Hi, i'm beginner in freepbx,
I have good knowledge about shell and linux configuration,
I have to do an simple call redirection, the call received in 1 fxo port,
have to be redirect in external call number by the second fxo port.
so, i installed freepbx, with an digium card 2 fxo port,
freepbx work fine, and hardware is detected.
Can i ask you where begin for juste make this simple redirection?
thanks a lot
A community support forum for FreePBX users.
Hi, i'm beginner in freepbx,
So, I'm getting an influx of telemarketers calling in. Common callerid that displays starts with 1613 and 1614. If I add 1613* as a blocked number, before it would send all calls with a callerid of that prefix to FreePBX default "this number is disconnected". Now, all those calls with those prefixes get through. If I call that number back I get a operator error (number not in service or something simular). So I think most of these guys are using fake caller ids or reverse caller id, or something. Is there a way to block this stuff?
I have become frustrated in trying to get the trunk sequence for matched routes to failover when the first trunk in the list is busy.
The system in question is running Freepbx v220.127.116.11 and Asterisk 1.87. In this particular instance, the first outbound trunk is an analog POTS circuit terminated on a DAHDI card, and the second outbound trunk is a SIP trunk. Local/toll free calls ONLY are routed to the DAHDI card, and my intention is to have all long distance calls routed out the SIP trunk, as well as local calls, but only if the DAHDI circuit is busy.
Ok, I am brand new to the concept of doing my own PBX.
First: What are the system/hardware requirements for a PC running the freepbx distro?
What services do I need to buy? I know I need a DID, and trunk? I see the ad for unlimted for 24.99 but wondering if there is something cheaper out there. (I may not need unlimited).
I'm looking for an device wich makes home automation integrated with a pbx ip or legacy pbx. An extension wich i call and with dtmf i can control lamps, doors, etc.
Any about that?
thanks in advance
I have the following problem, dual registration:
-- Registered SIP '220' at 18.104.22.168 port 18862 expires 3600
-- Saved useragent "SIPAUA/0.1.001" for peer 219
-- Registered SIP '220' at 22.214.171.124 port 5060 expires 120
-- Saved useragent "MxSipApp/126.96.36.199 " for peer 219
Any tips to restrict or prevent? as per extension "useragent" restriction using freepbx or .conf files modification?
Any Warning Script?
Anything can help me?
Thanks in advance.
I am using FreePBX 188.8.131.52
I changed a directory for call records in the variable MIXMON_DIR
via FreePBX Tools -> Advanced Settings -> Override Call Recording Location
but queues turn continue to be written in the directory /var/spool/asterisk/monitor
how I can change the default directory of call records for queues?
thanks for any ideas!
There is No "Fax Handling" section under "Inbound routing".
Fax Detect: yes
Detect Faxes: NoYes
Fax Detection type:
Fax Detection Time:
Which module to install to see "Fax Handling" section?
This is my first post. I am trying freepbx now - moved away from Trixbox as I want to have freepbx's flexibility.
I ran in to the old All circuits are busy now" default message and is seems that installing the Module - Route Congestion Messages does not help me a bit.
I configured it to have HangupCause 1,18,19 and 28 to play a different recorded message other than the default, tested the recording via the Announcement feature so i know the recording works (I recorded using freepbx)
But I still get the same "All circuits are busy now...." message.
I am very new to working with the freepbx at our church, and was not the one to set it up so please bare with me.