@ruben_seguridad wrote:

Hello,
I want to reset ZEND ID for FreePBX High Availability module.
In this URL explain how do this https://wiki.freepbx.org/display/FPG/How+to+Move+a+Deployment+ID+to+a+new+PBX
But in my acount not appear the button for reset ZEND ID.
Can you help me??

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@crimsonbadger wrote:

I’ve written this to hopefully save some time for you hard working, under-rated network techs, so read on :-

Symptoms – My New FreePBX 14.0 distro test installation – the internal phones suddenly are unable to make calls.

The Free PBX web interface showed some Daemons not starting and Asterisk module as errored, ie some critical errors

Logged into the console as root. Immediately a host of errors were shown, a quick scan through them came up with several that indicated a lack of disk space.

A quick df command showed that the mount point /dev/mapper/SangomaVG-root, on the root (/) was at 100%.

Back to the Web interface – Admins – System Admin – storage again showed root ‘/’ drive usage at 100%

After a lot of googling and some very helpful sites – this is how I created more space.

1) Increase size of the Virtual Hard Drive

Our FreePBX is a virtual machine on MS Hyper-V, so shutdown the machine and increased the size of the virtual hard drive for the virtual machine – added another 40 GB (Easy enough !)

2) Increase size of the partition

Using the ubiquitous and fantastic GParted partitioning tool, downloaded the GParted ISO, mounted this as VM media and booted into GParted (With a physical machine, create a bootable USB and boot into GParted).

Now it’s easy enough to select the partition with the SangomaVG mount point and drag the handle to extend the partition to fill the new space in the virtual drive.

Sooo after a reboot, hooray, I thought, job done !

BUT in the console, df to view the Filesystem – shows the Sangoma mount point as still at 100% and still at the old 50G size (should be 100G !). Now need to extend the File System to fill the partition.

3) Resizing the File System

Attempt 1 : Tried the usual way for a linux file system, resize2fs /dev/mapper/SangomaVG-root ( which is the file system name as shown by the df command), to expand the file system to fill the partition.

This FAILED with a message ‘Bad magic number in super-block

Attempt 2: command df –T shows that the file system is of type xfs

Instead of resize2fs, this would require the xfs_growfs command:
xfs_growfs /dev/mapper/SangomaVG-root -d to expand to fill the partition.

Although the command looked to run OK, this FAILED with message ‘data size unchanged’

Attempt 3: Using the linux partition editor - parted , then print commands, the partition shows flagged as lvm (logical volume, a bit like dynamic partitions, which apparently cannot be grown with xfs_growfs !

So more research:

Command vgs – shows the free space available for your file system, I had 42.6G spare (which would fill the partition)

Can now extend the xfs file system on the LVM flagged partition, with:

Lvextend –L +42.6G /dev/mapper/SangomaVG-root –r

df now shows spare space on the file system, usage at 54%

SUCCESS

Rebooted FreePBX and hey presto, daemons started and no Asterisk error

PHONES WORKING AGAIN !

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@volkswagner wrote:

Greetings,

I’m not able to get my NAT settings and firewall setup correctly.
I’m running FPBX-14.0.5.25(13.22.0), Distro behind MikroTik router.
I have Ports 10000-2000 forwarded to server (192.168.5.7) allowed from anywhere
and port 5060 forwarded only from two external IP address for two remote phones.
This setup works as follows:
Outside Extensions can make calls using trunk lines without issue.
Calls from External Extension to internal extension get’s one way audio (internal can hear)
Internal calls using trunks can hear outside party, but party can’t hear.
Internal calls to internal extensions (all behind same router in same subnet) no audio either direction.

I’ve tried changing NAT setting In “Advanced settings” (no,yes, routed) none of which seem to impact.

I’m not understanding how to interpret call trace so I’m asking for help.
The following is from an internal extension to extension (server and both devices on same subnet
and behind same router). What stands out to me is " Contact: <sip:47.20.xx.xx:5060>" which
is the public ip of the network, same as “External Address” in under .

I have also tried entering local network as 192.168.5.0/24
and 192.168.5.0/255.255.255.0

Previous version of FreePBX had NAT settings in the extension
setup. I don’t see a GUI option for setting NAT on an extension basis.

Any and all help is greatly appreciated.

https://pastebin.com/embed_iframe/5VwgErA6

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@mst wrote:

Experts, I have clarityel provider and looks like they block option to allow ANY CID on the trunk. Currently I have Force TRUNK CID. The problem is I would like to use other number than in Trunk for outgoing faxes.

Inbound is forwarded to extension (which is set on grandstream H812) but outgoing I use with prefix 9 and specific CID. When I choose Allow ANY CIDin the trunk I have I

chan_sip.c: Received response: “Forbidden” from ‘sip:FAXNUMBER@sip.claritytel.com:5160;tag=as5fa7c’
[2019-03-06 01:34:55] VERBOSE[28402][C-00001229] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)

In the extension I would use CIDnumber but I cannot. Is there any way I can fix it in freepbx config or calling claritytel is the only way?

thank you

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@philusa wrote:

I had to change my email password for email notifications. After doing that and saving, the email user name shows “admin”. That’s not the correct address and when I try to change it, it goes back to admin.
I vaguely remember when the system was setup that there was a problem with this and it was changed via command line. (not by me)
Help is appreciated.

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@sorvani wrote:

Thanks to @xrobau for this post: Polycom phones via http and HTTPS

I was able to use the same logic to backup the Yealink local contacts to FreePBX’s /tftpboot folder.

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@Whacka wrote:

I’m just an enthusiast that has a few cisco phones hooked up to a cisco switch and a dell server running Free PBX. I know google voice done Fu*ked up their protocol so it wont work by it’s self with FreePBX anymore. So with what’s going on now, how could I use google voice as a trunk with free PBX cheap. Or if there’s a better solution for a better price. Please let me know how to achieve that. I want to be able to receive outside calls and make out going calls.

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@asifm wrote:

Hi Guys,

I’ve got FreePBX Server up and running and it’s all fine. Everything with softphones on desktop work fine with no issues.
But when it comes to some Polycom phones that we are trying after a while I get UNREACHABLE error.

I’ve tried disabling NAT (We have a 1 to 1 NATting for our server), different options, increasing the timeout value but nothing has helped.

Has anyone else come across this issue or know what I can do to fix this?

Thanks

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@marmuSIP wrote:

Dear fellow freePBX users,

I just want to change the incoming CID to “0” if CID name or num = “unknown”. This should be quite simple, but I’m stuck figuring out how and where to inject my three lines.

For now I’m aware of these two possibilities

1. recommended: add custom context and set this in trunks settings
2. use auto generated custom context which is automatically included (from-trunk-sip-voipGwName-custom)

Going with the first solution I created a custom context “from-trunk-unknown” in extensions_custom.conf:
[from-trunk-unknown]
exten => _.,1,ExecIf($[ “${CALLERID(name):0:7}” = “unknown”]?Set(CALLERID(name)=0))
exten => _.,n,ExecIf($[ “${CALLERID(num):0:7}” = “unknown”]?Set(CALLERID(num)=0))
exten => _.,n,Goto(from-trunk,${EXTEN},1)

I set this in my trunk incoming settings:
USER Context: inbound
USER Detail
secret=*****************
type=peer
canreinvite=yes
context=from-trunk-unknown
dtmfmode=rfc2833

When I Apply changes the context “from-trunk-unknown” is loaded (fwconsole and full log show that), but it is never used. Inbound calls start in context “from-trunk-sip-voipGwName” and then continue in “from-trunk”.

Any help on how to get this working is highly appreciated.

Also I wonder if the User Context “inbound”, which seems not to be required in our setup, is causing
trouble.

Cheers,
Marcus

FreePBX 13.0.190.7

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@Bob327ss wrote:

I bought and installed Call Recordings Reports a couple months ago, and it seems to work fine, other than I can’t download the archive files to store them. They show up in the Recording Archives tab, but when I click them (in multiple browsers) I get an error:

" This site can’t be reached

The webpage at http://10.10.10.150/admin/ajax.php?module=recording_report&view=fdownload&command=download&filename=backup2.tar might be temporarily down or it may have moved permanently to a new web address.

ERR_INVALID_RESPONSE"

This is with FreePBX 13.0.195.26 if that matters.

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@okynnor1 wrote:

Hi,
I read the update to FastAGI in the announcement by Sangoma. I went to Settings --> Advanced Settings, yet I couldn’t find in order to enable it. Can you confirm the location in FreePBX 14?

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@dgrigg wrote:

I have an issue with FMFM and am hoping somebody may have seen it before and have a solution.

I am using FMFM to call an external number, a cell phone. For the most part it is working fine. However I occasionally get a recording of the “Too late” announcement that’s left in the voicemail for the follow me number. I’ve tracked it to cases where either the cell phone blocks the call or where the user declines the call. How can we avoid this?

Cell phone is setup to block call from numbers that are not in the contact list. So under normal circumstances, a call to the cell number from a number not in contacts never rings the cell phone and the caller ends up in cell phone vm box. A call from a contact rings the cell phone normally.

FMFM seems to be working as expected when caller is in contacts. Call forwards off to cell, user can pick up if desired, if not answered it goes to PBX vm. However if the call is actively declined at cell phone - user presses the decline button - it behaves the same as if the call had been blocked.

Either way, the caller is handled professionally and ends up in the proper pbx vm. It’s just that this artifact message of the too late announcement recording is left in the cell phone vm.

So it seems freepbs is seeing the cell phone vm picking up as a call completion but it is also sending the call back to the pbx vm. It’s not releasing the call to the cell phone vm even though that is first to answer, but it’s not ignoring it either.

Here are the details:

• FMFM using ringall.
• One number in the list, the cell phone. Might add additional in future.
• confirm is on, for ability to determine when the call is originating from freepbx as opposed to someone dialing the cell number directly. Open to other methods to do this if needed.
• all trunks are sip.

Tell me what else you need to know, and thanks in advance for any assistance you can offer!

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@rajeevkkrishhna wrote:

hello,

High queue of task processors

subm:cel_aggregation_topic
subm:manager_topic-

task processor queue size keeps on growing to more than 90,00,000 in five mins.
RAM usage goes upto 32 GB and voice quality drops and finally system hangs
I tried latest FreePBX with asterisk 13, 14 & 15 also.
OS:
Linux freepbx.sangoma.local 3.10.0-862.2.3.el7.x86_64 #1 SMP Wed May 9 18:05:47 UTC 2018 x86_64 x86_64 x86_64 GNU/Linux

Usage:
200 channels trunk Inbound with 13 queues & 5 extensions in each queue.
Total:
Trunk >> 1
Inbounnd routes: >> 14
Queues >> 14
Extensions >> 70

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@igor_stojanoski wrote:

Hello,

I`m having issue with follow me option to be specific i have problem with Ring Time.

Here is my logic:
Call extension 100 and ring only 1 sec transfer the call to Follow-Me List: ring on external number for 30 seconds If no answer send the call to extension 101.

I have changed Ringtime Default in Advanced Settings to 40seconds and call only stay at extension 100 and ring only external number to extension 100. Call is never send to extension 101.

I have set Destination if no answer > Extension 101

Additionally i have tested with changing timer to 10 seconds and if i use only 10 seconds the call is transferred to extension 101 > external number

Any chance you can tell me why for 30 seconds is not working and for 10 seconds is working.

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@josephchrz wrote:

hello i try to adda mis destination with name and a number. I have IVR setup correctly when i go to that mis destination it says can not be completed. I searched around and I’m much for figuring things out so i got lost. Can someone please help me to figure out what is wrong or What am i doing wrong?

Joseph

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@Jaybk26 wrote:

So I’m trying to reset the root password of our Sangoma FreePBX System 100, but instead of a serial port, there’s an ethernet port labeled “Console.” What steps do I need to take to access it “physically” through the ethernet console port? Thanks in advance!

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@intrigue wrote:

We have a strange intermittent issue with a ring group. This ring group has 2 members and on occasion when a call comes into the ring group, user A will answer and immediately have a busy signal while the continues to ring for user B. User B can pick up the call and the caller is there. This happens both ways, only occurring 3-4 times a week so it has been really hard to troubleshoot. Any suggestions would be appreciated.

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@FreerPBXer wrote:

Not having an issue getting Line 1 configured, registered, and working, but I can’t figure out how to configure Line 2 on the SPA112 in EPM Pro, and it’s dead on the ATA.

TIA!

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@brisbinj wrote:

I have a fresh distro install with sipstation trunks. I can dial out and receive calls no problem. The issue I am having is getting the Asterisk Phonebook to populate either the rapid dial keys or the Contact list of the Digium D50 phones.

I have tried using DPMA enabled under EPM Global with the Digium module disabled as suggested, and the phones will not load.

With DMPA disabled under EPM Global and the Digium module enabled the phones load and function as they should but no contacts in the contact list or on the rapid dial keys.

I also have 3 DID’s and cannot figure out how to get them to load as 3 lines on the line soft keys.

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@wingutechnology wrote:

ISSUE: Cannot hear when dialing direct extensions from external phone source.

I have noticed many similar support requests and searched the web extensively but have not identified a potential solution to this particular scenario. I hope to have provided enough info for my request. This seems to be a firewall issue, but after careful examination, I still have not identified the exact problem especially since some external calls work and others do not.

WHAT WORKS:

1. Internal extension to extension.
   End-user answers call and speaks/hears successfully.
2. External to IVR selection. Example, 1 for Customer Service, 2 for Maintenance, 3 for Billing.
   End-user answers call and speaks/hears successfully.
3. Dialing out.
   End-user makes call and speaks/hears successfully.

WHAT DOES NOT WORK:

1. External to direct dial extension.
   End-user answers call but cannot speak/hear.
2. External to IVR selection, then forwarded to extension if no-answer from end-user to IVR selection.
   Overflow End-user answers call but cannot speak/hear.

To confirm existing issue, I modified Inbound Routes “from” Set Destination–>Time Conditions–>IVR “to” Set Destination–>Extensions–>user’s extension. Thus, an external call direct to extension (bypassing IVR) still has the same result… End-user answers call but cannot speak/hear.

CONFIGURATION:
Router/firewall – Public Static IP
Forwarded/NAT ports to FreePBX server: 5061 TCP/UDP (PJSIP TLS), 10000-20000 UDP (RTP for SIP)

FreePBX 14.0.5.25 Asterisk 13, Installed using full automation, fully updated at this posting
Inbound DID - Any, CID - Any, Destination - Time Conditions/IVR/Direct Dial enabled
All extensions using PJSIP
SIP trunks provided by SIPstation (quantity 3)
FreePBX firewall enabled using wizard

Asterisk Log Snippet (tail -f full)
[2019-03-16 11:41:37] VERBOSE[25798] res_pjsip/pjsip_options.c: Contact 100/sip: 100@107.103.16.29:38892;transport=TLS;rinstance=743cbcddbac10b95 has been deleted
[2019-03-16 11:41:42] VERBOSE[20032] res_pjsip/pjsip_options.c: Contact 100/sip:100@107.103.16.29:38892;transport=TLS;rinstance=743cbcddbac10b95 is now Reachable. RTT: 511.265 msec
[2019-03-16 11:43:39] VERBOSE[20032] res_pjsip/pjsip_options.c: Contact 100/sip:100@107.103.16.29:38892;transport=TLS;rinstance=743cbcddbac10b95 has been deleted
[2019-03-16 11:43:39] VERBOSE[5691] res_pjsip_registrar.c: Added contact ‘sip:100@107.103.16.29:38892;transport=TLS;rinstance=743cbcddbac10b95’ to AOR ‘100’ with expiration of 60 seconds

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