@KermiTT wrote:
Hello,
I’m new at FPBx, I’ve set up a trunk and it’s working great for incoming/outgoing calls.
I was wondering how to set up voicemail notifications when an extensions have a new VM.
Do I need to have a mail server?
I’ve just set up the basic information at the extension settings, I mean I’ve enabled the VM notifications and type an email but nothing happens when I left a new VM
Thanks in advance
Posts: 4
Participants: 3
@Bradbpw wrote:
I’ve posted this before, but the post seems to be deleted with no explanation. Hopefully that’s a glitch and this is ok to post.
I am receiving this error:
There was an error updating the certificate: Error ‘Requested host ‘MyDeploymentID.deployments.pbxact.com’ does not resolve to ‘xxx.xxx.xxx.xxx’ (Found yy.yyy.yy.yyy)’ when requesting http://MyDeploymentID.deployments.pbxact.com//.freepbx-known/WholeBunchOfNumbersAndLetters_HidingInCaseTheyAreSecure
In the error above, xxx.xxx.xxx.xxx is my current IP address. The IP address it found, yy.yyy.yy.yyy, is a previous IP address of the server (We got a new ISP). It seems like the error is saying that the DDNS address (MyDeploymentID.deployments.pbxact.com) is trying to direct to the old IP address. But my DDNS settings in system admin and on the Sangoma portal both point to the new IP address, xxx.xxx.xxx.xxx. I tried deleting the old certificate and tried creating a new one (using let’s encrypt) and I received the same error.
I’m far from an expert, but it seems like the deployment address from the DDNS service hasn’t updated to my new IP address.
Any idea how to fix this?
Posts: 2
Participants: 2
@bradlarose wrote:
So I have enabled fax options under the user. I have enabled listen for fax in the inbound route…
I send a fax and FreePBX answers… detects it is a fax and doesn’t forward it to the destination but receives the fax…
I got into the UCP and there is no widget for faxes. So where do I view the fax that the system received.
Posts: 2
Participants: 2
@meisner wrote:
I did a search for ‘UCP widgets’ (thinking that’d be general enough) and did not find an answer.
I upgraded from Raspbx/FreePBX v13 to v14. There was a problem upgrading UCP due to ICU dependencies. I got that figured out and (post upgrade) I upgraded UCP to v14. Everything went OK during the upgrade.
But now I have no side bar widgets. I do have Call History and Call Events dashboard widgets, but noting under side bar widgets. Can someone shed some light here? It seems like there should be a number of widgets available when reading the docs.
Posts: 1
Participants: 1
@heminole4life wrote:
Asterisk Version: 13.14.0
FreePBX Version: 10.13.66-18
I have children that have a softphone installed on their ipods.
My goal is create an outbound route that has a time condition where no outbound calls can be made between 10pm and 6am for their extensions.
I’ve created the time condition and group
In extensions_custom.conf I added the below lines[from-child]
exten => _1NXXNXXXXXX,1,goto(from-internal,0001${EXTEN},1)
exten => _NXXNXXXXXX,1,goto(from-internal,0001${EXTEN},1)
exten => _NXXXXXX,1,goto(from-internal,0001${EXTEN},1)
include => from-internalI have the outbound route to strip the 0001 but did doesn’t look like it strips it.
I have looked all day for the answer and haven’t found too much. Then again I could be way off with setting this up correctly.
Posts: 1
Participants: 1
@cbfs wrote:
I updated FreePBX recently and the behavior of call recordings changed. After the upgrade, calls are only being recorded once answered. There are no ringback tones, the recording starts as soon as the extension is picked up. Also, IVRs are no longer recorded.
Does anyone know which module changed the call recording behavior? I’m not sure if this is controlled by the call recordings module or core module.
Is there any way to revert this in config files (or provide a setting) so that call recordings can be started immediately, including IVRs, before the extension is answered?
Thanks!
Posts: 2
Participants: 2
@jeff.wong wrote:
I have separately tested “Directory” & “Google TTS” (Text to Speech) both are working well.
My question is how to use Google TTS for Directory instead of default flite speech engine?When Directory return with several results, caller can press # for the list of extension and flite will read the extension’s name for them, but flite speech is too strange.
Posts: 1
Participants: 1
@Beachtech wrote:
I have an extension (222) that I want to limit to only be able to receive calls from three extensions (301, 302, & 303). I want to prevent other extensions in the company from being able to call 222. I don’t know what the term for this is otherwise I would have searched for it.
The systems is running FreePBX 13 and Yealink VoIP phones.
Posts: 1
Participants: 1
@jtharveyjr wrote:
I have tried to get a Panasonic KX-TGP500 SIP DECT Phone to work with a Sangoma PBXact UC40 to no avail.
I have ensured that the Panasonic settings match the settings in the FreePBX wiki and that the extension is also set per the wiki though I am getting no where.
extension is 4003
Here are the settings and any assistance would be greatly appreciated.
Registrar Server Address 192.168.1.15
Registrar Server Port 5060Proxy Server Address 192.168.1.15
Proxy Server Port 5060Presence Server Address 192.168.1.15
Presence Server Port 5060Outbound Proxy Server
Outbound Proxy Server Address 192.168.1.15
Outbound Proxy Server Port 5060SIP Service Domain
Service Domain 192.168.1.15SIP Source Port
Source Port 5060SIP Authentication
Authentication ID 4003
Authentication Password ••••••••••••••••
DNS
Posts: 3
Participants: 3
@StaceyB wrote:
After upgrading from 10.13.66 I am unable to view log files in the gui. No matter which log file I choose it just shows Loading… in the output. I can view the log files when I ssh to the box. Any ideas?
Posts: 2
Participants: 2
@netphoneusa wrote:
I have call recording set to automatically record my extension. I have noticed that when turned on and I am in a call, I have a few more milliseconds of delay between the end of my sentence and the beginning of the caller’s sentence. The delay is noticeable and causes us to “walk” on each other at times. If I turn call recording off the issue goes away.
Any insight on how to continue to use call recording without that delay would be appreciated.
Posts: 10
Participants: 2
@mcisar wrote:
Have a Ricoh multifunction copier/printer/fax device in the office. Have always had it hooked up for fax with an old ATA and that seems to work fine.
With that said the device is supposed to support SIP directly for the fax and although I’ve screwed around with it from time to time when I’m bored I’ve never been able to get it working.
Does anyone have any experience in setting up a Ricoh for SIP Fax (both from config on the Ricoh side, and any tweaks that might have been necessary on the FreePBX side of the equation)? Unfortunately its not on service contract so Ricoh isn’t really interested in helping out.
Nothing critical, but it would be nice to have it working just for the sake of having it working
Thanks,
Mike
Posts: 1
Participants: 1
@Bidhya wrote:
Hello everyone,
Actually I am working on making IVR using Freepbx. I have made it too. And I want to access the audio files recorded during the phone calls which is stored in /var/spool/asterisk/monitor using C# windows form application. So how can I access the audio files?
Posts: 1
Participants: 1
@Noppes wrote:
Dear Community,
I use FreePBX 14.0.1.20 and I have set in the General SIP Settings the Codec G729 as the first codec, and then uLaw and aLaw and GMS. I can call without problems from SIP phone to SIP phone if both supports the G729 codec. Soon as I call a device that don’t support the G729 codec (it doesn’t matter if I call from G729 phone to non G729 phone or from non to non G729 phone) I get a Declined 603 error soon as I answer the phone. What could be wrong? As I mentioned I have enabled several codecs and I thought if a device don’t support a certain codec they will choose one that both support.
Thanks for your help.
Norbert
Posts: 2
Participants: 2
@sofyane wrote:
Hello,
I have created a Queue for the CRM Operator’s and i forwarded all the call (after checking the time holidays/week end) to this Queue .
the problem is ,i want to give for the callers a ring time max value and when the callers call this number and no one answer the call is forwarded to an announcement .Can you please help me in this matter .
Best regards.
Posts: 6
Participants: 4
@Bowling_For_SIP wrote:
Hi All,
I have an issue that has been plaguing me recently.
I’ll give you the scenario that occurs:
A (e.g. 07XXXXXXXXX) calls in to B (e.g. 1XXX)
B then performs an assisted transfer (via the preset transfer key on their phone) to C (e.g. 2XXX)
A is then transferred through to C
The above works as expected at top level; the call is successfully transferred and A and C are able to talk. The issue arises once the call has finished. When I go to check the CDRs that have been created for the above event, it seems like it does three things:
Create a record within the CDR database for the entire length of the call ((A > B) + (B > C) + (A > C))
Create a record within the CDR database for B > C
Create a record within the CDR database for A > C
It’s fine that it doesn’t create a record for A > B as this isn’t really required. The only problem that I have with the above is that the third record is attributed to the person who was initially called. Here’s an example of some CDRs reflecting the above:
SRC: 07XXXXXXXXX | DST: 1XXX | DURATION: 19 | BILLSEC: 15 |
SRC: 1XXX | DST: 2XXX | DURATION: 4 | BILLSEC: 1 |
SRC: 07XXXXXXXXX | DST: 1XXX | DURATION: 5 | BILLSEC: 5 |
As you can see, the first two records are fine. It’s when it comes to the third one that the problem occurs. It’s still attributing the DST to User A instead of User C whom is the actual DST of the third record. This means that any time that has been spent on the phone by User C talking to User A after they’ve been transferred by User B is attributed to User B.
I’ve also completed a transfer using the Asterisk transfer (*2) and this has the same results as the above. I was concerned that the problem was only occurring due to the way that the handsets were transferring calls when pressing the preset “Transfer” button as I’ve read that they tend to be programmed to perform different actions and send different signals to the SIP server than a regular Asterisk transfer so attempting to do the same but using Asterisk’s built-in transfer shortcode was my way to make sure that this wasn’t the case.
If you have any ideas/thoughts on the above or need me to clarify anything, please let me know.
Cheers in advance!
Posts: 1
Participants: 1
@a5t1 wrote:
Current Asterisk Version: 11.25.3
FreePBX 13.0.192.19PBX Firmware:
6.12.65-20
PBX Service Pack:
1.0.0.0I still see conferences created under “conferences” but when I try to dial one, it connects for 1 second, then the call disconnects. If I use a user pin or admin pin, it has the same behavior, disconnects after 1 second.
I’ve reinstalled the module, can anyone point me to the next troubleshooting step
Posts: 2
Participants: 2
@mvogel4949 wrote:
When I create a new extension I am uncertain what the password is to the UCP. It is not the same as the VM password and it is not the same password that is listed in the user manager section of the extension. I realize I can change it but what exactly is the password generated initially?
Posts: 1
Participants: 1
@mvogel4949 wrote:
I have two phones/endpoints for a single pjsip extension. Is there anyway to distinguish the e911 for each endpoint? THanks
Posts: 2
Participants: 2
@foxfreejack wrote:
Hi all,
First post here after a few months of tinkering on FreePBX. I managed to configure announcements, IVR, queues and misc applications to configure a local number to simulate an incoming call.
The only configuration that I cannot reliably set is the language during a call.
What’s happening is if I choose French language (or let it timeout, which defaults to French), it goes through the language module and forwards to 3 announcements (the first two being optional messages, like service issues and holiday messages), the third is an announcement before routing to a queue. The third announcement is always played in both French and English message, but if I choose English, then only the English message is played.
That said, system built-in messages always do follow set language after the “choose language” IVR.
I’ve tried a few things, like, setting a specific language on the incoming rules instead of "default,” moving the set language to French immediately after the incoming route. Still does the same thing.
Every item in "System Recordings,” except the language prompt for an IVR, does show both languages in the “Supported Language” column. However, when I access any of these recordings it shows a warning message “You have a missing file for this language. Click any red recording above to replace it […]” selecting either language let me play the appropriate language files.
I am not sure why it reacts this way yet.
I am using the Legacy OS version of FreePBX 13.0.192.19 distributed from the FreePBX website.
Thank you very much in advance.
Posts: 1
Participants: 1
