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Call forwarding with time conditions and different external targets

October 23, 2018, 4:00 am
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@badmin wrote:

Hey guys,

I’m trying to setup an internal emergency service and have trouble wrapping my head around the configuration.

Let’s start with my actual goal: A user should be able to dial 999 from an internal phone (or the whole number with his mobile phone) and the call should be forwarded to a mobile phone from an agent on duty.

Current system:
Current PBX Version: 14.0.3.13
Current System Version: 12.7.5-1807-1.sng7

Things I set up so far:

  • Created a local calendar in my FreePBX with appointments for one agent that is on duty
  • Created a time condition with mode set to calendar and selected the above mentioned calender
  • Set the mobile phone as Misc Destination when a calendar appointment is found, otherwise an announcement is played
  • Created an inbound route for xxxxx999 with the created time condition as destination

With this setup a user can call the xxxxx999 with his mobile and is forwarded to the agent on-duty or receives the announcement based on the time.

I think I need to create additional calendars and time conditions for the other agents. But how do I tell the inbound route to choose the proper destination? Every agent has his personal mobile phone number and different on-duty days.

Additionally how do I set up the forwarding for internal calls? I was not able to define a proper destination on the extension 999.

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XMPP error after monday updates

October 23, 2018, 5:42 am
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@rbailey wrote:

had about 10 or so updates waiting on me Monday, I installed them and noticed today that XMPP wasn’t running.
did the fwconsole stop and start, got a failure. I did a gui uninstall and remove of XMPP and redownloaded and installed with same results. here is the console start error, looks to me that there is a missing module async , any ideas on how to fix.

[root@pbx-knox ~]# fwconsole stop
Running FreePBX shutdown…

Stopping UCP Node Server
Stopped UCP Node Server
Chat Server is not running
Shutting down Asterisk Gracefully. Will forcefully kill after 30 seconds.
Press C to Cancel
Press N to shut down NOW
[============================] 1 sec
[root@pbx-knox ~]# fwconsole start
Running FreePBX startup…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
16981 [============================]
Finished setting permissions
Running Asterisk pre from Firewall module
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Checking Vpn server
Starting Asterisk…
[============================] 1 sec
Asterisk Started
Running Asterisk post from Ucpnode module
Starting UCP Node Server…
[>---------------------------] < 1 sec
Started UCP Node Server. PID is 32313
Running Asterisk post from Xmpp module
Resetting PBX Users Failed: The command “node /var/www/html/admin/modules/xmpp/node/resetpbxusers.js” failed.

Exit Code: 1(General error)

Working directory: /root

Output:

Error Output:

module.js:338
throw err;
^
Error: Cannot find module ‘async’
at Function.Module._resolveFilename (module.js:336:15)
at Function.Module._load (module.js:278:25)
at Module.require (module.js:365:17)
at require (module.js:384:17)
at Object. (/var/www/html/admin/modules/xmpp/node/resetpbxusers.js:3:13)
at Module._compile (module.js:460:26)
at Object.Module._extensions…js (module.js:478:10)
at Module.load (module.js:355:32)
at Function.Module._load (module.js:310:12)
at Function.Module.runMain (module.js:501:10)

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Call Fall Sip Trunk

October 23, 2018, 6:52 am
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@arlit wrote:

hello I have a small problem with my pbx and incoming calls through the trunk. the calls leave normally towards telephone numbers. the problem is when calls call me enter my pbx and in the input path I have the extension 9152, the phone rings then when I answer it hangs alone. I communicate with the provider they tell me to check my pbx because they are all right on their part they say that when that happens my pbx responds with a “bye”. I would appreciate your help since I am new to this.

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Voice Mail Issue

October 23, 2018, 8:31 am
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@xdvst8x wrote:

When a user has email and the auto delete voicemail function enabled and another user forwards a voicemail to that user the voicemail gets emailed but not deleted. is this by design?

FreePBX Version: 14.0.3.26

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Voicemail Transcription with IBM Service and that script

October 23, 2018, 1:31 pm
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@kristiandg wrote:

In reference to this thread:

I was going to try to set this up. I did activate the new voicemail script (as it’s showing in the email the “transcribed text” section, but I’m having trouble getting it to connect to IBM in the first place. It seems they may be using service accounts instead of a login/pass (which L/P is what the script is geared for).

Anyone have familiarity with doing the tie-in with this script to the IBM service? It did create some default service credentials, but again, they’re more token-like, not Login/Pass based. The downloadable JSON file says it has an API Key and Secret (though I can’t find the “secret” within the file, only the API key) - FIgured the Key was the login and Secret was the password… I’m a little stuck…

Here’s an example output from a JSON credential (I’ve already deleted the credential and edited some of what was in the example below so it’s not valid), but if anyone knows what elements I’d extract to place in the Username/Password fields of the script, that would be greatly appreciated:

{
  "apikey": "Q507dlsfDGVFNiRrgTkl7cZR2VGalOCkLQhfVtLQ8H2c",
  "iam_apikey_description": "Auto generated apikey during resource-key operation for Instance - crn:v1:bluemix:public:speech-to-text:us-east:a/4107aa09a77e4266a49128857817c305:f00755dd-a49a-48e6-800c-cd0d1a2c0591::",
  "iam_apikey_name": "auto-generated-apikey-fda2b702-f25d-4c8e-99a7-6feb808edc34",
  "iam_role_crn": "crn:v1:bluemix:public:iam::::serviceRole:Manager",
  "iam_serviceid_crn": "crn:v1:bluemix:public:iam-identity::a/4107aa09a77e4266a49128857817c305::serviceid:ServiceId-dddd91a6-650a-4739-8d17-56685b24763c",
  "url": "https://gateway-wdc.watsonplatform.net/speech-to-text/api"
}

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Zulu 3 - iPhone App

October 23, 2018, 1:45 pm
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@mvogel4949 wrote:

Does the extension I create for the Zulu iphone app have to be a pjsip extension? I created a SIP extension that worked for a desktop app but the mobile app does not seem to connect.

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Help please, BLF showing remote VPN phone as busy

October 19, 2018, 9:12 pm
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@Ducktour wrote:

I’ve searched the community and Wiki pages but can’t so far find the answer.

I am up to date with all software and firmware.

I successfully created a Sangoma S500 VPN extension at my home office. I have BLF set up via EPM and at my store all phones on the LAN side see my home extension as BLF red and busy although pressing the BLF will connect calls to my home phone OK.

On the home office VPN phone all the BLF work fine, I can see when phones at the store are busy, etc.

I’m missing something and any advice is appreciated.

core show hints 3180

shows state as ‘in use’ regardless of phone is powered up or even disabled in VPN Server Clients

image
image.png1366x728 25.8 KB

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Zulu - iPhone not ringing when locked

October 23, 2018, 10:46 pm
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@gwntc wrote:

Zulu beta is configured and works on my iPhone when it is unlocked. If the phone is locked, it never rings. I checked all the notification settings on my phone and it appears to be ok.

When I go to Admin > Zulu and click on the circle next to my user, I get the following error

Undefined index: status
File:/var/www/html/admin/modules/zulu/Zulu.class.php:109

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Registration

October 24, 2018, 6:30 am
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@Chimeara wrote:

I have downloaded and installed FreePBX version 14.0.3.1 and am trying to register but keep getting the following error
"Unable to display activation page. Error returned was:
> Resolving timed out after 3513 milliseconds "
I don’t have any problems with other workstations on the same LAN accessing the internet through the modem.

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You don't have permission to access /admin/ajax.php on this server

October 24, 2018, 11:17 am
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@hardocp wrote:

I am currently running asterisk / freepbx 14.0.4.1 on centos 7 with asterisk 14 as well

This was installed from scratch and is not a distro install

Everything seems to be running ok with the following excpetion

When i first log into the Freepbx server i get the permission error in the topic title

I am able to click on the error and it goes away and then i can access the other parts of Freepbx generally – its really just the dashboard tab which gives me the error

I ran fwconsole chown to try and fix it but that did not seem to work

any ideas or assistance would be greatly appreciated

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Auto cleaning of logs?

October 24, 2018, 3:22 pm
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@kristiandg wrote:

Good evening. I just got off an outage call where a system went down because it’s /dev/mapper/SangomaVG-root partition was at 100% Come to find out, the log files are what burned up all the space.

Is there not something built-in that clears old logs? Obviously I deleted the files and restarted the system and am now back down to 17% used partition space, but I would have thought there was a routine that ran to keep that log folder tidy.

Thanks.

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Can there be too many extensions for Round Robin ring plan

October 24, 2018, 4:15 pm
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@rnrstar wrote:

Is there a best practices as to how many extensions are too many for a round robin ring plan?

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Erroneous Marking Release

October 24, 2018, 12:58 pm
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@chriskinsey wrote:

So, Andrew is correct - the problem is fixed, but you can only get the fix if you ask the system in a very specific manner (while clicking the right mouse button with your left hand and pressing the space bar with your nose while hitting escape with your left ear)
And in more things which make you go hmmmm - Andrew reports that FreePBX 15 is still alpha. Somebody should really tell marketing:

Author: Jim Machi – VP of Marketing, Sangoma. Today, at the AstriCon Users & Developers Conference, Sangoma just announced that Asterisk 16 and FreePBX 15 are now available.
Source: https://www.sangoma.com/asterisk-16-freepbx-15-now-available/

No mention of Alpha or I certainly would not have upgraded. Last mention of Alpha was August. Was perfectly happy on FreePBX 14 and Asterisk 13. Relied on Sangoma press to do upgrade. Stupid user me believing Marketing.

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In past versions was there an option to email recordings that is now missing?

October 25, 2018, 12:11 am
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@Wanderer wrote:

I upgraded from FreePBX 11 to FreePBX 14 several weeks ago and just recently I had a user contact me because he was no longer able to use *1 to record a call. I figured out why that wasn’t working but then he said that the only way he could get the finished recording was to go into the UCP and download it. He said that in the past the system would email the recording to him at the conclusion of the call.

Maybe I am making things up in my mind, but I could have sworn that there used to be a place in the extension settings where you could enter an email address, and if a user created a recording it would send it to that email address. This was back when all the settings for an extension were on a single page rather than broken up into pages with tabs. Am I imagining things, or was that something that was actually possible in older versions? And either way, is there some way to do it now? I’d want to email the recoding to the specific user that initiated it by pressing *1, not to some general email address.

Also, do such recordings get automatically purged after they are downloaded via the UCP, or after a certain length of time? Or if there is a way to email them, are they deleted after they are emailed?

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Error after replacing CMOS battery and memory stick/updates

October 25, 2018, 6:24 am
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@Bill_Clarke_Jr wrote:

I have an error popping up and will not let me apply config. Here is what shows when I try.

exit: 1
[FATAL] SQLSTATE[HY000] [2005] Unknown MySQL server host ‘localhost:3306’ (1) SQLSTATE[HY000] [2005] Unknown MySQL server host ‘localhost:3306’ (1)

Trace Back:

/var/www/html/admin/libraries/BMO/Database.class.php:70 PDO->__construct()
[0]: mysql:host=localhost:3306;dbname=asteriskcdrdb
[1]: freepbxuser
[2]: 1e7121ad5ed1

/var/www/html/admin/modules/cdr/Cdr.class.php:35 Database->__construct()
[0]: mysql:host=localhost:3306;dbname=asteriskcdrdb
[1]: freepbxuser
[2]: 1e7121ad5ed1

/var/www/html/admin/libraries/BMO/Self_Helper.class.php:116 Cdr->__construct()
[0]:

/var/www/html/admin/libraries/BMO/Self_Helper.class.php:36 Self_Helper->autoLoad()
[0]: Cdr

/var/www/html/admin/libraries/BMO/Hooks.class.php:163 Self_Helper->__get()
[0]: Cdr

/var/www/html/admin/libraries/BMO/Hooks.class.php:37 Hooks->preloadBMOModules()

/var/lib/asterisk/bin/retrieve_conf:26 Hooks->updateBMOHooks()

William Clarke Jr
William.ClarkeJr@techsolutions.cc

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Auto provision issues with Yealink W60B via EPM

October 25, 2018, 7:55 am
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@andrwsryan wrote:

Hi All - I am having issues with auto provisioning a Yealink W60B via EPM. I have successfully provisioned 2 - Yealink T46S phones over https with this install, so i know there are no firewall issues. EPM template is set up for both the T46S and the W60B. Using the exact same provisioning URL, username, password, the W60B will not provision. W60B firmware is at the latest release.

FreePBX 14.0.4.1

Are there some logs that i can monitor to see what is going wrong? Any additional info i should provide?

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Very Basic Question - Multiple Handsets using Same Extension (number)

October 25, 2018, 9:23 am
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@billden wrote:

I want my VoIP phone to be able to pick up on a call in progress and simply join in without having to go through a bunch of conferencing steps.

I need to be able to put a single extension “123-4567” on two different phones in the same office. I want to be able to answer an incoming call on one handset and, while the call is still in progress, pick up the phone call on the handset and join the conversation. I want it to work just like (or a as similarly as possible) to a home phone where you can get a call in the kitchen and have someone else pick it up in the bedroom.

I know this may sound very simple but I have been told over and over again in the past that VoIP simply does not do this. Can it be done?

This is for a classroom standalone implementation so anything I am willing to try anything to make it work.

Thanks.

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White dashboard

October 25, 2018, 12:20 pm
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@hjbtyvv wrote:

When I try to go to the dashboard by entering http://192.168.0.196, I get a white screen with this on it:

0 System Admin 14.0.20 Copyright 2018 by Sangoma Technologies Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at https://www.freepbx.org/legal/

Any ideas?

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Router port forwarding necessary

October 25, 2018, 1:27 pm
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@frenaud wrote:

I’m getting conflicting information regarding port forwarding. Our NAT-enabled FreePBX box is sitting behind a Cisco router. Do we need to forward TCP 5060 and UDP 10000-20000 to the PBX? We do not have any external phones, they are all on the same LAN as the PBX.

I’m asking because we cannot receive inbound calls and when we make an outbound call, outgoing audio works, but we do not hear anything.

I thought that since our PBX establishes the outgoing connection to our SIP line provider we didn’t need to forward anything. Maybe I need to add something to our inbound ACL?

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Music on hold using soundcard line input

October 25, 2018, 1:32 pm
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@giggabotch wrote:

Hi Guys. I need help!
I’ve installed Alsa tools and utils with the intent of inputting audio on a soundcard for music on hold. Before we get into a discussion as to the legalities, I should mention that this is for a radio station installation of Freepbx.
I created a custom category for default MOH with line command “arecord -c 1 -f MU_LAW -r 8000 -t raw”
I can verify that sound card and alsamixer interface is working in CentOS. But I hear no audio on hold and I see this constant error in the log files
“[2018-10-25 20:26:36] WARNING[10781] res_musiconhold.c: poll() failed: Interrupted system call”

???

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