:

• FreePBX : 14.0.1.24
• Asterisk: 15.1.4

The issue here is the following:

I installed FreePBX before on the recommended settings but I got the same.
I upload or make the recordings and they are playing back by a browser without any problem.
When I try to play the recording through IVR or Play recording directly from incoming it gives me this error:

[2018-04-06 13:53:25] VERBOSE[16447][C-0000000b] pbx.c: Executing [2@play-system-recording:2]             Playback(“SIP/77709012579-00000009”, “custom/test”) in new stack
[2018-04-06 13:53:25] WARNING[16447][C-0000000b] file.c: File custom/test does not exist in any format
[2018-04-06 13:53:25] WARNING[16447][C-0000000b] file.c: Unable to open custom/test (format (alaw)): No such file or directory
[2018-04-06 13:53:25] WARNING[16447][C-0000000b] app_playback.c: Playback failed on SIP/77709012579-00000009 for custom/test

so I changed the permissions tried somethings and I found out that when I play it through Languages and then selecting the recording it works perfectly…

Now I want to use IVR and that won’t let me use language.

can someone help me out to find a fix for it?

Greets,
Joris Dijkstra

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@byrnejb wrote:

FreePBX 13.0.192.16
Asterisk 13.17.0

We setup our FreePBX system to automatically forward calls received after hours off site to a 1-800 line via analogue POTS. And we created an announcement to tell callers that this is what is happening.

We used the Time Conditions and Time Groups applications to set up the after-hours checks. We used the System Recordings module to create the sound file that we wish played. We used the Miscellaneous Destination application to provide the target 1-800 number. We used the the Announcement application to play the sound file previously created and to subsequently go to the Misc. Dest., likewise previously defined. And we used the Time Conditions application to transferl to the after-hours announcement at the appropriate times.

The above simply provides the details of the setup process followed; both as an aide-memoire and to allow review for any defect or omission. The resulting setup appears to work as expected.

We have a few details that we would like to have smoothed out.

1. Calls forwarded by this setup have an excruciatingly low volume for both parties. Is there any way to turn up the gain in both directions?

2. There is a considerable delay in connecting off-site after the announcement has played and this is simply dead air. Is there any way to have the dialling start before the announcement has finished and to interrupt it if the line is picked up while playing?

3. There is no ring tone from the time the announcement completes until the off-site line is answered. Is there a way to provide this?

4. If the number is busy then the call is simply dropped. There is no busy signal returned to the in-call. Is there a way to configure our set up to report the busy condition; even if that involves another announcement?

Thanks,

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@joelwhrs wrote:

I am having a very weird echo with a single client that has a SIP trunk with VoIP Innovations. They can randomly hear an echo of themselves. The person on the other end has crystal clear audio with no echo. There doesn’t seem to be any rhyme or reason as to what types of calls echo, as it happens for other people on SIP trunks, analog lines, and cell phones. I know that echos are usually associated with calling analog lines, yet it definitely isn’t limited to this as they even had an echo when calling us (we are on VoIP Innovations as well). When a call is established between 2 VoIP Innovations clients, the RTP traffic is routed between the 2 endpoint RTPs without being relayed by the provider, which rules out any issues on the providers end.

I got a packet capture while I was on the phone with them and they said they had a very slight echo. I checked the packets and sure enough, I could ever so slightly hear the caller on the callers inbound RTP stream (me talking). I did not hear any echo on my end while I was on the phone. The packets only show a max jitter of 9.67ms and an average of .37ms.

The PBX is NAT’d behind a Sophos XG firewall with QOS enabled and priority given to SIP and RTP. This client is setup exactly the same as our other clients, with the exception of the Sophos XG UTM. Our other clients are using Sophos UTM 9 products. We’ve never experienced this issue before.

Any ideas on next steps? I’m getting towards the end of my abilities on this one.

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@helpaircare wrote:

Wondering if anyone can help me out… My “Call Recording Reports” module doesn’t display any recordings at all. It just sits there at “Loading, please wait…” and never loads a list of recordings. I FTP’d in through filezilla and deleted ALL recordings, thinking one of them might be corrupt, but now a few days later the problem persists. Seems to be frontend related. Any ideas what I can look at that might be broken ?

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@kd2dwj wrote:

When trying to open certain modules, like extensions I get the following error

PDOException (HY000)
SQLSTATE[HY000]: General error: 1 Can’t create/write to file ‘/var/tmp/#sql_4d0_0.MAI’ (Errcode: 2)

Any ideas how to fix this issue?

TIA

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@MiskoR wrote:

Hello all,

Running freepbx ver 14.0.2.14 on rpi3, asterisk ver 13.20.0

After troublesome xmpp installation (mongodb failed to start after installation, due to the limited sd space) I am getting these errors when I try to login from UCP web chat client:

2018-04-06 18:01 +02:00:
██╗ ███████╗████████╗███████╗ ██████╗██╗ ██╗ █████╗ ████████╗
██║ ██╔════╝╚══██╔══╝██╔════╝ ██╔════╝██║ ██║██╔══██╗╚══██╔══╝
██║ █████╗ ██║ ███████╗ ██║ ███████║███████║ ██║
██║ ██╔══╝ ██║ ╚════██║ ██║ ██╔══██║██╔══██║ ██║
███████╗███████╗ ██║ ███████║ ╚██████╗██║ ██║██║ ██║ ██║
╚══════╝╚══════╝ ╚═╝ ╚══════╝ ╚═════╝╚═╝ ╚═╝╚═╝ ╚═╝ ╚═╝

Release 0.4.90

2018-04-06 18:05 +02:00: User: 101Authenticated against freepbx!
2018-04-06 18:05 +02:00: Creating user: { email: ‘xxx@xyz.com’,
password: ‘xyz’,
firstName: ‘101’,
lastName: ‘101’,
username: ‘101’,
displayName: ‘abc’,
freepbxId: ‘4’,
uuid: undefined }
2018-04-06 18:05 +02:00: ERROR SAVING NEW USER: { ValidationError: User validation failed: password: Path password is invalid (101).
at MongooseError.ValidationError.inspect (/var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/error/validation.js:57:23)
at formatValue (util.js:351:36)
at inspect (util.js:185:10)
at exports.format (util.js:129:20)
at Console.log (console.js:43:37)
at /var/www/html/admin/modules/xmpp/node/node_modules/lets-chat-freepbx-auth/lib/auth.js:65:21
at /var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/model.js:4074:16
at /var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/services/model/applyHooks.js:175:17
at _combinedTickCallback (internal/process/next_tick.js:73:7)
at process._tickDomainCallback (internal/process/next_tick.js:128:9)
errors:
{ password:
{ Path password is invalid (101).
at MongooseError.ValidatorError (/var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/error/validator.js:25:11)
at validate (/var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/schematype.js:782:13)
at /var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/schematype.js:829:11
at Array.forEach (native)
at SchemaString.SchemaType.doValidate (/var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/schematype.js:789:19)
at /var/www/html/admin/modules/xmpp/node/node_modules/mongoose/lib/document.js:1548:9
at _combinedTickCallback (internal/process/next_tick.js:73:7)
at process._tickDomainCallback (internal/process/next_tick.js:128:9)
message: ‘Path password is invalid (101).’,
name: ‘ValidatorError’,
properties: [Object],
kind: ‘regexp’,
path: ‘password’,
value: ‘xyz’,
reason: undefined,
‘$isValidatorError’: true } },
_message: ‘User validation failed’,
name: ‘ValidationError’ }

It looks to me that mongo db structure was not initialized properly but I do not know how to create it…

Any help will be appreciated…

MiskoR

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@AHF wrote:

Hi,

I’m trying to clean up a poorly-documented phone system running FreePBX 12.0.76.4*. Part of that is going to involve sorting through the hundred or so DIDs and finding out which ones route to no-longer-used extensions. Rather than clicking through each inbound route one at a time, is there a report or page that will give me each DID or inbound route and its destination(s)?

*To be upgraded later, but there’s a lot of old business-critical processes that involve hastily-written scripts across multiple servers with no documentation and no sanity-checking. I want to make sure the server room won’t spontaneously explode if I change software versions.

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@Ducktour wrote:

image.jpg1029x582 149 KB

This morning’s upgrade freezes at FreePBX Framework 14.0.2.10 will be upgraded to online version 14.0.2.14
Thoughts?

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@razaqad wrote:

Hello.

I have two freepbx servers.
Server A and server B.
Both are inside the same lan 192.168.0.0/24
No nat involved.

Server A ip 192.168.0.231
extensions 101-150

Server B ip 192.168.0.186
extensions 201-250

Freepbx and asterisk version 14.

Already connected via sip trunk for internal calling between the two servers.

Both servers have dahdi trunks. No sip trunk for calling outside world.

What i want to achive is, how do i use dahdi trunk of server A to call outside world, from extension of server B. for example extenion 201 dials my mobile number and the call is dialed via dahdi trunk of server A.

Thanks

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@adolfoc wrote:

I’ve wiped my system, reinstalled v13, restored from backup, attempted the upgrade to v14 again, and I keep getting this same error. FreePBX (or fwconsole will not start).

PHP Fatal error:  Incompatible file format:  The encoded file has format major ID 4, whereas the Loader expects 7 in /var/www/html/admin/modules/directorypro/enc/functions.inc.php on line 0
Whoops\Exception\ErrorException: Incompatible file format:  The encoded file has                                    format major ID 4, whereas the Loader expects 7 in file /var/www/html/admin/mod                                    ules/directorypro/enc/functions.inc.php on line 0
Stack trace:
  1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/directorypr                                     o/enc/functions.inc.php:0
PHP Fatal error:  Incompatible file format:  The encoded file has format major ID 4, whereas the Loader expects 7 in /var/www/html/admin/modules/directorypro/enc/functions.inc.php on line 0
Whoops\Exception\ErrorException: Incompatible file format:  The encoded file has                                     format major ID 4, whereas the Loader expects 7 in file /var/www/html/admin/mod                                     ules/directorypro/enc/functions.inc.php on line 0
Stack trace:
  1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/directorypr                                     o/enc/functions.inc.php:0

any ideas?

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@junalmeida wrote:

Cannot make calls between extensions on freepbx. Here is my environment:
freepbx and asterisk running in a docker, at a simple ubuntu machine on azure.

5060, 5160 and 10000-20000 ports are forwarded from the public ip to vm, and from vm to docker.

Here is the log of a call:

-- Executing [s@macro-dial-one:51] NoOp("SIP/400-000030f4", "") in new stack
-- Executing [s@macro-dial-one:52] Dial("SIP/400-000030f4", "SIP/500,,HhTtrIb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- SIP/500-000030f5 Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] NoOp("SIP/500-000030f5", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:2] Set("SIP/500-000030f5", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:3] While("SIP/500-000030f5", "0") in new stack
-- Jumping to priority 6
-- Executing [s@func-apply-sipheaders:7] Return("SIP/500-000030f5", "") in new stack
  == Spawn extension (from-internal, 500, 1) exited non-zero on 'SIP/500-000030f5'
-- SIP/500-000030f5 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
-- Called SIP/500
-- Connected line update to SIP/400-000030f4 prevented.
-- Connected line update to SIP/400-000030f4 prevented.
-- SIP/500-000030f5 is ringing
   > 0x7fe08c009f90 -- Strict RTP learning after remote address set to: 10.0.4.150:16400
-- Connected line update to SIP/400-000030f4 prevented.
-- SIP/500-000030f5 answered SIP/400-000030f4
-- Channel SIP/500-000030f5 joined 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
-- Channel SIP/400-000030f4 joined 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
[2018-04-08 10:28:03] NOTICE[5152]: chan_sip.c:29611 check_rtp_timeout: Disconnecting call 'SIP/400-000030f4' for lack of RTP activity in 31 seconds
[2018-04-08 10:28:03] NOTICE[5152]: chan_sip.c:29611 check_rtp_timeout: Disconnecting call 'SIP/500-000030f5' for lack of RTP activity in 31 seconds
-- Channel SIP/400-000030f4 left 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
-- Channel SIP/500-000030f5 left 'simple_bridge' basic-bridge <91c72704-da14-4652-a996-cdadcc492d49>
  == Spawn extension (macro-dial-one, s, 52) exited non-zero on 'SIP/400-000030f4' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/400-000030f4' in macro 'exten-vm'
  == Spawn extension (from-internal, 500, 2) exited non-zero on 'SIP/400-000030f4'
-- Executing [h@from-internal:1] Macro("SIP/400-000030f4", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/400-000030f4", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/400-000030f4", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("SIP/400-000030f4", "SIP/500-000030f5 monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("SIP/400-000030f4", "attendedtransfer-rec-restart.php,SIP/500-000030f5,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <SIP/400-000030f4>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("SIP/400-000030f4", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/400-000030f4' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/400-000030f4'
[2018-04-08 10:28:04] WARNING[5152]: chan_sip.c:4077 retrans_pkt: Retransmission timeout reached on transmission lC2NwJbGosBXLyvMRQHgUQ.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

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@cazimmerman wrote:

I would like our staff to be able to make calls through the voicemail System (ie: when they check their v/m if they need to return a call with our caller id). They have the option to make outgoing calls under the Advanced Option however, the message says “Please wait while I connect your call” and then nothing happens. Here is what the log is showing:

[2018-04-07 17:48:49] WARNING[13012][C-00000014] pbx.c: Channel ‘PJSIP/10284-00000016’ sent to invalid extension but no invalid handler: context,exten,priority=yes,10214,1
‘from-internal’

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@rmblakes wrote:

Hi All,

Hoping someone can help me with this one…

I cannot get the outbound or inbound audio to work until I put the hold music.

Here’s my current setup details:

Raspberry Pi, FreePBX 14.0.2.10
Sophos UTM with SIP Helper and NAT UDP 5060 > FreePBX Box.
1 x iPhone 7, Zoiper Free, logged in as Test
1 x Samsung Galaxy S9, Zoiper Free, logged in as User1

---

Dial (Using 4G Cell network) Test > User1 =GOOD
Dial (Using 4G cell network) Test > Landline = No two way audio (cannot hear either party) until HOLD button pressed on Zoiper, then audio works.

Dial (Using internal wifi network with Freepbx on Subnet) Test > User1 = GOOD
Dial (Using internal wifi network with Freepbx on Subnet) Test > Landline = GOOD

Anything I should be trying next? Why would putting the call on hold allow the voice to work?

Thanks

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@cazimmerman wrote:

When I turn it on external calls ring two times and then go to voicemail. This happens even if you try to answer them on the first ring. It it on the primary extension and it is set to ringallv2.

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@Connex wrote:

Hello everyone!

I’m trying to configure so that the the url woulth automatically redirect to https.

I added this in etc/httpd/conf.d/freepbx.conf

<VirtualHost :80>
RewriteEngine on
RewriteCond %{HTTPS} !=on [NC]
RewriteRule ^/admin(.)$ https://%{HTTP_HOST}%{REQUEST_URI} [R=301,L]

It works but it only redirects the admin panel. It won’t redirect the ucp page.

Is there anything else that I need to add?

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@np_pyro wrote:

I have a server currently running the distro version 10.13.66-20.

When reloading the config, I receive the following errors in freepbx.log:

[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/restapi/ari/modules/restapi.module to /var/www/html/recordings/modules/restapi.module, but /var/www/html/recordings/modules doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/modules/webrtcphone.module to /var/www/html/recordings/modules/webrtcphone.module, but /var/www/html/recordings/modules doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/theme/webrtcsounds to /var/www/html/recordings/theme/webrtcsounds, but /var/www/html/recordings/theme doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/theme/webrtcimages to /var/www/html/recordings/theme/webrtcimages, but /var/www/html/recordings/theme doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/theme/webrtc.css to /var/www/html/recordings/theme/webrtc.css, but /var/www/html/recordings/theme doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/js/jssip-devel.js to /var/www/html/recordings/theme/js/jssip-devel.js, but /var/www/html/recordings/theme/js doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/js/phone.js to /var/www/html/recordings/theme/js/phone.js, but /var/www/html/recordings/theme/js doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/js/webrtc.js to /var/www/html/recordings/theme/js/webrtc.js, but /var/www/html/recordings/theme/js doesn't exist
[2018-Apr-09 14:01:51] [ERROR] (bin/retrieve_conf:291) - Tried to link /var/www/html/admin/modules/webrtc/ari/js/adaptor.js to /var/www/html/recordings/theme/js/adaptor.js, but /var/www/html/recordings/theme/js doesn't exist

What should have created the /var/www/html/recordings/…(misc dirs)… and what is the best way to remedy this? Obviously I could create each directory manually, but is that really solving the root of what the problem is?

I did confirm that the /var/www/html/recordings directory is indeed missing.

I also did a force download and re-install of the recordings module. Disabling is a bit harder as it has several other dependent modules.

I also did an fwconsole chown.

I intend to update to the -22 patch, but it seems that 21 and 22 are only updates to asterisk and likely won’t have much effect.

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@Gsmitt wrote:

Hello all,
I’m very new at FreePBX. My Testing Environment as follow;
Using FreePBX 14.0.1.36
My first test is creating an ext. and when there is no answer on that ext. it will redirect to my cell phone, when answering my cell phone, I would like to Press 1 etc… to accept it. What I have done to accomplish this as follow;

1. Created Misc Destination “My-Cell”
2. Created Ext 101
3. Created Ring Group with 101 Extension list. Enabled “Confirm Calls”, Configured “Destination if no Answer “Misc. Destinations=MyCell”.

When I pick up the call from my cell phone it answers right away, I would like it to keep ringing until I press 1 on my Cell Phone. Or is this a Cell phone configuration?
Any help would be appreciated
Thank you in advance

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@readingna wrote:

Good afternoon,

I have just successful installed Asterisk 15.3.0 and FreePBX 14.0.

There would appear to be an error in the .htaccess file located in /admin directory. The wrapper for the the RewriteEngine is missing:

I updated the file to correct the error and now I am getting: “You have 1 tampered files” error.

How do I correct this error.

Thanking all in advance.

Cheers
Nigel

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@ckattimeris wrote:

Hello
we recently update to the following version and we are experiencing issues with HA Module.

Asterisk Version : Asterisk 13.9.1
FreePBX Version : FreePBX 13.0.194.5

seems that out of the blue , pacemaker failovers to the backup node but only few modules not all of them and the asterisk service is not running properly.

we disabled the second node and we are using single node now but still noticed that the asterisk volume is unmounted and we need to restart pacemaker to fix the problem.

anyone noticed similar issue?

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@antonis77 wrote:

Hello,

I would like to know what firewall freepbx is using ?

Does it use the firewalld or the classic iptables command set ?

If i give iptables -L i can see a bunch of rules which are not visible in the GUI

Thanks

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