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[Solved] Reload failed because retrieve_conf encountered an error: 1 after upgrade from 13 to 14

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I have the same issue.

My temp solution: fwconsole ma uninstall ucp certman


No dial tone on FXS port

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Sometimes, the configuration gets borked and recreating it solves the problems, especially with DAHDI.

[Solved] Reload failed because retrieve_conf encountered an error: 1 after upgrade from 13 to 14

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Seems like the upgrade to php5.6 via ppa messed up my php installation as now some packages have to be php-xxx, some are php5-xxx and some are php5.6-xxx. Another issue on the dashboard which appeared today guided me into the right direction:

Cronmanager encountered 1 Errors
The following commands failed with the listed error
/var/lib/asterisk/bin/module_admin listonline > /dev/null 2>&1 (1)

So i executed:

root@asterisk:~$ sudo -u asterisk /var/lib/asterisk/bin/module_admin listonlin
[FATAL] PEAR must be installed (requires Console/Getopt.php). Include path: .:/usr/share/php

root@asterisk:~$ sudo pear install db-1.7.14
sudo: pear: command not found

root@asterisk:~$ apt install php-pear
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following NEW packages will be installed:
  php-pear
0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded.
Need to get 0 B/285 kB of archives.
After this operation, 2,186 kB of additional disk space will be used.
Selecting previously unselected package php-pear.
(Reading database ... 92702 files and directories currently installed.)
Preparing to unpack .../php-pear_1%3a1.10.5+submodules+notgz-1+ubuntu14.04.1+deb.sury.org+1_all.deb ...
Unpacking php-pear (1:1.10.5+submodules+notgz-1+ubuntu14.04.1+deb.sury.org+1) ...
Processing triggers for man-db (2.6.7.1-1ubuntu1) ...
Setting up php-pear (1:1.10.5+submodules+notgz-1+ubuntu14.04.1+deb.sury.org+1) ...

root@asterisk:~$ sudo pear channel-update pear.php.net
Updating channel "pear.php.net"
Update of Channel "pear.php.net" succeeded

root@asterisk:~$ pear install db-1.7.14
WARNING: "pear/DB" is deprecated in favor of "pear/MDB2"
downloading DB-1.7.14.tgz ...
Starting to download DB-1.7.14.tgz (134,864 bytes)
.............................done: 134,864 bytes
install ok: channel://pear.php.net/DB-1.7.14

and finally the "Apply config" from gui succeded. Is it possible to check for the existence of pear, php-prear or the pear/DB and have a more explicit error message in that case?

I added [Solved] to the title.

Call Recording Issues Freepbx 13

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Has anyone else had issues with call recordings on Freepbx 13? We have just upgraded two pbx's and there seems to be issues with calls being recorded. We have the queues setup to record also but there is no audio file present. I have set the recording location with read and write for everyone, no files. Same in /var/spool/asterisk/monitor folder. The recordings just seem to be random. For the calls that do get recorded, there is no link in the CDR Reports or in the UCP.

No dial tone on FXS port

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So i was just messing about with the ports and found that if i call the extension (3000) and then plug the phone into the port it will actually answer the call and function, so to me this suggests that there is an issue with the signals between the phone and port?

No dial tone on FXS port

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That is clearly not how it is supposed to work

No dial tone on FXS port

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Well obviously not, but it shows that it is capable of working. So it looks like the reason it isnt working is to do with signalling/voltage changes as when i attempt to dial in/out normally it clearly isnt sending/receiving the correct signal

2 Calls being created when calling a number

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This is a bit of an odd issue and it isn't happening all the time.

When someone makes a call from an extension, can be either internal or external, sometimes 2 calls to the same number are initiated.

We are using Yealink T23G Handsets.

Has anyone else noticed behaviour like this before?

Current Asterisk Version: 13.16.0

Thanks


Direct calling extension from external, forwarded to another system

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This topic comes up quite a bit:

Strange Caller Dialed Number Behavior

2 Calls being created when calling a number

Call Recording Issues Freepbx 13

How to create the specific dial pattern in the free PBX outbound routes GUI?

Hyper-V Integrated Services (VSS deamon)

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Why would we install something that very few people use. Your missing the point here. It's in the repo doe you to install. Why would we install it on every system when less then 1% use hyper V with FreePBX.

Direct calling extension from external, forwarded to another system

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Thank you for you answer, but not quite. Third topic is very simmilar, but I need something different (or I'm not undestanding it correctly :wink: )

I don't need to call an IVR on 3cx, because I don't have any on it. Routing this calls to sip trunk directly will not allow me to call exact extension present on 3cx, but only default number connected with this trunk on 3cx side (or maybe I'm wrong, and it works in different way?).

I need to do something, that would make direct dialling from freepbx IVR (during automatic message) to extension present in 3cx working, Now, when I dial 161 number during IVR recording when calling from mobile (for example), I can hear the sound of dialling, but nothing happens on handset, and freepbx is taking me back to IVR recording again. Logs are claiming that 161 number is busy. But, when I answer the call on "after IVR default handset", and then manually redirect it to 161, it will work just fine, I can talk with this extension from mobile without any problems.

My english is not perfect, so please excuse me if I said something wrong, or ununderstable :slight_smile:


Why register multiple lines on same phone to same extension?

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Yes, you can register multi-times. change the " Max Contacts " in ext > advance ... Why multiple on one phone..well; any incoming call will ring the not-in-use phone/line .You do get the CID of the next caller. You hang up the first and grab the 2nd if it's an important call, else you let it go to voice mail.

iSymphony Stuck Calls

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So I have seen this scenario when using a very specific call flow:

  • A user has a live call and transfers the call to a queue using attended transfer
  • call is received and answered by queue
  • user completes the attended transfer
  • call is answered by queue agent

Essentially the Caller ID of the call changes while it is in the queue which confuses iSymphony. i9 is aware of this issue, the work around is to use blind transfer not attended.

How to configure cisco 8961 with sip firmware 8961.9-3-2-10

How to divide internal and external incoming calls

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Go read the reply I left in ....Why register multiple lines on same phone to same extension?

How to configure cisco 8961 with sip firmware 8961.9-3-2-10

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If it cannot register, the most probable cause is an error on the XML file, you will have to check it and find the error.

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