@sandysp4u wrote:

Hello Everyone,

I am new to IPPBX we bought a new commercial version of sangoma pbxt 100. I have terminated SIP and we are able to receive and do the call from inside to outside and from outside to inside, i have also created a group and matched it with a inbound route. I have configured fallow me feature for extension and its working fine when an internal ext call internal ext the calls get diverted to external phone number when its not picked but when external number calls the ext receive the call and if he don’t pick up, the call is not getting diverted to his external phone number.
I have searched entire forum and tried many things but nothing worked out for me.
any kind of help is really appreciated.
Regards,
Sandeep

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@awh wrote:

I’d like to notify Microsoft Teams when an agent logs in or our of a queue. I can’t find any way to do this. My idea right now is to create a custom extension and have agents dial that to join/leave the queue, but I want to make sure that I’m not overlooking a more simple solution first.

Thanks!

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@oTTo001 wrote:

Hi,
I am having a FritzBox 7490 with recent firmware and a freepbx 15.0.16.75
The Fritz is connected to POTS and forwarding it to asterisk as a pjsip trunk, what is working fine.
Now, there is an (analogue) phone connected to the Fritz as FON1 which I would like to use as extension in asterisk. (I have some FritzBoxes in the network and this is working fine).
On this particular Box the registration of the extension fails (I suppose this is because the same IP is already registered as trunk in asterisk). I am having the following in full log then:

[2020-10-15 07:22:42] WARNING[16109] res_pjsip_registrar.c: AOR '' not found for endpoint 'Fritz_pj_66666666'
[2020-10-15 07:22:42] SECURITY[10939] res_security_log.c: SecurityEvent="RequestNotSupported",EventTV="2020-10-15T07:22:42.177+0200",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="Fritz_pj_66666666",SessionID="BD77EE1C44304207@192.168.1.169",LocalAddress="IPV4/UDP/192.168.1.13/5060",RemoteAddress="IPV4/UDP/192.168.1.169/5060",RequestType="registrar_requested_aor_not_found"

where Fritz_pj_66666666 is the name of the trunk and 192.168.1.169 the IP of this FritzBox. I have tried to setup this extension as pjsip and as sip as well.
Does anyone know this problem? Should this work? How?
If I can provide any further information or logs please let me know.

Best regards and thanks in advance,
Otto

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@monta wrote:

I receive 2 calls simultaneously on the grandstream gxp2160 phone: 1 from trunk and 1 from another extension. How do I prioritize the extension and its ringtone? I understand that the first incoming call defines the ringtone but I would like to change the sound if the call from the extensions arrives later.
Sorry for my english.
Thank you very much
Andrea

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@franciswurtz wrote:

Hello, first sorry for my poor english. I have a newbie question.

I am a new user of FreePBX which I love. We have an internal server with the latest free version of Sangoma FreePBX. I have configured our old Linksys (now Cisco) model SPA942 brand phones and everything has been working fine so far except for a small bug. When a voice message arrives in the box, the lamp indicating the presence of message (WMI) suffers from a delay of up to 1 hour, which in my opinion corresponds to the moment when the phone will re-register with the server. Same thing when we delete the message. I have been able to work around the issue by forcing the phones registration expiration to 20 seconds instead of 3600 (see creenshot). I know this is probably not the best solution, but so far everything seems to be working fine.

Do you have any possible solutions to suggest to me to resolve this type of problem? Do you think that forcing phones to reregister in such a short time can bring problems?

Thanks in advance for your help!

screenshot.png716×151 21.3 KB

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@bajramia wrote:

Hi All,
We are having issue that the remote caller ID is not showing on transfer. So if one of my coworker transfer a call to me I see his ID and not the transferring ID

Current Asterisk Version: 16.11.1

Module | Version | Status | License |
±--------------------±-----------±----------------------------------±------- -----+
| accountcodepreserve | 13.0.2.2 | Enabled | GPLv2 |
| amd | 13.0.3 | Enabled | GPLv3+ |
| announcement | 13.0.7.8 | Enabled | GPLv3+ |
| areminder | 14.0.4.20 | Enabled | Commerc ial |
| arimanager | 13.0.5.4 | Enabled | GPLv3+ |
| asterisk-cli | 14.0.2 | Enabled | GPLv3+ |
| asteriskinfo | 13.0.7.2 | Enabled | GPLv3+ |
| backup | 14.0.10.11 | Enabled | GPLv3+ |
| blacklist | 14.0.3 | Enabled | GPLv3+ |
| broadcast | 14.0.1.13 | Enabled | Commerc ial |
| builtin | | Enabled | |
| bulkhandler | 13.0.18 | Enabled | GPLv3+ |
| calendar | 14.0.3.9 | Enabled | GPLv3+ |
| callaccounting | | Not Installed (Locally available) | Commerc ial+ |
| callback | 13.0.5.5 | Enabled | GPLv3+ |
| callerid | 13.0.8.20 | Enabled | Commerc ial |
| callforward | 14.0.1.5 | Enabled | AGPLv3+ |
| calllimit | 13.0.5.7 | Enabled | Commerc ial |
| callrecording | 14.0.16 | Enabled | AGPLv3+ |
| callwaiting | 14.0.1.1 | Enabled | GPLv3+ |
| cdr | 14.0.5.22 | Enabled | GPLv3+ |
| cel | 14.0.4 | Enabled | GPLv3+ |
| certman | 14.0.17 | Enabled | AGPLv3+ |
| cidlookup | 14.0.1.12 | Enabled | GPLv3+ |
| conferences | 13.0.23.17 | Enabled | GPLv3+ |
| conferencespro | 14.0.2.13 | Enabled | Commerc ial |
| configedit | 13.0.7.1 | Enabled | AGPLv3+ |
| contactmanager | 14.0.5.14 | Enabled | GPLv3+ |
| core | 14.0.28.80 | Enabled | GPLv3+ |
| cos | 13.0.12.7 | Enabled | Commerc ial |
| customappsreg | 13.0.5.7 | Enabled | GPLv3+ |
| cxpanel | 14.0.4 | Enabled | GPLv3 |
| dahdiconfig | 14.0.1.6 | Enabled | GPLv3+ |
| dashboard | 14.0.10 | Enabled | AGPLv3+ |
| daynight | 14.0.1 | Enabled | GPLv3+ |
| dictate | 13.0.5 | Enabled | GPLv3+ |
| directory | 13.0.19.12 | Enabled | GPLv3+ |
| disa | 13.0.6.12 | Enabled | AGPLv3+ |
| donotdisturb | 14.0.1.1 | Enabled | GPLv3+ |
| dundicheck | 2.11.0.3 | Enabled | GPLv3+ |
| endpoint | 14.0.56 | Enabled | Commerc ial |
| extensionroutes | 13.0.10.7 | Enabled | Commerc ial |
| extensionsettings | 13.0.4 | Enabled | GPLv3+ |
| fax | 14.0.2.9 | Enabled | GPLv3+ |
| faxpro | 14.0.16 | Enabled | Commerc ial |
| featurecodeadmin | 13.0.6.4 | Enabled | GPLv3+ |
| findmefollow | 14.0.1.28 | Enabled | GPLv3+ |
| firewall | 13.0.60.14 | Enabled | AGPLv3+ |
| framework | 14.0.13.40 | Enabled | GPLv2+ |
| freepbx_ha | 13.0.11 | Enabled | Commerc ial |
| hotelwakeup | 14.0.1.6 | Enabled | GPLv2 |
| iaxsettings | 14.0.1.4 | Enabled | AGPLv3 |
| infoservices | 13.0.1.4 | Enabled | GPLv2+ |
| irc | 13.0.1 | Enabled | GPLv3+ |
| ivr | 14.0.9.9 | Enabled | GPLv3+ |
| languages | 14.0.1.5 | Enabled | GPLv3+ |
| logfiles | 13.0.10.10 | Enabled | GPLv3+ |
| manager | 13.0.2.7 | Enabled | GPLv2+ |
| miscapps | 13.0.3.2 | Enabled | GPLv3+ |
| miscdests | 13.0.9 | Enabled | GPLv3+ |
| music | 13.0.22.8 | Enabled | GPLv3+ |
| oembranding | 14.0.9.67 | Enabled | Commerc ial |
| outroutemsg | 14.0.1 | Enabled | GPLv3+ |
| paging | 14.0.16.5 | Enabled | GPLv3+ |
| pagingpro | 14.0.2.26 | Enabled | Commerc ial |
| parking | 13.0.19.11 | Enabled | GPLv3+ |
| parkpro | 14.0.2.11 | Enabled | Commerc ial |
| phpinfo | 13.0.2 | Enabled | GPLv2+ |
| pinsets | 13.0.13 | Enabled | GPLv3+ |
| pinsetspro | 13.0.9.14 | Enabled | Commerc ial |
| pm2 | 13.0.7.1 | Enabled | AGPLv3+ |
| pms | 14.0.2.61 | Enabled | Commerc ial |
| presencestate | 14.0.1.10 | Enabled | GPLv3+ |
| printextensions | 13.0.3.2 | Enabled | GPLv3+ |
| queuemetrics | 2.11.0.3 | Enabled | GPLv3+ |
| queueprio | 13.0.7 | Enabled | GPLv3+ |
| queues | 14.0.2.34 | Enabled | GPLv2+ |
| queuestats | 14.0.1.43 | Enabled | Commerc ial |
| qxact_reports | 14.0.7.25 | Enabled | Commerc ial |
| recording_report | 14.0.3.13 | Enabled | Commerc ial |
| recordings | 13.0.30.14 | Enabled | GPLv3+ |
| restapi | 13.0.21.2 | Enabled | AGPLv3 |
| restapps | 14.0.22.8 | Enabled | Commerc ial |
| ringgroups | 14.0.1.15 | Enabled | GPLv3+ |
| sangomacrm | 14.0.25.25 | Enabled | Commerc ial |
| setcid | 13.0.6.3 | Enabled | GPLv3+ |
| sipsettings | 14.0.27.29 | Enabled | AGPLv3+ |
| sipstation | 14.0.4.3 | Disabled | Commerc ial |
| sms | 14.0.4.9 | Enabled | Commerc ial |
| soundlang | 14.0.9 | Enabled | GPLv3+ |
| superfecta | 14.0.27 | Enabled | GPLv2+ |
| sysadmin | 14.0.38.21 | Enabled | Commerc ial |
| timeconditions | 14.0.2.21 | Enabled | GPLv3+ |
| tts | 13.0.13 | Enabled | GPLv3+ |
| ttsengines | 13.0.7.5 | Enabled | AGPLv3 |
| ucp | 14.0.3.13 | Enabled | AGPLv3+ |
| userman | 14.0.15 | Enabled | AGPLv3+ |
| vega | | Not Installed (Locally available) | Commerc ial+ |
| vmblast | 13.0.11 | Enabled | GPLv3+ |
| vmnotify | 14.0.1.9 | Enabled | Commerc ial |
| voicemail | 14.0.6.13 | Enabled | GPLv3+ |
| voicemail_report | 14.0.6 | Enabled | Commerc ial |
| vqplus | 14.0.4.20 | Enabled | Commerc ial |
| weakpasswords | 13.0.2 | Enabled | GPLv3+ |
| webcallback | 13.0.11.5 | Enabled | Commerc ial |
| webrtc | 14.0.3.8 | Enabled | GPLv3+ |
| xmpp | 14.0.1.20 | Enabled | AGPLv3 |
| zulu | 14.0.58.9 | Enabled | Commerc ial |

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@mike_b wrote:

Some companies give you more than 10 digits, for example if they use an alpha-numeric phone number (for example, 800. callthis). I just ran into an issue where the phone would not connect if I put in the whole string (“number cannot be dialed as is”), but it does connect if I put in 10 digits. (in this case 800-callthi). I have used FreePBX for years and have never run into this issue (although I can’t remember if I ever dialed more than 10 digits). Is this a new “feature”? Do I need to change my calling plan to allow more than 10 digits?

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@Maico wrote:

Good afternoon, don’t notice my english because I’m using google translator .

I have a problem with the external connections of my FreePBX 15.0.16.75, all the calls I make from outside last only 30s, even * 43 the echo test after 30s hangs up the call.

This is the sip settings configuration. I changed the sip port from 5060 to 8765, to test and still giving the problem.

image.png1191×678 44.4 KB

This one is my Firewall that is running in a hundred, where it is doing nat for Freepbx.
iptables -t nat -A PREROUTING -i $WAN1_eth -p tcp --dport 10000:20000 -j DNAT --to-dest 10.10.10.199
iptables -t nat -A PREROUTING -i $WAN1_eth -p udp --dport 10000:20000 -j DNAT --to-dest 10.10.10.199
iptables -t nat -A PREROUTING -i $WAN1_eth -p tcp --dport 8091 -j DNAT --to-dest 10.10.10.199:8765
iptables -t nat -A PREROUTING -i $WAN1_eth -p udp --dport 8091 -j DNAT --to-dest 10.10.10.199:8765

Here is the pjsip set logger on.

CLI log

And if I make any connection or echo test on the same network where the serivador works normally without a problem.

Thank you.

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@JonW wrote:

Hello,

I need to upgrade a customer from FPBX 12.0.76.6 to the latest FPBX 15 version. However, I cannot upgrade the production server as we have attempted twice and have been met with issues each time. As such, is there a way to perform the upgrade on a lab environment server, while leaving the production server untouched and function (i.e. not de-activating deployment ID)? I know that Digium allow multiple coterminous registrations for this purpose.

Any help is greatly appreciated!
Jon

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@jasmantle wrote:

FreePBX 13. For a remote site (50 year old copper unable to provide acceptable Internet), I have a WiFi link to a building, acriss an airfield, to provide our main Internet service. It is (has been) astoundingly reliable and consistent (and affordable). But every once in a while, it will drop, then come back a few seconds later.

To protect against drops for business-critical data (including VOIP), a router, pre-WiFi, will fail-over to (expensive) LTE data, and then fail-back when the primary link has been recovered.

During a failover, VOIP continues. FreePBX reached out to the VOIP supplier through the LTE IP address, both sides reconnect, and life continues.

After a fail-back, and depending on timing, we reconnect to the VOIP supplier, and we can make outgoing calls. But the VOIP supplier cannot reconnect to us, so we miss incoming calls. I suspect this blockage is an combination of timing, open ports, and firewalls closing ports which are perceived to be idle.

However, if I SSH in and issue “fwconsole reload” then two-way communication is re-established.

The Question (finally): Is there a way, though the web GUI, to do a Reload (other than an Apply Config, after updating modules)? I have admin staff at this site that can handle a Web interface and can trigger a reload, who are not going to be able to SSH to a server and use a command-line interface.

Jim

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@jpbrubaker wrote:

Greetings all, FreePBX newbie here

I’m running FreePBX 13.197.22 at the moment.
PBX Distro 10.13.66-6
Asterisk 11.20.0

I usually update modules on a regular bases. But today, I went in to System Admin and saw there is an upgrade to the PBX Distro. It’s 10.13.66-22

Is it “safe” to run this update? Is it needed?

Sometime in the next six months I would really like to upgrade FreePBX to a newer version all together but that’s a different topic.

Thanks!

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@iversa wrote:

I have setup a FreePBX on vultr.com. After going through the setup wizard and enabling the firewall I’m locked out now. I think forgot to whitelist my IP

I still have access to the command using the Vultr interface. How can I whitelist my IP to gain access again to the portal?

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@deano wrote:

I have 2 deployments that I need to upgrade. In prep or the upgrade, I choose to do a fresh install of 10.13.66-22, update it all and then activate it. They attempt the upgrade. I complete the upgrade and each time, 7 times now, I can’t access the web gui. I have scoured the websites, google and this site and still no such luck. I have tried the fwconsole chown, I have verified the permissions in the database with the .conf file. Still no luck. Has to be permissions related, cause will update the URL from just the ip to /admin/config.php

Anyway, any help would be appreciated. I’m not so sure I trust the restore of backup onto a newer version.

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@ilouiemiami wrote:

Guys, I installed an SSL from Let’s and it went smoothly, however, my issue is that when I call the domain without the HTTPS, all the browsers show it as unsecured. Is there something else I’d need to install to be secured?.. TY

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@elettrasas wrote:

Hi guys! I want to share with you a project i am working on!

Let’s say in your home or firm you only have a CardDAV server for managing your contacts but you also want to get them on your phone… Well basically you can’t.

So i have created this library that allows you to:

1. Read a CardDAV server and get back a XML phonebook to use with your phones
2. Set the inbound CNAM with the help of Superfecta CID
3. Set the outbound CNAM with the help of the homonym module

More info on my GitHub Page. I’m planning to make it even more integrated with FreePBX (making it a module) and so on.

The library is totally free for personal use (if you want to use it commercially please contact me) so if you like it, you can donate. I greatly appreciate any help.

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@Maico wrote:

Hi guys.

I have 3 scripts to create to perform an integration with my system, but every time I restart Freepbx 15 the chmod 755 scritp.sh permissions are lost. I already tried to put the command chmod 755 /home/asterisk/scripts/scripts.sh in /etc/rc.d/rc.local, but it seems that when he runs fwconsole chown he ends up losing his permission.
Would you like to add these scripts to fwconsole?
thank you

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@WarTech wrote:

In FreePBX 10.13.66-22, where can I set the default outgoing caller ID?
I don’t want to type the company name and reception phone number for every extension.

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@zzzzzxx wrote:

We have an office with a working w52p handset and extension. We recently bought 2 more handsets to register / pair with the base station which we have done. But how to we add 2 more extensions to the free pbx side? Do we add them like normal and select Brand Yeahlink and model W52p? Do we use the original mac address of the base station as I dont believe individual handsets have mac addresses?

On the YeahLink Gui of the base station we have the account 2 and account 3 already defined and registered to the base, Just missing the free pbx side of things…

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@oliviatrott099 wrote:

Hi everyone,

I have been using Avoxi for 3 months now and I’m really annoyed with the voice quality of their hosted phones. Sometimes the voice is really choppy, and in the worst cases, some calls are automatically dropped. This really presents a negative image of our business to potential customers.

At the same time, their support service proved to be a bit of a disappointment as well. So I’m looking for some fine alternatives. I did do a bit of research and found Acefone, which fits our budget perfectly.

Looking forward to your valuable suggestions. Feel free to let me know about your service providers too.

Best Regards,

Olivia Trott

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@tech_conmet wrote:

Good evening,

my freepbx install (15.0.16.75) is activated, goes on internet, firewall is disabled, but cannot install any new language from console neither GUI.

This is console output:

fwconsole sound --install it

In Soundlang.class.php line 1272:

  <html>
  <head><title>524 Origin Time-out</title></head>
  <body bgcolor="white">
  <center><h1>524 Origin Time-out</h1></center>
  <hr><center>cloudflare-nginx</center>
  </body>
  </html>

sounds [--list] [--listglobal] [--install INSTALL] [--uninstall UNINSTALL] [--global GLOBAL]

Any advice?

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