I will try this
Thanks
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Repeat IVR recording
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Voicemail Greeting Upload in UCP not working
Sounds like one of the intermediate directories in the voicemail hierarchy is missing.
The first thing I'd try is setting up a VM greeting using the "old" tools. This creates all of the directories "down to" the one for your extension (when it installs the first one).
I don't have any custom VM greetings in my system, or I'd just look up the hierarchy and post the "mkdir -p" command to build it out.
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Update modules, HD full, system down. :(
That sounds like a good feature request (or even a bug-report, since the process failed for your installation). Use the Issues tab to request it - be sure to check if someone hasn't already beat you to is (Andrew is pretty good about doing this for us "mere mortals",
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Time Condition BLF Hint not updating until call received
The issue occurred again this morning and the user reporting it said it happens every day. I'm not sure that's true, but possibly every day recently based on what I am seeing in the logs. If I run through 'grep schedtc /var/log/cron*' the last time I see it run is on Aug 9, which is the same as it was from my post on Aug 11. It looks like once the issue starts, it won't resolve until I manually run the command.
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Fresh SNG7 install, DAHDI fails to load
Do you have DAHDI hardware? It looks like your DAHDI software setup is going, but the hardware doesn't seem to be responding.
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Auto dialing a number
Yeah. Check out "callfiles" for a lot more information.
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New FreePBX Build - No registrations?
Thanks, any idea what 2 unknown refers to?
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Assistance request - Possible virus on PBX
Allright; everyone here seems to confirm that the issue is "fortigate" related.
One of the thing that I find interesting is that the rule relating to the port 5060 allows connection from the IP range of my SIP provider only.
I'm going to assume that basically, even that rule is not set up properly.
Keep in mind that as per the firewall expert, the requests would originate FROM my PBX, as per the logs.
I'll let you know what the outcome is. Thank you everyone that provided an input.
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FreePBX / PBXACt for a new Hotel
Cool.
Don't - add it to their bill. Bill out each room as if they were going to be a 24x7 call source (I usually charge about 0.4 cents per minute) and bill them by the day. That works out to about a $5 "upcharge" for "unlimited domestic calling from your room." If that seems too much, don't charge that much.
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I can not use Voicemail - Error
Sounds like a bug - the Issues tab is your friend. Someone else today is having a related problem, perhaps you could work together.
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Time Condition BLF Hint not updating until call received
The questions I'd like to see answered are:
1) Is crond failing? If the cron program is stopping, your cron jobs will not run.
2) check the existence and permissions on the file /var/www/html/admin/modules/dashboard/scheduler.php to make sure it is executable and exists. If it's executable, it probably exists, so ....
As a "bandaid" fix, I'd start watching crond to make sure it is running all the time.
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Assistance request - Possible virus on PBX
To be clear:
If a connection is initiated at your PBX (or otherwise inside your network) and it opens a dialogue with the attacker, the firewall will honor that request. The settings would have to be pretty exact for that to work. If something in your network is opening the port, then he could be correct. Having said that, though, that's a pretty far-fetched scenario. It would require some very specific knowledge of your network and PBX setup.
The typical vector for these types of attacks is through an open port 5060. That's why we recommended changing your "incoming call" port to something outside the norm. Note that the 5060 port doesn't have to change in the PBX - open a port on the Fortigate and redirect it to the PBX 5060 port.
If you don't have external phones connecting to the server (all of your extensions are behind the Fortigate) there's no reason to allow incoming (or even outgoing) connections that reference your PBX port 5060, especially if you change the port.
Note that there are other attack that can be initiated that would open a dialog this way, but the FreePBX system is usually very good about blocking those attacks.
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Fresh SNG7 install, DAHDI fails to load
Unfortunately that is not the issue. The error about key not available is generaly related to compilation issues due to kernel mismatch, which is very strange because I didn't compile DAHDI by myself, it is the version installed by the SNG7 installer.
In any case, DAHDI should load even without any cards installed.
In fact, with FreePBX Distro 10.13.66.18 on the same server, without any cards installed, DAHDI loads with no errors.
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Auto dialing a number
Ok so here is what i ended up with.....
This is a script that I can call from a browser on the local network.
Obv ext and number have been changed so have usernames and passwords
<?php
$extension = $_REQUEST['exten'];
$dialphonenumber = $_REQUEST['number'];
$timeout = 10;
$asterisk_ip = "127.0.0.1";
$socket = fsockopen($asterisk_ip,"5038", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: abc123\n");
fputs($socket, "Secret: def456\n");
$wrets=fgets($socket,128);
echo "Response 1: $wrets
";
fputs($socket, "Action: Originate\r\n" );
fputs($socket, "Events: off\r\n" );
fputs($socket, "Channel: SIP/$extension\r\n" );
fputs($socket, "Exten: $dialphonenumber\r\n" );
fputs($socket, "Context: from-internal\r\n" ); // very important to change to your outbound context
fputs($socket, "Priority: 1\r\n" );
fputs($socket, "Async: yes\r\n\r\n" );
fputs($socket, "Action: Logoff\r\n\r\n" );
$wrets=fgets($socket,128);
echo "Response 2: $wrets";
fclose($socket);
?>
In the manager_custom.conf I have the following added lines:
[abc123]
secret=def456
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
The script returns...
Response 1: Asterisk Call Manager/1.3
Response 2: Response: Success
Which looks like it does as expected, but the logs throw back
[2017-08-16 13:14:20] ERROR[8585] utils.c: fwrite() returned error: Broken pipe
Some other suggestions on the t'internet said it could be a copy and paste issue from windows (like new lines), so I re-wrote it all by hand.
Still returns the error
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Can I change the rtp ports and reduce the number of open ports?
Hi Guys can I reduce the number of rtp ports in freepbx because 10000-20000 is a very large number How ever only 5-6 extensions are using nat/external sip calls .
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IVR. Directdial to other PBX that is connected via IAX
I have 3 PBX's connected via IAX(trough VPN).
Site-1 extensions 11xx. site-2 extensions 12xx. site-3 extensions 13xx
extension-to-extension dialing between the 3 pbx's works fine.
All have separate incoming PSTN lines and have IVR on these incoming lines.
IVR setup has "enable direct dial" ENABLED.
External callers can dial anyone's entension in the PBX they called into. But they cannot dial an extension on another PBX.(the IVR responds: We have not received a valid response)
How do I get this to work?
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Auto dialing a number
In your first echo, get rid of the embedded CR and use a \n.
I don't remember if the username and password need \r and \n, but try that and see if you are getting further. Also, try running the program from the console and see what happens. You're not really getting a lot of debug information from Asterisk.
Also, double check to make sure where your manager logins are coming from. I think there's an Asterisk CLI command that lists the managers - make sure 'abc123' is in there.
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Can I change the rtp ports and reduce the number of open ports?
We do that to reduce the likelihood of attackers being able to snag your RTP sessions. It's an "on purpose" thing.
You can, of course, reduce the RTP port range - if you decide to, don't go more than 4:1 (10 extensions, 40 ports).
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IVR. Directdial to other PBX that is connected via IAX
One way: Virtual extensions that "mimic" the extensions you want to dial (add dial strings to the other servers).
Another way is to look at the context that the IVR lives in and add information in that context that makes the 11xx and 12xx extensions visible.
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Send Volumes
Maybe go back to forcing updates but with better quality assurance?
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