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Looking for Consultant in Las Vegas Area

November 30, 2018, 11:21 am
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@rnrstar wrote:

We are running FreePBX with the HA module with two nodes using the UC400. We previously were running on an Dell Poweredge 720R. We are seeing the same problem across both hardware platforms. Also on the server is FOP2. The symptoms we are seeing are:

  1. Active calls will shoot up to over 100, often to 200 or more even though there are not that many active calls.
  2. FOP2 will become unresponsive in that calls will appear stuck in the queue even though they ended, Users are unable to set their presence. Users will show on a call when they are not.
  3. Calling into a queue and all the caller will hear is silence instead of the on hold music. Caller will sit there until a reboot or until we do an fwconsole restart.
  4. Usually the system will stop accepting external calls.
  5. Outbound calls result in silence. Meaning that we get dial tone, are able to dial a number but upon sending, we only get silence.
  6. This will happen several times in one day and then go two weeks without issue.

We are looking for a consultant that can come in and take a look at the system and let us know what we need to do to further troubleshoot this and get it resolved once and for all. Any suggestions on consultants in the Las Vegas, NV area or someone remote that can help. All the suggested vendors on voip.com for the Las Vegas area are no longer in business.

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Dahdi Config with FreePBX

December 1, 2018, 3:35 am
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@abbassalameh wrote:

Hello, I have a dahdi module with 3 FXO analog ports connected to a PSTN central Panasonic, I want config dahdi to enable users on PSTN to make a VoIP call when he calls the extension on PSTN,
Like That:
example: 300 + VoIP account
300 is the extension of PSTN connected to a dahdi port
In another way, I need to config dahdi to make PSTN accessible for VoIP’s account, that’s mean VoIP account can call another VoIP account who is redirected directly to the dahdi port then to PSTN extension.
Any help please?

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External calls to Auto Answer Intercom

December 1, 2018, 4:59 am
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@rob521 wrote:

Appreciate if anyone can point me in the right direction. Running Asterisk 14.

I am not sure if what I am asking is possible - I have been getting nowhere for days.

Simply, I want a DID to route directly to an extension with intercom enabled. This is simple enough.

Now, how do I get the extension to Auto Answer an external call? This is critical in this situation.
Internal calls are Auto-Answering fine, but this is not what is needed.

Is it configuration?, or can I simply change the DID to be recognised as an internal call somehow?
Or dial a feature code?

I am a moderately experienced user - not advanced. So if you are able to assist, pls keep the explanation relatively simple if possible. Thanks!

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Smartphones and FreePBX

December 3, 2018, 1:28 am
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@Tommaso wrote:

Hello everyone! I’m really inexperienced with everything related to FreePBX, but at the moment i managed to totally replace my old 3CX system, but now i have a big question… How do i setup connections between a softphone installed on a smartphone and my FreePBX?
I watched CrossTalkSolution’s video and googled a lot and I understood that you have to open a port on your router and this represent a big security issue so you have to configure a VP_N connection in order to avoid this problem!

There is someone there that can give me some advices/tips to show me the way to setup a connection between a smartphone and my FreePBX in order to use remote extensions?

Regards!

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Enable an alert when an extension becomes free

December 3, 2018, 5:50 am
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@Tommaso wrote:

Hello everyone!
the title says it all ^^
On my older PBX system (3CX) i was able to make an internal call and if it was busy i was able to choice to active this funzion or not.
Essentially i just want to be able to call someone in the office, if she/he is actually busy i press a button on my phone and then, ASAP the person is free, my phone will ring so i know i can call them back… is this possible?

i would assume that this is a configuration related to the phones and not to the FreePBX, but i really cant figure wich one it is!

Phone that i’m using are mainly Yealink TP 21 E2 (plus some snom300).

Thanks for your time and help,
Regards!

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Queue linear Ring Strategy + Dynamic Agents

December 3, 2018, 6:00 am
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@PitzKey wrote:

Hello community,

We are trying to setup a Queue that has 4 dynamic agents, 101, 102, 103 & 104.
We want the queue to start at 101 for 10 seconds then 102 10 seconds etc. it should ring at each extension for 10 seconds in an order.

So I changed the queue ring strategy to linear, set the agent timeout for 10 seconds, the queue called them in the order they were configured in the dynamic agents section, except the agent at the bottom of the list. It never called that extension the last, sometimes it called it the first and sometimes the third. first I thought it’s something with 104 so I placed 102 at the bottom but it had the same behavior as 104. No matter in what order they logged in, the bottom extension always had problems.

I read the Wiki, and saw that “linear : Rings agents in the order specified (for dynamic agents in the order they logged in).” meaning, that with dynamic agents the queue will preserve the order they logged in (last logged in will be called last) Then why is it calling the agents in the order they are configured in the dynamic members section?
So I tried setting up the queue as Static agents, but it still had issues with the order…

Finally, I created a new Queue, and static agents are now being called in the order they are configured in the queue, and dynamic agents are called in the order they logged in.

Bravo! It works as documented…

But, The reason I tested linear strategy is because we do want the queue to maintain the order configured for dynamic agents. Is there any way to accomplish that?

Thank you

P.S. I also tested rrordered, but with dynamic agents it also calls them in the order they logged in. :disappointed:

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SIP phones not NATted / call terminates after 30s / internal NAT phone network problem

December 3, 2018, 7:13 am
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@tpreissler wrote:

Hi,

I have received my Sangoma deskphone today and setting up was a breeze … but unfortunately calls get terminated after 30s. Before you say “Fix your NAT”, please let me explain.

My Freepbx 14.0.5.2 is sitting on a DMZ, as I wanted to use it remotely, too.
Now I managed to connect various softphones and even analoge phones and they are work ok. All that is on a different network, and to get to the PBX itr goes through a firewall (OPNSense) and FreePBX that goes out
via a SIP (or landline, either way works).
Now I setup a separate network just for deskphones, connected to a second interface on FreePBX, 192.168.x.y. All connected up and “working”, my Sangoma gets provisioned, NTP time, firmware, what not.

Now when I make a call outbound via SIP trunk on the Sangoma it gets disconnected after 30s. When I call the same number from a softphone or the analog phone, it’s fine.

When I look into the network traffic I can see that the firewall(!) is actually seeing the IP from the Sangoma phone, 192.168.1.133, and it is trying to connect to my public IP:5060. This won’t work and that’s why the call get disconnected.

NAT -outbound to the internet- works. I do not have a problem with softphones on the other network or also via analog (->FreePBX->SIP trunk out). When I make a call from a softphone they talk directly to the FreePBX,
and there are no packets unanswered going out to my public IP - the call is fine.

In a nutshell, how do I NAT my phones on my phone network? I have “Local Networks” 192.168.1.0 listed, but the firewall can still see directly phone traffic.

What’s the proper way here?
Does the Sangoma need to be NATted behind the FreePBX or do I need to have a route back to FreePBX for the phone network - ?

Regards

Tom

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How do I mount a cifs NAS share in freePBX

December 4, 2018, 2:39 am
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@subs wrote:

Since I updated my freePBX to version 14.0.5.2 from version13 a short while ago, it seems my cifs NAS share doesn’t mount so my scripts to pass recordings to a long term archive no longer work - or at least they no longer move everything over, but rather just copy it to the mount point where the shared drive should be.

It has taken me a while to realise quit what was happening, but I am not sure how to resolve this.

This is the (generalised) mount line in the /etc/fstab file
//NAS/path/ /mnt/point cifs credentials=/root/credentials.txt,uid=500,rwx,suid 0 0

Obviously there is a matching credentials file

Anyone able to give me an idea of what I might be missing?

TIA

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FreePBX with multiple Companies

December 4, 2018, 4:27 am
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@mst wrote:

Experts,

I wonder if there is any way to use mutiple companies in one PBX. For example having compony A with IVRs inbound rule would be easy, but what about the outbound rule?

I know lets say 9 digit for comp A dialing out 999-xxx-xxxx compB 8 for dialing out 888-xxx-xxxx and C 7 777-xxx-xxxx

What about if all wants to use same dial plan for all phone? Do I need special modules for that or can do with existing in FreePBX 14?

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Notification that called party is on the phone

December 4, 2018, 7:00 am
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@BryanRagon wrote:

For internal calls, is there any way for the calling party to be notified that the called party is on the phone? I know that the calling phone could have the called phone as a BLF key and then the user would know before placing the call. But we have more extensions than we can add as BLF keys. It would be great that if the called party was busy, their phone would ring, but somewhere on the calling phone’s display it would say “User on the phone, ringing” or something like that.

We have a user who insists our old Inter-Tel system did something like that.

We currently use Yealink SIP-T48S phones.

Thank you!!

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Error never seen

December 4, 2018, 9:28 am
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@Cricchetto wrote:

Entering the ip of the pbx in the browser gives me this error. This morning I loaded audio files for entertainment in the conference. Then the error came out

Whoops \ Exception \ ErrorException (E_WARNING)

file_put_contents(): Only 0 of 189 bytes written, possibly out of free disk space

The pbx is loaded into a kvm on SeFlow and can be accessed by VNC or SSH. What files should I delete and how do I find them to solve the problem?

Thank you all

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Am I Being Hacked? Peer

December 4, 2018, 12:43 pm
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@1sae wrote:

image
image.png854x166 5 KB
I’m seeing a lot if channels like this pop up in my server. I don’t know the IP and I’ve been blocking them one by one in the firewall. In SIP General settings I have “allow anonymous Inbound SIP calls” set to no. In Chan SIP Settings I have “Allow SIP Guests” also set to no. Am I missing something?

I’m running FreePBX firmware 10.13.66-22, asterisk ver 13.19.1

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Saving phone call recordings to second drive

December 4, 2018, 2:26 pm
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@LigerXT5 wrote:

As simple as I presummed it should be, I have been having some odd trouble on a recent freepbx install. The client had requested all their departments calls, in and out, to be recorded. 20 phones in total. The second drive that I set PBX via advanced settings to is a 1TB drive. Formatted and empty. I can create files to it, granted as root, however in the asterisk logs, it says it is failing to save the files, as the location does not exist.

For testing purposes only, thinking maybe it was a permissions issue, I chmod the drive as 777, however no change on recordings to be saved.

The location the recordings should save to, /mnt/recordings/

Error on each call:

[2018-12-04 22:23:24] VERBOSE[2762][C-000001cc] app_mixmonitor.c: Begin MixMonitor Recording SIP/GXWT10-In-0000089f
[2018-12-04 22:23:24] WARNING[2762][C-000001cc] file.c: Unable to open file /mnt/recordings/2018/12/04/in-1231231234-1231231234-20181204-222324-1543962204.2207.wav: No such file or directory
[2018-12-04 22:23:24] ERROR[2762][C-000001cc] app_mixmonitor.c: Cannot open /mnt/recordings/2018/12/04/in-1231231234-1231231234-20181204-222324-1543962204.2207.wav

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Removing Photo Thumbnail from YeaLink t46s

December 4, 2018, 3:20 pm
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@SperoHealth wrote:

Hello,

I am trying to figure out how to remove the photo thumbnail on my YeaLink phone during call status (outgoing and incoming calls) which causes numbers to scroll and it just takes up a lot of room on the phone. Any insight would be helpful. Here is an image of what I mean

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Adjust moH volume

December 4, 2018, 4:04 pm
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Yum Broken After 13->14 Update

December 4, 2018, 7:01 pm
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@1sae wrote:

I just ran the FreePBX 13->14 update and it mostly seems to have gone ok except for one thing. Yum Update seems broken.

# yum update
error: Failed to initialize NSS library
There was a problem importing one of the Python modules
required to run yum. The error leading to this problem was:

   cannot import name ts

Please install a package which provides this module, or
verify that the module is installed correctly.

It's possible that the above module doesn't match the
current version of Python, which is:
2.7.5 (default, Aug  4 2017, 00:39:18)
[GCC 4.8.5 20150623 (Red Hat 4.8.5-16)]

If you cannot solve this problem yourself, please go to
the yum faq at:
  http://yum.baseurl.org/wiki/Faq

No matter what I search I can’t find a solution other than one possible option, but I can’t test it out cause it’s “Subscriber Exclusive Content” on this page: https://access.redhat.com/solutions/3134931

Any help on how to fix this?

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No incoming call's after module update

December 5, 2018, 3:45 am
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@MythOfTheLight wrote:

No incoming calls after updating modules. Outgoing work.
Current PBX Version: 14.0.5.2
Current System Version: 12.7.4-1803-1.sng7

Installing the latest version of FreePBXDistro - no incoming with identical config. Installing the old version 10.13.66 without updates - it works. After updates - no incoming calls.

List of modules:

Module Name Current Version New Version
Preserve Accountcode 13.0.2 13.0.2.2
Announcements 13.0.7.1 13.0.7.3
Backup & Restore 14.0.3.15 14.0.10.1
Blacklist 13.0.14.8 14.0.1
Bulk Handler 13.0.14.4 13.0.14.7
Calendar 14.0.2.2 14.0.2.6
Callback 13.0.5.2 13.0.5.3
Call Forward 14.0.1.2 14.0.1.3
Call Recording 13.0.11.5 14.0.5
CDR Reports 14.0.5.10 14.0.5.14
Call Event Logging 14.0.2.3 14.0.2.8
Certificate Manager 13.0.37 14.0.3.1
CallerID Lookup 14.0.1.5 14.0.1.7
Conferences 13.0.23.9 13.0.23.12
Contact Manager 14.0.3.4 14.0.4.10
Core 14.0.5.11 14.0.18.37
DAHDi Config 14.0.1.1 14.0.1.3
System Dashboard 14.0.3.3 14.0.4.1
Digium Phones Config 13.0.7.3 13.0.7.4
DISA 13.0.6.1 13.0.6.6
Fax Configuration 14.0.2.2 14.0.2.6
Follow Me 14.0.1.16 14.0.1.20
System Firewall 13.0.49.2 13.0.57.1
Asterisk IAX Settings 14.0.1.3 14.0.1.4
Info Services 13.0.1.2 13.0.1.3
Online Support 2.11.0.7 13.0.1
IVR 13.0.27.6 14.0.4
Languages 13.0.6 14.0.1.2
Asterisk Logfiles 13.0.10.4 13.0.10.5
Music on Hold 13.0.22.3 13.0.22.4
Paging and Intercom 13.0.26.3 14.0.6
Parking Lot 13.0.19.7 13.0.19.8
PIN Sets 13.0.8 13.0.10
Process Management 13.0.4.2 13.0.5.1
Presence State 14.0.1.5 14.0.1.7
Queues 14.0.2.11 14.0.2.22
Ring Groups 14.0.1.4 14.0.1.5
Asterisk SIP Settings 14.0.26.7 14.0.27.5
SMS 14.0.4.3 14.0.4.6
Sound Languages 14.0.4.2 14.0.5
CID Superfecta 14.0.4 14.0.7
System Admin 14.0.11.2 14.0.22
Time Conditions 14.0.2.12 14.0.2.15
User Control Panel 14.0.2.1 14.0.3.1
User Management 14.0.3.37 14.0.3.44
Voicemail 14.0.1.17 14.0.4.1

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Search on UCP

December 5, 2018, 5:30 am
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@venizia03 wrote:

Hello!

It look like the search functionnality on UCP for call history is only based on the phone number. Is this correct?
I would like to do search on date field. Is there a way to do it?

Thx in advance!

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HA for FreePBX recording files

December 5, 2018, 7:02 am
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@psdk wrote:

Hi to all,

We’re working on HA solution for 2 FreePBX instances. we run in with pacemaker and everything is ok. problem is about recording files. we are going to use DRBD but as we can see, there are some limitations on it. something like brain split issue.

I want to know if someone has any other solution? if yes, what?
and if DRBD is the best, how you cope with its limitations?

Thanks.

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Re-route calls based on sip peer availability

December 5, 2018, 9:50 am
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@ashcortech wrote:

Trying to automate a fix for an issue I have. I’ve moved my freepbx distro to a cloud based server. When I used to have it on local hardware and my internet would go down, my sip provider would detect the PBX was offline and automatically failover calls to my cell.

Now that the pbx is in the cloud it never goes offline so if my local internet is down calls will just go to voicemail instead of being re-routed to the failover number (as this is a function of the sip provider).

I’m wondering if there is a way (a script perhaps) that I could detect if all the sip peers were offline and forward all inbound calls to an external number?

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