no PW work after installation, system screen click to reboot, loads to cli, login prompt, NO PW will work I have re-installed xxx times still nothing pulling my hair out!
14 posts - 4 participants
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New install NO passwords work
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Stuck Self Signed Certificate
I am having an issue where the self signed continues to be presented even though I have loaded my own certificate, set it to default and even deleted the old self signed certificate. I checked the files in /etc/asterisk/keys/integration/ and they are updated with my certificate.
4 posts - 3 participants
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Link between two Freepbx work nice, but there is issue when call an extension not found in the same range
I linked between two FreePBX.
calls between this work nice.
but when call a number in the range of extensions but extension not found in another PBX the call go to inbound route.
For example
PBX_A pattern is 1xx and have extension 100 to 110
PBX_B pattern is 2xx and have extension from 200 to 210
When ringing on 211 from PBX_A the call go to inbound Route in PBX_B.
I want to return status code from PBX_B extension not found.
Is that option available on FreePBX?
3 posts - 3 participants
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Contacts problem with Cisco 7841 and freepbx
Hello
I have this probleme when i try to acces the contacts page on my ipphone
and this is my SEP(MAC).conf.xml file
<device>
<fullConfig>true</fullConfig>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>British Summer Time</timeZone>
<ntps>
<ntp>
<name>192.168.0.254</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<tftpDefault>true</tftpDefault>
<members>
<member priority="0">
<callManager>
<name>192.168.0.199</name>
<description>Station</description>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<processNodeName>192.168.0.199</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>1</callLogBlfEnabled>
</commonProfile>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>1</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>10:30</displayOnTime>
<displayOnDuration>06:05</displayOnDuration>
<displayIdleTimeout>00:05</displayIdleTimeout>
<webAccess>1</webAccess>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<deviceSecurityMode>1</deviceSecurityMode>
<idleTimeout>0</idleTimeout>
<directoryURL>http://192.168.0.199/directory.xml</directoryURL>
<servicesURL>http://192.168.0.199/directory.xml</servicesURL>
<idleURL></idleURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort></backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy></registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x--serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>true</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<startMediaPort>10100</startMediaPort>
<stopMediaPort>10300</stopMediaPort>
<voipControlPort>5060</voipControlPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<phoneLabel>Cisc7841</phoneLabel>
<natReceivedProcessing>false</natReceivedProcessing>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>1004</featureLabel>
<proxy>192.168.0.199</proxy>
<port>5060</port>
<name>1004</name>
<displayName>GRace OKK</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>1004</authName>
<authPassword>btfood123</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
<messagesNumber>*98</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>1004</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID></featureID>
<featureLabel></featureLabel>
<speedDialNumber></speedDialNumber>
</line>
</sipLines>
</sipProfile>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Phonebook</name>
<url></url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>ccf</name>
<url></url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
</device>
5 posts - 2 participants
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Forwarded Google Voice Number Will Not Answer
I have a Google Voice Number that I was to forward to my FreePBX. I have linked the number in Google Voice and it forward the call. However in bypassed the Inbound route and somehow goes directly to Ring Group 200.
Normally My calls come to an IVR that plays a please hold message and then will Ring Ring Group 200. I will play my music on hold while it rings.
However When Google call forward to it, I never see the IVR in the call logs. I see 4 CDR logs for each call and the Application it Dial and it goes to 200. I then rings hunt group 200 3 times with no music on hold and then Google takes a message
CDR Records - The one on the bottom is direct dial to SIP trunk and it worked. The top 4 are from 1 call forwarded call by google.
3 posts - 3 participants
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Sangoma Realtime API is not installed/enabled
Hello guys, I recentry upgraded from FPBX16 to FPXB17 and everything seems to be working fine, but when I access the dashboard it keeps saying that sangomartapi is not installer/enabled when it actually it is, if I do fwconsole ma list I get :
sangomaconnect | 17.0.1.52 | Enabled | Commercial | Sangoma
but on dashboard it says :
I can’t find any relevant error on the logs, I already tried reinstalling this module, but the problem remains; This is not a big deal but it’s annoying because I would rather see everything green on the dashboard. I also noticed that no matter what it says the last update was 2 days, 15 hours, 11 minutes, 34 seconds, ago which is when I first installed the module, I tried other browsers and incognito mode with the same result.
I’m trying to figure out where this error comes from so I could perhaps patch the code, but I realized if I do that I get another error saying about tampered file; I also looked for a way to just override this message but so far this is proving to be very difficult, so I’m wondering if anyone here could shed a light would be greatly appreciated.
Thank you.
7 posts - 3 participants
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FreePBX17 on Debian Call Recordings To/from Trunks Not Working
Server details:
FreePBX 17.0.19.23
Debian GNU/Linux 12 (bookworm)
I am have having an issue with call recordings where calls originate from inbound routes or originate from an extension going externally through an outbound route. Internal calls extension to extension are recording and can be played back using CEL.
Looking at the files in var/spool/asterisk/monitor all file permissions are set to 644. The files using inbound/outbound routes are listed in CEL but have a file size of 1KB - nothing recording.
I have set the inbound/outbound routes call recording to Force as well as Settings → Advanced settings → Call Record Option to Yes.
The system logs do show that MixMonitor creates a file and starts recording and closes filestream then ends recording as well.
Appreciate any advise where to look or configuration settings that I may have missed.
Cheers.
6 posts - 3 participants
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GraphQL API for CDR not working
Hi,
I am trying to fetch CDR Reports from APIs. I am interested in fetching recordings of calls.
I tried the following query from the Explorer:
{
fetchAllCdr(first: 10, startDate: “2022-1-1”, endDate: “2026-1-1”{
edges {
node {
id
timestamp
}
}
}
}
but this yields null as edges - and it’s wrong because i have many calls.
Removing the startDate/endDate leads to Internal Server Error.
Can someone help me?
EDIT: by including “message” in the query, I was able to understand that I was using the wrong format.
Now, edges is anyway null, and cdr fields reports only few calls.
4 posts - 3 participants
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Apply changes error message - [Whoops\Exception\ErrorException (2)]
Hi Everyone,
1st time seeing bellow message, Is someone seen this message before ? and any idea how i can fix it?
All module are up to date, No any disabled module.
FreePBX-17 system.
fwconsole r --verbose
Reload Started
In Tts.class.php line 129:
[Whoops\Exception\ErrorException (2)]
Trying to access array offset on value of type bool
Exception trace:
at /var/www/html/admin/modules/tts/Tts.class.php:129
Whoops\Run->handleError() at /var/www/html/admin/modules/tts/Tts.class.php:129
FreePBX\modules\Tts->doDialplanHook() at /var/www/html/admin/libraries/BMO/DialplanHooks.class.php:109
FreePBX\DialplanHooks->processHooks() at /var/www/html/admin/libraries/Console/Reload.class.php:323
FreePBX\Console\Command\Reload->reload() at /var/www/html/admin/libraries/Console/Reload.class.php:95
FreePBX\Console\Command\Reload->execute() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Command/Command.php:312
Symfony\Component\Console\Command\Command->run() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:1022
Symfony\Component\Console\Application->doRunCommand() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:314
Symfony\Component\Console\Application->doRun() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:168
Symfony\Component\Console\Application->run() at /var/lib/asterisk/bin/fwconsole:163
reload [--json] [--dry-run] [--skip-registry-checks] [--dont-reload-asterisk]`3 posts - 1 participant
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Editing existing modules
I’m running freepbx 17.0.19.23 and in my logs i get alerts about the modules api, voicemail and backup using a deprecated way of adding console commands. I figured out it was because of the lazy commands, and editing a few lines in the modules fixes this. The problem is that now i have security alert about some files being tampered, the ones i just edited. Is there a way to edit those modules without triggering the alerts?
2 posts - 2 participants
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Let's Encrypt renewal not working on FreePBX 17
My v16 systems are still fine, but the systems I’ve migrated to 17 are unable to renew. I find it interesting that it was originally able to acquire the certificate with the firewall turned on, but now unless the firewall is disabled the renewal fails.
Has anyone else had a similar problem on v17, and/or has anyone found a resolution?
4 posts - 3 participants
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FreePBX 17 TLS OK, but no SRTP
I am not sure what is causing this issue
Certificates installed,
HTTPS, working
SIP trunk connects
Extention connects
but with SRTP enabled on both client and server i get the following error
[2025-01-17 17:32:58] VERBOSE[680969] netsock2.c: Using SIP RTP Audio TOS bits 184
[2025-01-17 17:32:58] VERBOSE[680969] netsock2.c: Using SIP RTP Audio CoS mark 5
[2025-01-17 17:32:58] ERROR[680969] res_pjsip_session.c: 100: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
am i missing something? I had this working on FreePBX 16 no issue. the settings seem the same but its not working.
5 posts - 2 participants
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Unable to disable firewall [forked and retitled by mod]
fwconsole moduleadmin disable firewall
The following error(s) occured:
- Cannot disable: The following modules depend on this one: sysadmin
1 post - 1 participant
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FTP from FPBX17 (Debian 12 Bookworm) to Windows Server 2019
We have spun up a new FPBX17 server on a Dell ESXi server. We are currently working on setting up our FTP backup to a file share server (also on an ESXI server, on the same VLAN too). In FPBX, FileStore and the backup and restore module are configured correctly. On the Windows server, we do not have IIS blocking any IP addresses. We have also confirmed that our FTP service user account is working and that we have the correct credentials in our FPBX system. I followed the FreePBX deployment process (onto Debian 12) by using the instructions here: https://sangomakb.atlassian.net/wiki/spaces/FP/pages/230326391/FreePBX+17+Installation
Is it possible I am missing an FTP component on the Debian server? The Windows server is configured correctly, as we still use FTP to back up our old FPBX15 server.
Help a noob out, plz.
Thoughts or opinions are greatly appreciated. If more information is needed please feel free to ask. Thanks!
1 post - 1 participant
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Queue module
What can you say about the date for the QUEUE module, which is January 30, 2025, when the basic QUEUE module or the advanced PRO functions actually expire? Can anyone provide more information?
2 posts - 2 participants
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Got locked out on setup
Installation script completed successfully.
I went to Web GUI Wizard. After answering “Yes” to “Automatically configure Asterisk IP Settings?” I lost connection to GUI, and also unable to connect via SSH.
What do I do now?
3 posts - 2 participants
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Softphones: Echo Problems / Recommended Softphones
Hello community,
we run a FreePBX with appr. 100 extensions. Most extensions connect via Grandstream-Devices to the FreePBX. Additionally we have a few Fritz!Boxes and 2 softphone-apps.
The 2 softphones (different apps, using udp for on-prem-communication) produce a lot of echo. One started to echo after upgrading from V15 to V17. The other used to echo with V15. We tried different headsets. An OS-Upgrade from Win 10 to Win 11 didn’t change anything.
Going through the FreePBX-extension-settings I did not find any value for e.g. “echo reduction”.
Is there a way to fight echos in FreePBX?
Why would there be an echo after upgrading from V15 to V17?
Is there a list of recommended softphones (Linux and Window-Clients) for FreePBX?
Thanks a lot
Martin
2 posts - 2 participants
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Whow to set caller ID into Contact Header to YES
Hi, everyone,
I’m encountering an issue and hope someone can clarify this for me.
In my provider’s documentation (which is designed for FreePBX 16.0.26 with Asterisk 18.13.0), they mention enabling “Caller ID into Contact Header” in the following path:
Settings > Asterisk SIP Settings > SIP Settings [chan_pjsip] > Caller ID into Contact Header.
However, in my current setup (FreePBX 16.0.40.11 with Asterisk 20.4.0), this option seems to be missing from the interface. Here’s a screenshot from my system for reference:
Here’s also an extract from my provider’s documentation showing the recommended setting:
I found a related forum post discussing Contact Header manipulation (contact-header-manipulation) by @joni1802, but it send me to the same place as my provider documentation.
My Questions:
- Has this option been deprecated or moved in newer versions of FreePBX or Asterisk?
- Is there a way to manually enable this feature, perhaps through the config files or another method?
Any information or clarification would be greatly appreciated!
1 post - 1 participant
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BLF on Cisco SPA508G
I know there are several posts on making BLFs work on the Cisco phones. I am just getting back into Asterisk and FreePBX (after many years of not working with it). I have the latest version installed and working. I have been trying to get a couple of Cisco phones (for testing) working, and they are connecting and registering on the PBX just fine. However I (like many others) have not been able to figure out how to make the outer line buttons on a 508G work like a BLF.
Which brings me to my question - the other posts about this subject make references to setting the Server Type under Attendant Console settings to Asterisk (which I did). My basic question is can you USE the remaining 7 line buttons on this phone for BLF to display what is going on with other extensions on the system, or are these Attendant Console settings ONLY for the use of a side car that has to be attached to this phone?
1 post - 1 participant
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Voicemail folders won't load in UCP
We recently updated the system and now we get an error when loading folders in the voicemail section of UCP. An error references transcriptionURL.
3 posts - 2 participants
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