@kyiu wrote:
I'd like to enable firewalld on my freebpx14. For some reason the firewalld got turned off (stopped) after the system is started. The default firewall was not enabled during the installation.
Does Freepbx use a separate file to control start/stop of daemons? I've configured the firewalld (daemon) to start automatically at boot. Something else stops the daemon after the system has started. Can someone please help?
Posts: 1
Participants: 1
@ymartinez wrote:
Good Day,
I am creating this topic because our Digium Phones are not taking the correct time zone when connecting to FreePBX server. Every time we reboot the phones, they show December 31st. Phones are configured via EPM module using diferent templates. We have others brands and they are OK.
We found a similar topic: Https://community.freepbx.org/t/digium-phones-date-and-time/20638
Please help, we have more than 20 phones facing the same issue.
Posts: 1
Participants: 1
@LamerDrv wrote:
Hi.
My setup is FreePBX (13.0.192.16) + Cisco Unified Call Manager Edition (CME).
I had created trunks on both. I can make calls in both directions. What I want is displaying callee name when I make call. I need Remote-Party-ID. On FreePBX I set
trustrpid=yes
sendrpid=rpid
in extension properties and in trunk properties.
If I make call from FreeBPX to CME then trasfering Remote-Party-ID to caller phone works normally.If I make call from CME to FreePBX it doesn’t work.
While SIP debug I see that picture:In point 1: FreePBX sends SIP ringing (180) WITHOUT Remote-Party-ID to CME. CME forwards this sip-ringing to caller phone adding Remote-Party-ID containing only phone number. Caller phone gets Ringing with Remote-Party-ID which contains number but DOESN'T contain name.
In point 2: called phone send SIP ringing (180) WITH Remote-Party-ID (containing proper name) to FreePBX. FreePBX forwards this sip-ringing to CME, but CME doesn't forward this message to caller phone.
I debug local call on FreePBX (between two phones registered on FreePBX). During local call FreePBX send proper Remote-Party-ID in very first ringing(180) sip-message.
How can I get the FreePBX to include the proper Remote-Party-ID in first ringing (180) sent to trunk in point 1 ?
Posts: 1
Participants: 1
@unison wrote:
Hi
We use a bond (bond0) that consists of eth0 and eth1, to provide protection from a ethernet port/cable or switch failure.
We are using mode 4 (802.1ad/lacp), and our configuration looks like:
/etc/sysconfig/network-scripts/ifcfg-bond0
DEVICE=bond0
BONDING_OPTS="mode=4 miimon=100 lacp_rate=1"
BOOTPROTO=static
ONBOOT='yes'
IPADDR=192.168.15.11
NETMASK=255.255.255.0
GATEWAY=192.168.15.1We would like to be able to use the systemadmin interface to make changes to the interface, however if we do that, the configuration that is created by systemadmin drops the BONDING_OPTS line.
Is there any method that is available to ensure that is retained?
Cheers
Posts: 1
Participants: 1
@Madhava wrote:
Platform: Distro SNG7-FPBX-64bit-1707-1 with FreePBX 14 and Asterisk 14 as Standard Installation
Hardware: Lenovo IdeaCentre Q180 with 4GB RAM and 500GB HDD and Intel Atom D2700 @ 2.13 GhzIssue: System wont boot or display any error but simply seen cursor blinking after the BIOS lenovo logo. System boots normally without any key press or intervention if the USB memory stick used for installing the system is plugged-in.
There were no installation time errors and everything finished smoothly.
Any help to fix this will be greatly appreciated.
Posts: 5
Participants: 2
@vbman213 wrote:
I am getting "bad destination" errors after disabling voicemail on extensions.
Under Optional Destinations: I see "Error" and "Bad Dest: goto0". This repeats for Busy and Not Reachable.
Posts: 1
Participants: 1
@Thomas233 wrote:
Hello,
I have a FreePb v14 system running. It works great
But there is one thing I could not yet really get over to.I have tried to generate a Let`s Encrypt certificate for this machine using the FreePbx administration web frontend for certificates (Admin -> Certificate Manager -> Create New Lets Encrypt Cert).
This are my systems details (values slightly modified ONLY SIMILAR EXAMPLE FOR DEMONSTRATION!):
Hostname: srv01.xyz.com
IP: 218.17.123.123
Version: FreePBX 14.0.1.4 (from latest official distro)DNS entry for xyz.com:
A 3600 srv01 218.17.123.123Lets Encrypt cert details:
Certificate Host Name: srv01.xyz.comThis is the error output on generation:
There was an error updating the certificate: Error 'Requested host 'srv01.xyz.com' does not resolve to '91.130.242.22' (Found 213.17.123.123)' when requesting "srv01.xyz.com//.freepbx-known/c70667a06e8fb13d35fb770ddc2c0023"The strange thing is, I can reach the URL srv01.xyz.com//.freepbx-known/c70667a06e8fb13d35fb770ddc2c0023 without any problems from outside with any browser and the response does come from the host/apache with the IP 213.17.123.123.
Firewall module is enabled and configured on the pbx, the needed LE exclusions were made.
The IP which was resolved by the LE module "91.130.242.22" seems to be the last IP before my ISP hands over the connection to my network.
In my network, the pbx device is in a DMZ and public IP is set for it.
I have no other active firewall. I can also not image that my ISP blocks any ports as all parts of the device are connective (ssh, apache, ssl etc.) from outside.How does the certificate manager resolve the IP, does it need any further settings ?
Why does it resolve to the last gateway of my ISP and not to that what is set in the DNS entry ?I am curious about that. Please help ! Thank you !
Posts: 4
Participants: 3
@eggythetech wrote:
I have an odd thing happening on one server.
We have Freepbx Distro and use the UCP to allow users to update their phone BLF and Speediales .
We use Polycom Phones and it works on 10 Servers running Freepbx PBX Firmware:10.13.66-21
I have one server that is giving me an error in UCP under device manager
"Only licensed for Sangoma devices"
i don't know why this server says this?
i have the latest licenses.
Posts: 2
Participants: 2
@CrainBramp wrote:
So I've looked around a bit, but am not finding any answers to this particular question. I've been trying to install the FreePBX distro on ESXi, but the install always comes up corrupt, missing fwconsole and a number of other items. Looking around, this seems to be because my setup doesn't have DHCP, and uses static IPs only, thus the install isn't getting any outbound access.
Is there a way to set, prior to install, a static IP?
Thanks,
Don
Posts: 3
Participants: 3
@zillion08 wrote:
Hi Everyone,
The answer is probably right under my nose.
I have static users in a queue. When one of them is on the phone, and someone calls into the queue, the user cannot hangup their phone (Say they are done when their original call) and pickup the person in the queue. They have to wait for the retry.
Simple Terms of what they want: When they are done talking, they want to be able to hang up and their phone will ring right away.
I know I set the Retry to 10. But that isn't fast enough.
Any help would be appreciated!
Thanks everyone.
FreePBX: 13.0.192.18
Phones: Sangoma s500
Posts: 2
Participants: 2
@mitterhuemer wrote:
Hello,
i started learning Developing FreePBX Modules with MVC Model a few weeks ago.
My Project is a new simple Endpoint Manager which should support Third Party Phones from Manufacturers that are not supported by sangoma.
The first brands i want to add are Gigaset and some Panasonic phones.
The first release should only have basic Features
- Adding new Extensions
- Rebuilding Configs
- Update Phones by SIP Notify
- Very Simple Basefile Editor
- __ PLACEHOLDERS __ for the most important sip Settings
- Default Support for TLS/SRTP if the phones support this. The Port Calculation is already done.
- Multi Account Support with IPEI for DECT Systems.
My developement already can build config files for the Manufacturers and the CFG Files with mac adress.
Future Plans (if i did not already give up
)
* Function Keys with templates without Basefile edit
* Brand Settings page to set specific values without modify the Basefile manual
* Firmware Management
* Simple Background image ManagementWhat does the community think about that idea?
I already have provisioned my first Aastra Phone with my own Endpoint Manager, but this was just a test.
They will not be added to my EPM because they work already perfectly with the Commercial EPMA first Video of the Graphical overview, i use the FreePBX Framwork based on the "Hello World" Application.
The Config creation is not inside the Video but already works with console commands.I want to start making a new epm. The OSS EPM is totally overloaded, i think it has already too much features and broken things. This EPM should work together with the commercial EPM and not getting in conflict with.
As i said this is a first idea of EPM. Maybe it will never be released. Maybe i get it working well enough to think about a release
Regards,
Posts: 9
Participants: 3
@davidber wrote:
I just installed two different installs on Vultr using the ISO. Both are 14.0.1 configs.
Everything is fine but when I go to the UCP to upload .wav files for Voicemail I get a "There was an error. See the console log for more details"
I am lost.
I uploaded a .wav for the IVR and that went fine.
I uploaded two .wav for announcements and that went fine.This seems specific to UCP.
To make sure I was not crazy, I went into the user management -> user -> UCP -> Voicemail and made sure everything was set to yes. I also tried .mp3, .wav, .m4a to make sure it was not a conversion issue.
One of the systems, I upgraded all the modules in the admin section, the other is default.
Any clues?
Posts: 1
Participants: 1
@xvart wrote:
I have an application where I don't want voicemail at all, is there a way to disable this globally or from a database update, wading through 300 extensions isn't my idea of a fun.
Posts: 1
Participants: 1
@lynsix wrote:
This is running on RHEL 7, with Asterisk 14 (latest version) and version 14 of Freepbx
I followed Asterisk install from source from:
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+SourceI seem to have worked around most issues that arose from this setup however I can't seem to figure out how to get past this. When running ./install -n from my freepbx folder I get the following:
Checking if SELinux is enabled...Its not (good)!
Reading /etc/asterisk/asterisk.conf...Done.
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Error!
Could not determine Asterisk version (got: Unable to open specified master config file '/etc/asterisk/asterisk.conf', using built-in defaults).
Please report this.
Posts: 18
Participants: 3
@JimFromRamJack wrote:
This just came up. We maintain offices & call people in several states from a central Call Center, & present the CallerID of the office in the state in question, based on Area Code. Y'all helped me get this fixed earlier this year. It works fine.
Here's a problem: If a callee has a cellphone registered in State A, but they've moved to State B & haven't arranged a new cell # yet, FreePBX will treat them (by Area Code) as if they're still in State A. This means when we call them, the CallerID looks out-of-state, so it's less likely they'll pick up.
We know (from studies & reports) that people are strongly DISinclined to pick up a toll-free CallerID, and somewhat disinclined to pick up calls from out of state. So even if we match this callee's State A Area Code like the scammers do, since they're in State B, it will look "out of state" & they might not pick up.
Finally a question: Is there any way to programatically set outbound CallerID other than by some part of the phone number?
Posts: 6
Participants: 3
@simons wrote:
Good afternoon,
I'm trying to replicate an older PC which has Trixbox 2.6.2.2 installed on it but cannot find the idiots guide to installing it on a new PC.
I have the 2.6.2.2 downloaded as an ISO so just need to know what to do next!
Regards
Simon
Posts: 4
Participants: 3
@ladilla2 wrote:
Hello. I'm trying to fix an issue we have in our office phone. On a regular basis, the caller doesn't hear the callee or viceversa.
We tried several times until this issue was reproduced and we were able to record the details of the call (picture attached). We ran the command "tcpdump -s 0 -i any -w sip-trace.pcap" to obtain this information. The caller is 787-431-2415 and the callee is 813-879-6800. The callee transferred the call to extension 316. At this point the person at extension 316 could hear the caller, but the caller (787-431-2415) couldn't hear the callee.
Thank you.
Posts: 1
Participants: 1
@mokanman wrote:
The user guide states that call waiting "settings are controlled in the Feature Codes module, Extensions module, and Advanced Settings module". I have an extension set up with an analog phone attached to a Grandstream HT503. The HT503 in this case acts as a conduit between the analog phone and the PBX (RASPBX to be more specific). The HT503, being both an FXO as well as FXS device also acts as a conduit between a POTS line and the PBX as well. All of this functions very well with the exception of Call Waiting.
Call waiting is enabled for the Extension (and the Extension page shows this) and CW is enabled in the Advanced Settings page. The Feature Codes page, however, does not show the Callwaiting section as indicated in the user guide.
I do not have POTS call waiting and realize that if I am on a POTS call that another call to that POTS line will result in a busy signal for the caller. What I am looking for is to be notified of an incoming POTS call if I am on a digital call or be notified of an incoming digitial call if I am on a POTS call. The current setup sends the call to voicemail which is acceptable but not preferable.
I am unsure as to whether the lack of these settings on the Feature Codes page is due to the use of an ATA or some other setting that I must modify. Hopefully someone here can point me in the right direction.
Posts: 1
Participants: 1
@wa4zlw wrote:
I completely reloadede the PBX from the ISO on Vultr and first thing I did was upgrade the system then the freepbx modules. I added in my extensions and then trunks. Then inbound routes. When I try and generate outbound routes the DIAL PATTERNS tab is missing information and also the wizard doesnt generate anything except [object] please explain how this can be rectified as I am totally down now
Posts: 1
Participants: 1
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