I tried updating through the GUI, but it fails. What are my other options? It is on 13.0.192.8
2 posts - 2 participants
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Installing Updates
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FreePBX - Shared voicemail
Hello,
my environment:
- FreePBX 15.0.37 (mode Extensions)
- iSymphony 3.5.9.8
- Bria like softphone
I have found similar questions that mine but I didn’t find a clear answer. I’d like know if it is possible that a group of extensions/users have their own voicemail and manage a general voicemail (team’s voicemail). I don’t want a VM Blasting (I don’t want a copy of the VM message in the voicemail’s users).
In the end, I’d like that in the iSymphony see like:
thanks by advance.
regards,
1 post - 1 participant
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Call Recording Reports force backup now
the call recordings report failed the last backup with an error of (The requested driver seems invalid)
im looking for a way to force a backup now to free up space on the hard drives and restart this process since its supposed to be weekly but after that failed once it just gave up.
1 post - 1 participant
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Notifications via telegram bot
Hi all!
I want to set up sending notifications about unanswered calls to free pbx. I need notifications to be sent to a telegram group through my telegram bot.
I have recently started to work with freepbx, so forgive me if I don’t understand something.
3 posts - 3 participants
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Regarding the issue with TLS encryption ports being enabled
I plan to map the SIP service on FreePBX to the public network, but directly mapping TCP ports seems a bit insecure.So I plan to map the TLS encryption port to the public network. When I opened the port and was preparing to submit the TLS certificate, I couldn’t find the submission portal. How can I submit the TLS certificate to FreePBX and make it effective?
1 post - 1 participant
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Need help how to configure free pbx with Indian sip providers
Installed free pbx but don’t know how to configure with Indian sip providers. please help anyone who are in India… It’s its is helpful to my job…
Thanks in advance...
2 posts - 2 participants
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Can't set up GSC3506 with freePBX server
Hello,
I’m a newbie in IP telephony and freepbx but still already installed and set the network of about 30 ip phones GRP2604P on FreePBX server that is installed in Proxmox.
But now I’ve got a new task - create an emergency alert system for couple building and a warehouse.
Decided to use existing server and to buy some IP Speakers and/or IP horns.
Bought a Grandstream GSC3506 1-way public address SIP speaker but having a hard time setting it up.
The problem is speaker doesn’t ping freepbx server and freepbx server doesn’t ping speaker. Obviously I can’t register it.
While I can enter speaker web-interface, router can ping speaker, proxmox can ping speaker and vise versa. So the issue looks to happen only with FreePBX server. The speaker is not registering by the server.
15197 [2024-02-15 14:04:13] VERBOSE[2422] res_pjsip_registrar.c: Added contact 'sip:115@192.168.3.30:25163' to AOR '115' with expiration of 3600 seconds
15198 [2024-02-15 14:04:13] VERBOSE[2422] res_pjsip_registrar.c: Removed contact 'sip:115@192.168.3.30:36012' from AOR '115' due to remove existing
15199 [2024-02-15 14:04:13] VERBOSE[2422] res_pjsip/pjsip_options.c: Contact 115/sip:115@192.168.3.30:36012 has been deleted
15200 [2024-02-15 14:04:13] VERBOSE[2422] res_pjsip/pjsip_options.c: Contact 115/sip:115@192.168.3.30:25163 is now Reachable. RTT: 10.754 msec
15202 [2024-02-15 14:04:16] VERBOSE[11060] res_pjsip/pjsip_options.c: Contact 115/sip:115@192.168.3.30:25163 has been deleted
Here are some additional info:
GSC3506
Software Version
Boot
1.0.3.1
Core
1.0.3.1
Prog
1.0.3.8
Locale
1.0.3.5
Res
1.0.3.1
Asterisk Version: 18.16.0
FreePBX 16.0.40.7
Can someone help me to figure this out?
1 post - 1 participant
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P-preferred-identity with CID of the extension needed for TopLink Express Clip-No-Screening outbound
Hi, I want to use TopLink Express as an outgoing provider with clip-no-screening. Toplink support says the PBX (FreePBX 18.16.0) have to send the CID as p-preferred-identity. The used extensions have a valid number in the CID field. Support says no CID is sent as p-preferred-identity.
I found some information related with this problem in this forum, But the information seems to be old, only by editing files or slightly different.
Any good solutions?
1 post - 1 participant
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Custom Inbound Route
My goal is:
- Caller calls my inbound route
- Call has a Diversion header added
- Call is diverted to an 888 number
I made a custom destination that I point my inbound route to:
exten => s,1,Set(PJSIP_HEADER(add,Diversion)=sip:${FROM_DID}@mydomain:5060)
exten => s,n,Dial(PJSIP/8888888888@mydomain:5060)
129445 [2024-02-15 17:42:19] ERROR[4515] chan_pjsip.c: Unable to create PJSIP channel - endpoint 'mydomain:5060' was not found
129446 [2024-02-15 17:42:19] NOTICE[31165][C-0000079c] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
Questions:
- I know I have the @ wrong on my dial what do I need to change it to so it will go to my outbound routes?
- Maybe @david55 will know but how can I amend my set diversion header to include: reason=unconditional;privacy=off
Thanks for the guidance.
4 posts - 2 participants
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Whitelisted IP banned anyway
What am I doing wrong?
System Firewall 15.0.43
System Admin 15.0.33.13
1 post - 1 participant
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Different blacklist for different queues
Hey there, this is my first post here!
We have one asterisk with a blacklist activated, but my boss told me “Can we have different blacklists for each queue?”
I’ve been searching but i didnt get anything, I suppose that I have to manually edit my dialplan, but the thruth is that I have inherited the system and we have certain custom configs that don’t help while trying to modify something
Can someone help me out? Is this even possible to do?
Thanks and regards for any information
1 post - 1 participant
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Can't enble WebRTC Phone for UCP
Good morning,
I just installed FreePBX 16 on Debian, my goal is to enable WebRTC Phone in UCP, I’ve installed all the packages and created an user that can access UCP interface (in user setting “Phone” is setted to “Yes”), I’ve setted a certificate with Let’s Encrypt and still I can’t see the phone icon in the “Add Widget” UCP panel.
Am I missing something here?
Thanks in advance for the help!
1 post - 1 participant
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Getting the end time for a calendar event group based time condition
Hi folks,
I’m trying to get some information about the event that is triggering a time condition and access that information in the dailplan.
I intend to use the end time of the event to play a voice recording letting the caller know when the event is planned to end. Like: “The person you’re trying to reach is currently in a meeting, and is planning to return at HH:MM”.
I’ve figured out how to do the talking part, but somehow I need to pass the event data (the event end time) to the dailplan to parse which numbers to read . That is where I’m stuck. Does anyone know how I can access this information?
I’m using FreePBX 16.0.40.7 with Asterisk Version: 18.20.2
1 post - 1 participant
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Automatic call recording delete script
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Trunk Access Two Different Routers
Good afternoon,
Bare with me on this one.
My provider gives me two routers which give me two numbers with two different Internet packages.
Say the 1st one is 22xxxxxx10 and the 2nd one 22xxxxxx11.
The provider doesn’t split the lines from one router they come from two separate routers as you see.
Both routers have the same gateway 192.168.1.1
Respectively i get two sets of sip credentials from both lines.
1st lets say 22xxxxxx10@xxx.provider.net with its password and the 2nd say 22xxxxxx11@xxx.provider.net with its password.
My goal is to add them both to the FreePBX as trunks.
If i try to use via one router and keep the other out only one account registers and the other fails as it kinda makes sense.
My question is as follows, can this be achieved with only one NIC say to create a sub interface and if so what are the routing steps?
If i add a second NIC and add the second router there will that solve my issue or more routing is needed there also?
Thank you for your time
4 posts - 2 participants
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letsEncrypt Generation problem
Hello everyone who can help me?
I have a problem generating an ssl certificate.
16 posts - 5 participants
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How to ignore calls, that they can be accepted on other clients
Hi
I have a new SIP phone number and I want to set it up so that an audio file is played when calls are received outside my office hours. However, during my working hours, I would like freepbx/asterisk to ignore the call completely, that the phones can accept incoming Calls from other SIP-Clients, which are using the same SIP configuration from the provider, as I added it to freePBX.
For the time conditions, I need a ‘do nothing’ option for when the destination does not match. Is there a workaround for this?
2 posts - 2 participants
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FreePBX servers replication behind a Loadbalancer
Dear FreePBX Friends,
I hope this message finds you well. I am currently working on a project and could use some guidance from our team.
Here’s the setup: I am planning to deploy a clustered load balancer, specifically a Citrix Netscaler, to manage traffic across multiple FreePBX instances. Let’s say we have 6 FreePBX servers configured . My objective is to ensure that all backend FreePBX servers remain synchronized. that means if I create an extension 1001 it should replicate all servers . I don’t need it immediately to be in sync , I can run a CRON job in in every 60 min intervels or do it manually when it requires
During my research, I’ve encountered various options for achieving synchronization, with rsync catching my attention. My question is: If I synchronize the directory containing FreePBX files, will that be sufficient to replicate changes across all 6 FreePBX servers? Or do I need to perform a separate database synchronization?
I would greatly appreciate your insights and recommendations on the best approach for ensuring synchronization among the FreePBX servers in this clustered setup.
Thank you for your assistance.
1 post - 1 participant
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API Web-app type, authentication issue
Hi
I’m trying to setting up a Web-App to handle some API Gql.
Ussing Web-App mode, I’m basing on this part of document.
Just trying this way bellow:
I just replaced the authorization server with http://my_freepbx_erver:83/admin/api/api/authorize
But unable to authenticate.
It returns
| error | invalid_client |
| message | Client authentication failed |
Same result with Single Page (and not secure, so it’s not a good way)
I wondered if somebody else used a web-app mode successully.
That works correctly with machine to machine, but the goal is avoid to store the client_secret on the remote server calling the FreePBX server.
If someone can help me please.
Or maybe it’s a bug.
There is no document on FreePBX help to put us on rails.
1 post - 1 participant
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There was an issue configuring TLS encryption port listening
I plan to enable TLS encryption ports for the SIP service of FreePBX for making VoIP calls directly on the public network. However, I encountered an issue during configuration. The TLS encryption port section in the pjsip.transports configuration file defaults to only listen to all IPv4 ports, but I want port 5061 to work directly on both IPv4 and IPv6 protocols to make up for the low IPv6 activation rate in my region, According to the Google search engine, it was found that 0.0.0.0:5061 can be changed to [:]: 5061, but doing so will cause another problem: the original IPv4 will no longer listen,
So can I ask everyone how to solve this tricky problem correctly?
1 post - 1 participant
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