@jazzb wrote:

Hello All,

My free PBX setup is sending the last 4 digits to our UK IDSN line. From our desk phones this has always worked. But from the Skype/Lync trunk skype sends the full E164 number so I’ve modified “/etc/asterisk/extensions_override_freepbx.conf” as below to just take the last 4 numbers.

[macro-dialout-trunk-predial-hook]
; get only last 4 for caller ID
exten => s,1,Set(CALLERID(num)=${CALLERID(num):-4})
exten => s,n,MacroExit()

This seems to be working fine, and if I call my mobile I get the correct caller ID show up. However, if I call an international number/mobile I get the main ISDN/Billing number show up – not my personal extension.
Any ideas?

Thanks in Advance.

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@fastdrw wrote:

I first want to thank you Tony and team for creating the migration tool. It will be a godsend for migrating customer sites from Elastix to FreePBX.

I have found a couple of issues with the migration tool.

1. Directory /var/spool/asterisk/monitor? (9.13GB) [yN] y does nothing. Is the a bug or due to the size is y ignored.
2. Every extension migrated the voicemail is set to no. This causes a tremendous amount of work enabling and setting all the passwords for voicemail.
3. No Voicemail greetings or recordings are migrated.
4. All Ring Groups and Queues where the fail over destinations were set to a voicemail (unavailable) error out because no Voicemail boxes are enabled on the destination server.

All of these issues may have something to do with the size on the var/spool/asterisk/monitor directory, but having all of the extensions set to voicemail = off really becomes a pain.

This was my third migration of the same two systems so I could fully understand all of the processes which take place. I have putty text captures of both systems, prior, during and afterwards if you want to see them.

Thanks.

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@Connex wrote:

Hi!

Are UCP instant messaging not supposed to be saved at all as a messaging history? Is there a way to have them saved in the chat history? Seems like a very basic and essential feature for any chat.

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@learjet3204 wrote:

hopefully you can follow this.

we are randomly having a transferred call connect to a call in our queue.

recpt ext. 201 answers call
transfer to another ext.
another call comes in
recpt ext 201 answers and sends that to queue group 799

transferred call into the queue ends up talking to the other transferred call.

here it is in the log.

Wed, May 2 2018 4:05 PM	00:00:05	SHOP PHONE 2 <7122606043> called Rhonda B <201>
Wed, May 2 2018 4:05 PM	00:00:11	Rhonda B <201> answered
Wed, May 2 2018 4:05 PM	00:00:00	Rhonda B <201> transferred [blind] SHOP PHONE 2 <7122606043> to Extension 799
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Tyler D <223>
Wed, May 2 2018 4:05 PM	00:01:52	SHOP PHONE 2 <7122606043> entered Queue (799)
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Marc P <209>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Tristan E <210>
Wed, May 2 2018 4:05 PM	00:00:00	Rhonda B <201> hung up
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Scott F <204>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Eric B <216>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Kyle P <217>
Wed, May 2 2018 4:05 PM	00:00:16	SHOP PHONE 2 <7122606043> called Tyler D <223>
Wed, May 2 2018 4:05 PM	00:00:16	SHOP PHONE 2 <7122606043> called Eric B <216>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Tristan E <210>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Scott F <204>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Kyle P <217>
Wed, May 2 2018 4:05 PM	00:00:15	SHOP PHONE 2 <7122606043> called Marc P <209>
Wed, May 2 2018 4:05 PM	00:00:05	SHOP PHONE 2 <7122606043> called J&W DIESEL AND <6053325111>
Wed, May 2 2018 4:06 PM	00:01:15	J&W DIESEL AND <6053325111> answered
Wed, May 2 2018 4:06 PM	00:00:07	SHOP PHONE 2 <7122606043> called Marc P <209>
Wed, May 2 2018 4:06 PM	00:00:06	SHOP PHONE 2 <7122606043> called Kyle P <217>
Wed, May 2 2018 4:06 PM	00:00:06	SHOP PHONE 2 <7122606043> called Eric B <216>
Wed, May 2 2018 4:06 PM	00:00:06	SHOP PHONE 2 <7122606043> called Tyler D <223>
Wed, May 2 2018 4:06 PM	00:00:06	SHOP PHONE 2 <7122606043> called Scott F <204>
Wed, May 2 2018 4:06 PM	00:00:06	SHOP PHONE 2 <7122606043> called Tristan E <210>
Wed, May 2 2018 4:07 PM	00:00:00	SHOP PHONE 2 <7122606043> hung up
Wed, May 2 2018 4:07 PM	00:00:00	J&W DIESEL AND <6053325111> hung up

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@LesD wrote:

I have started playing with UCP and find that clicking on number links brings up: Select an Action > Originate Call > Initiate

It then shows From: (my extension number) To: (contact name or CID name)

I click on the Originate button below that and nothing happens.

There is a line below the button and in the bottom section far right there is the number ‘200’ which means nothing to me.

I have checked various permissions: ‘All Users’ group is set to Enable Originating Calls and the extension user has been tried with being set to Inherit and also explicitly set to yes.

Any suggestions on what I try next.

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@nmarques wrote:

Do System Recordings, Announcements, or IVRs have a maximum length? I didn’t see any settings, but want to inform users if there is a max length.

Thanks.

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@Autourdupc wrote:

Hi all.

On Freepbx 13.0.195, running on a Raspbx Pi 2 (Raspbian system)

When I want to launch the default backup, I get error messages and backup is not done.
Note that I did not modify any settings, settings are thoses by default.

(Note also that I have a second Freepbx server on another Raspberry runninf a freepbx 14.0.3.1 and it works fine).

Here are the details of the backup process I can see in the pop-up message box while backuping…

Saving Backup 1…done!
Initializing Backup 1
Backup Lock acquired!
Running pre-backup hooks…
Adding items…
Building manifest…
Creating backup…
Storing backup…
Whoops\Exception\ErrorException: unlink(/var/spool/asterisk/backup/Default_backup/20170801-000002-1501538402-13.0.192.15-1240444684.tgz): Aucun fichier ou dossier de ce type in file /var/www/html/admin/modules/backup/functions.inc/class.backup.php on line 806
Stack trace:

1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/backup/functions.inc/class.backup.php:806
2. Whoops\Run->handleError() :0
3. unlink() /var/www/html/admin/modules/backup/functions.inc/class.backup.php:806
4. FreePBX\modules\Backup\Backup->maintenance() /var/www/html/admin/modules/backup/functions.inc/class.backup.php:374
5. FreePBX\modules\Backup\Backup->store_backup() /var/www/html/admin/modules/backup/bin/backup.php:148

Any help ?

Regards,
Laurent.

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@toborgps wrote:

So I have a Raspberry Pi that runs our phone system comprising of about 6 phones. It is constantly giving me an error that it runs out of storage making me re-install the whole thing. Can I clear files so it can go back to functioning correctly. (Ex clearing call logs etc…) I have to reinstall the img. every 2 weeks. Does anyone know of a fix?

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@matthewljensen wrote:

I made a stupid mistake and let our server get full from a packet capture. I got reports of big problems, especially attempting to use the park module, and I immediately realized my mistake. I couldn’t access the gui, and since I had the logs in the temp folder, I just rebooted the server and it seemed to come up OK. I was able to access the gui fine this time, so I looked in system admin-storage. The boot drive has plenty of space, but the other drive is completely full. Things quickly detiorated again and now I can’t access the gui anymore. But running ls -alh in the root folder doesn’t give me any clue where the storage is going. Also, we have raid on this system, 2 100 gb drives. I’m attaching a screenshot I took from the system admin. Also, asterisk is not working properly now.

PBX Storage.JPG1398x456 50.5 KB

Any help would be greatly appreciated.

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@Connex wrote:

I’m pretty sure this has been asked many times before but I just wanted to clarify.

I know both FreePBX and Elastix are feature rich, so my question is not about features, although if there is a significant difference please let me know.

I wanted to ask about their licensing and their limitations. This is important for us while we are still deciding which PBX to choose and stay with for a long term.

For example, apparently Elastix free version can only do 16 simultaneous calls. I know we probably wouldn’t have clients as large but the fact that it has a limitation is already a bummer. Does anyone know what other limitations are there in Elastix? I would ask on their forum but their registration doesn’t even work. The good thing is, there is a lot of functionality that works right out of the box for no extra charge in Elastix where FreePBX charges for a lot of modules.

In comparison, how is FreePBX standing next to Elastix? What about the features and functionality?

Elastix web interface is much more polished I can tell that for sure.

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@digiteltlc wrote:

Distro freepbx 10.13.66-22

After modules upgrade, there are four left :

Bulk Phone restart
Freepbx Framework
Sangoma Property Management
Endpoint manager

They cannot be upgraded because download hangs.

Tried from CLI the cause is :

fwconsole ma upgrade framework

No repos specified, using: [standard,extended,unsupported,commercial] from last GUI settings

Starting framework download…
Processing framework
Verifying local module download…Redownloading
Downloading…
0/10027259 [>---------------------------] 0%The following error(s) occured:

• Error opening http://mirror1.freepbx.org/modules/packages/framework/framework-13.0.195.tgz.gpg for reading

File can be downloaded correctly if that link is copied into browser.
How can i solve this ?

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@comtech wrote:

FreePBX/Asterisk 13

Is there a CLI command or DB entry that I can use to show if a dynamic member is paused or unpaused?

Thanks!

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@Rogue_Bob wrote:

Hi.

Installed FreePBX Distro not so long ago. It’s look pretty functional in IVRs Queues and similar things. So want to test in following conditions. We got OpenScape for most calls as main station. But in some cases as recording calls, voice menus, greetings it looks little clumsy(sry for Unify fans). And FreePBX in our domestic network. Planning to configure so:
PSTN <=E1=> OS4000 <= SIP trunk => Asterisk
Numeric Plan for example:
For OS4000 is 100-469 EXTs
For Asterisk 470-499

If EXT not need any additional features as rec or personal greeting call works like this:
PSTN - OS4000 - Asterisk(Greetings + DISA) - SIP trunk to EXT on OS4000
If EXT is in call center or some other key call function can placed on Asterisk as SIP EXT:
PSTN - OS4000 - Asterisk(different Greeting/IVRs + Voice rec) - SIP EXT on Asterisk.

So anybody have any idea which type of trunk be more useful? Atm have some issue with OS4000 side. Got STMI2 Q2316-X with nonfunctional old configuration. Guides starts mostly from default configuration. And can’t find any way to reset this unit. If some only knows “how to” will be great to got this info.

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@wassy83 wrote:

Hi to all,
I want to know if is possible a workaround to this situation:

let’s say I have 4 extensions 101 102 103 104
ring group 1000(101,102,103,104)
queue 2000(101,102,103,104)
call waiting is disabled on all extensions.

CASE 1:
an external call is incoming
ring group 1000 is ringing
101 pick up the call
everything is ok

CASE 2:
an external call is incoming
ring group 1000 is ringing
no one can pickup in time the phone before a second external call is incoming
on the other side the first incoming call will hear the free ring tone but the second incoming phone will listen the queue message.

this is unwanted situation for me, cause the client have to listen the entire queue message even if the extensions are free but they were simply ringing.
I tried “Skip busy Agents to no” but this is not working.
any workaround to this?
many thanks

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@Hawkeye wrote:

Call comes into PBX, call is sent to an extension which has FOLLOW-ME enabled, with Initial Ring Time = 0
Ring Time = 20
and then the follow-me list with an extension and a telephone number#

When the call reaches the follow-me remote phone number, the cell phone displays either Anonymous + phone number or Private Unknown

There is no forced callerID on outbound route or trunk.

The call flow is as follows:
Origination => FreePBX => 2nd FreePBX => Carrier for remote termination

The partial paste of a call from the viewpoint of the 2nd freepbx is below.

<— SIP read from UDP:xxx.193.49.61:5060 —>
ACK sip:1XXXX959991@xxx.193.49.40:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.193.49.61:5060;branch=z9hG4bK18b94df7
Max-Forwards: 70
From: “REDACTED” sip:XXXX775411@xxx.193.49.61;tag=as531c3935
To: sip:1XXXX959991@xxx.193.49.40;tag=as54e6af69
Contact: sip:XXXX775411@xxx.193.49.61:5060
Call-ID: 7c4cbf00377142c80ea8893b0020dcde@xxx.193.49.61:5060
CSeq: 102 ACK
User-Agent: FPBX-14.0.3.1(13.19.1)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Channel SIP/7530791895-0000af96 joined ‘simple_bridge’ basic-bridge <70b480b3-8248-4468-bf87-6e3c52325e96>
> 0x7fbf5809d230 – Probation passed - setting RTP source address to xxx.193.49.61:11230
> 0x7fbf3c022eb0 – Probation passed - setting RTP source address to xxx.193.49.45:14820

<— SIP read from UDP:xxx.193.49.45:5060 —>
INVITE sip:1XXX4022553@xxx.193.49.40 SIP/2.0
Via: SIP/2.0/UDP xxx.193.49.45:5060;branch=z9hG4bK56a8f0d8
Max-Forwards: 70
From: sip:XXXX775411@xxx.193.49.45;tag=as2d073b01
To: sip:1XXX4022553@xxx.193.49.40
Contact: sip:XXXX775411@xxx.193.49.45:5060
Call-ID: 7320c23a0c28a67521f8922d0f4e7af2@xxx.193.49.45:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.195(13.17.0)
Date: Fri, 04 May 2018 13:54:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 254

The CNAM is not present on the outbound leg of the call to the follow-me number.

The trunk used on our PBX going to the 2nd FreePBX is below:
host=XXX.193.49.40
type=peer
context=from-trunk
dtmfmode=auto
trustrpid=yes
sendrpid=yes
disallow=all
allow=ulaw
directmedia=no
nat=no
qualify=yes

Hide CallerID in trunk setting is set to NO
The Outbound CallerID in trunk field setting is blank.

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@alexramsey92 wrote:

Reference: Server Config Action -> User Config Action

Inbound Route -> Ring Group (2 extensions) -> User Call forwards at Extension 101
Call Forwarding does not seem to work. Using Grandstream phones with FreePBX hosted in ESXI.

I change the config to:
Inbound Route -> Extension 101
Call Forwarding configured by user works!

I want to be able to ring multiple Extensions at the same time (just 2 phones in a house) but also allow call forwarding.

Seems like a simple thing to do but is my hunch correct that Ring Groups aren’t compatible with user defined call forwarding?

Another consideration is a Time Condition. I’ve just re enabled a Time Condition which will forward to the Extension 101 when in business hours. Outside of Business hours it will ring Extension 999 (Voicemail box for main business).

I suppose I could also create a Misc Dest. with the Cell Phone and then have the Ring Group include a Misc Destination (I think that is possible?). This way during business hours, the cell phone would ring always too?

What if I set it up so that the office phone if not answered, would then auto forward to cell? Is that possible? I think that would be an easy solution.

Is there good reading material regarding what Call Flow control options exist and how they logically map together? I feel confused by this aspect of PBX. The documentation has for the most part provided a good foundation of information for localized topics, but I feel like I need a high level call flow doc, if exists.

Everything else is working well, using FlowRoute. Have the server running inside ESXI as a VM. Battery backup on the machine. Router based firewall and responsive firewall. Seems like a great way to save $$$.

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@nmartine wrote:

We are having problems with calls received from traditional POTC (regular landline) phones calling our FreePBX 13. Most of these regular POTC phones are connected sharing a line with Fax machines.

We are recording the calls in FreePBX and we noticed that every time a call is dropped, we hear a 3 beep sound and the call is immediately cancelled. We have been searching for answers and apparently those 3 beeps mean CALL CANCEL.

Any idea if there is a way to avoid the call to be dropped from FreePBX side?

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@unison wrote:

Hi

I’m looking to implement any new providers using PJSIP.

I have several trunks from a single provider. In the case at hand, I’m trying to get three trunks working - but this equally applies if you are trying to get two trunks going to the same provider.

Using chan_sip this works with the following configuration:

Outgoing

Trunk Name: 11111111
Peer Details:
username=11111111
type=peer
sendrpid=pai
secret=********
qualify=yes
port=5060
host=123.123.123.123
fromuser=11111111
dtmfmode=inband
disallow=all
context=from-trunk
allow=alaw&ulaw

Inbound
Register String: 11111111:********@123.123.123.123:5060/11111111

The seconds trunk is configured as:

Outgoing

Trunk Name: 11111112
Peer Details:
username=11111112
type=peer
sendrpid=pai
secret=********
qualify=yes
port=5061
host=123.123.123.123
fromuser=11111112
dtmfmode=inband
disallow=all
context=from-trunk
allow=alaw&ulaw

Inbound
Register String: 11111112:********@123.123.123.123:5061/11111112

In summary, they are:

• all registered to the same IP Address
• Each registration is to a unique port

This works well, with no issues…

However when migrating to PJSIP, I have tried various approaches; but all inbound calls appear to hit the last registered number…

Within each of the PJSIP trunks I have tried to set the Server and Client URL to be:

• sip:11111111@123.123.123.123:5060
• sip:11111112@123.123.123.123:5061

and have set the AOR Contact to be:

• sip:123.123.123.123:5060
• sip:123.123.123.123:5061

But to date, all changes have had the same result - the inbound call appears on the last trunk to register against our provider.

Ive also tried changing the endpoint identifier order to:

endpoint_identifier_order = auth_username,username,ip,anonymous

However, same result.

Any ideas?

Cheers

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@tonjaw wrote:

Hi,

I created a small script to change the values of CallerID and Context in sip_additional.conf file. My script will be triggered by third party’s application (Hotel Management System) in order to run. Here is the sample:

The original values are:
[81104]
Context= from-trunk
CallerID= Room 1104 <81104>
My script will change it to:
[81104]
Context= from-internal
CallerID= John Six <81104>

But I cannot apply the changes by executing any of the following command:
asterisk -rx ‘sip reload’
or
rasterisk -x reload
or
amportal admin reload

When I check it in WebGUI, it still shows the original values of Context and CallerID. Even when I restart freePBX, the original values remain intact and the changes I made in sip_additional.conf will be overwrited by the original values.

What’s the command to commit and reload the sip_additional.conf? Or is there any way to change Context and CallerID instead of editing sip_additional.conf file which can be triggered by third party’s application?

I use freePBX 2.11.0, asterisk 11.20.0, elastix 4.0.0

Thx

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@degenerationx wrote:

Hi,

I need to create a specific dialplan which gets connected as follow:

Step A: Extension 100 dial an outbound number (e.g. 202-123-4567) using Google Voice trunk.
Step B: First, call gets connected to some 3rd party Number (e.g. 647-XXX-XXXX).
Step C: After call is connected, Wait for 2 seconds, then Press 9, now wait for 2 seconds, and then dial 202-123-4567 followed by ‘#’ sign.
Step D: Extension 100 finally gets connected to 202-123-4567

In simple explanation:
Ext. 100 -> Google Voice Trunk dial same number 647-XXX-XXXX -> Press 9, wait for 2 seconds pause, then dial Ext.100’s outbound number 202-123-4567 -> Ext100 connected to 202-123-4567

I tried looking into creating Outbound Route or Miscellaneous Destinations, but I have been unsuccessful in doing so. Would greatly appreciate specialist’s input.

Thanks in Advance!

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