Thank you, we’ll try it.
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Backup and Restore from FreePBX12 to 15
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Responsive Firewall always blocks "good" external users
Right, self signing the module doesn’t work.
So I just hacked validator.php and commented out the sig checking rules, but I can still only load the PHAR directly with PHP, I can’t get fwconsole or the GUI to load the firewall.
What’s so strange is that despite all of the changes I made nothing is effective. I’m quite certain that this is all due to signature issues. The files are definitely changed and I rebuild the PHAR file, and all of the rules are still unchanged.
I see your le cert rules, I’m just doing the same thing but making the rules more sane.
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Responsive Firewall always blocks "good" external users
Keep hacking. Did you try copying over the module sig?
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Override caller ID
In the Follow-me feature, a message to press the #1 key is required to answer the call. If no such message preambled then is not from my server.
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Cant add customer to send traffic/prefix stripping
Hello i would like to do voip call termination and i want to know on my free pbx how to create a route and add customer to send traffic also when sending a call i want to drop customer prefix or number and pick sip number thirdly i want to know about billing how will i bill my customers?
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CLI way of getting bulk handler information
Hello @longqvo,
There is a fwconsole bulkimport option in the fwconsole shell commands. I think it would be a nice feature to add a fwconsole bulkexport optin to it with a slight effort.
This is the Code for the import option:
[root@pbx ~]# cat /var/www/html/admin/modules/bulkhandler/Console/Bulkimport.class.php
<?php
namespace FreePBX\Console\Command;
use Symfony\Component\Console\Command\Command;
use Symfony\Component\Console\Input\InputArgument;
use Symfony\Component\Console\Input\InputInterface;
use Symfony\Component\Console\Input\InputOption;
use Symfony\Component\Console\Output\OutputInterface;
class Bulkimport extends Command {
protected function configure(){
$mod_info = module_getinfo('callaccounting', MODULE_STATUS_ENABLED);
if (!isset($mod_info['callaccounting'])) {
$helptext = 'Import a file: fwconsole bulkimport --type=[extensions|dids] filename.csv --replace(Replace the existing values)';
} else {
$helptext = 'Import a file: fwconsole bulkimport --type=[extensions|dids|callaccounting] filename.csv --replace(Replace the existing values)';
}
$this->setName('bulkimport')
->setAliases(array('bi'))
->setDescription('This command is used to import extensions and dids')
->setDefinition(array(
new InputOption('type', 't', InputOption::VALUE_REQUIRED, 'Type of file'),
new InputArgument('filename', InputArgument::REQUIRED, 'Filename', null),
new InputOption('replace', null, InputOption::VALUE_NONE, 'To replace existing values'),))
->setHelp($helptext);
}
protected function execute(InputInterface $input, OutputInterface $output){
$filename = $input->getArgument('filename');
$type = $input->getOption('type');
if ($input->getOption('replace')) {
$replace = true;
}
if (file_exists($filename)) {
$data = \FreePBX::Bulkhandler()->fileToArray($filename);
} else {
$output->writeln('<error>The specified file does not exist or we cannot read it</error>');
return false;
}
if (!$data) {
$output->writeln('<error>The file provided did not process properly. Check the file formatting</error>');
return false;
}
switch ($type) {
case 'dids':
$output->writeln('Importing bulk dids');
$ret = \FreePBX::Bulkhandler()->import('dids', $data, $replace);
break;
case 'extensions':
$output->writeln('Importing bulk extensions');
$ret = \FreePBX::Bulkhandler()->import('extensions', $data, $replace);
break;
case 'callaccounting':
$output->writeln('Importing CallDECK Ratepatterns');
$ret = \FreePBX::Bulkhandler()->import('callaccounting', $data, $replace);
break;
default:
$output->writeln('<error>You must specify the file type of --type=dids or --type=extensions</error>');
return false;
break;
}
if (!$ret) {
$output->writeln('<error>The import failed</error>');
return false;
} else {
return true;
}
}
}
Do you think it would be needed? It could be even be contributed to the community.
Can you share your thoughts on that?
Thank you,
Daniel Friedman
Trixton LTD.
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Sometimes caller can't hear me
RTP settings in PABX are 30000 - 40000
I have double-checked that UDP ports are all correctly forwarded (see below)
It is difficult to give a percentage… Sometimes got it twice for the same caller, and sometimes no issue for the 10 next incoming calls.
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Critical Information for all DPMA users
I still use 2.9.9 on my D65 phones. But I had to install DPMA 3.6.3 (not 3.6.2), because playing mailbox messages didn’t work anymore. DPMA 3.6.3 is already online, but it seems not in the freePBX repository.
I followed the update instructions for 64-bit systems from here
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Installation
EDIT: I seems 3.6.3 is now also available in the freePBX repositories. So there is no need for the terminal…except if someone doesn’t want to update all packages.
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Sometimes caller can't hear me
Hi !
here is the log for an incoming call one way SIP
https://pastebin.freepbx.org/view/5c61af0e
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Call records rotation
Is it possible to do calls recordings rotation via FreePbx 13 web interface?
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Busy tone, when trying to dial with Cisco 7941 Ip-phone
Hello Everybody. I have some problems with a cisco ip phone. The phone was in a bootlop, and i upgraded the firmware, after that it is working fine. I downloaded a sample .cnf file, and modified the basic things. / server address, sip account, ntp-address / It SEEMS works fine, but i think i made a mistake in the configuration file. The phone was successfully registering on my FreePBX 13.0.197.28 server, and i able to call it from another phone. But when i want to call another extension from the phone, i heard the busy signal after i push ANY number, and on the phone screen a “Reorder” sign appearing. This account was used before another client, and everything was fine with it.
When i push the button on the phone, i can see the following lines in the /var/log/asterisk/full
[2021-04-06 12:17:11] VERBOSE[2202][C-00000004] netsock2.c: == Using SIP RTP TOS bits 184
[2021-04-06 12:17:11] VERBOSE[2202][C-00000004] netsock2.c: == Using SIP RTP CoS mark 5
Has anyone seen anything like this?
Thank you for your help in advance.
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Incoming call from mobile phone dropped on ring group
I don’t understand what is happening here. My gut feeling is that here may be the problem:
AGI Script agi://127.0.0.1/dialparties.agi completed, returning 4
Has anybody an idea what’s causing the return of 4 here?
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Override caller ID
Ahh, you didn’t say you were going to use call confirm. That does make a difference.
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Responsive Firewall always blocks "good" external users
There is a whole development outline and how to properly write, update and do bug fixes of code (and even submitting new modules) right here:
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The #OpenSourceLounge
Next Open Source Lounge is this Thursday, April 8th, at 7PM Eastern https://meet.sangoma.com/open-source-lounge
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Queue Agents Invalid when there is a single agent
Hey everyone,
Our client was complaining that random queue calls end up to the failover destination without ringing in the queue. After looking into this, it turns out, that it appears to happen when there is a single agent in the queue and they are already on a queue call, and there’s another incoming call. While the second call is ringing the agent is being marked as invalid resulting the third call to not be able to join the queue since there are no valid members.
I am able to reproduce this using Asterisk 16.15.1 and FreePBX 14.0.16.4, Queues 14.0.2.34
The queue without any calls:
yplab*CLI> queue show 3233
3233 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0%, SL2:0.0% within 60s
Members:
Desk1 (Local/64066@from-queue/n from hint:64066@ext-local) (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
No Callers
First call ringing:
3233 has 1 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 24s talktime), W:0, C:2, A:1, SL:100.0%, SL2:100.0% within 60s
Members:
Desk1 (Local/64066@from-queue/n from hint:64066@ext-local) (ringinuse enabled) (dynamic) (Ringing) has taken 2 calls (last was 88 secs ago)
Callers:
1. Local/3233@from-internal-000000fa;2 (wait: 0:06, prio: 0)
The call is answered:
yplab*CLI> queue show 3233
3233 has 0 calls (max unlimited) in 'rrmemory' strategy (6s holdtime, 24s talktime), W:0, C:2, A:1, SL:100.0%, SL2:100.0% within 60s
Members:
Desk1 (Local/64066@from-queue/n from hint:64066@ext-local) (ringinuse enabled) (dynamic) (in call) (In use) has taken 2 calls (last was 104 secs ago)
No Callers
Second call ringing: Agent becomes invalid!
3233 has 1 calls (max unlimited) in 'rrmemory' strategy (6s holdtime, 24s talktime), W:0, C:2, A:1, SL:100.0%, SL2:100.0% within 60s
Members:
Desk1 (Local/64066@from-queue/n from hint:64066@ext-local) (ringinuse enabled) (dynamic) (in call) (Invalid) has taken 2 calls (last was 127 secs ago)
Callers:
1. Local/3233@from-internal-000000ff;2 (wait: 0:04, prio: 0)
Resulting the third call not be able to join the queue, I see in the logs:
app_queue.c:8456 queue_exec: Unable to join queue '3233'
It happens with both, chan_sip and chan_pjsip extensions
Relevant queue settings:
Call Confirm = Yes
Restrict Dynamic Agents = No
Agent Restrictions = Call as Dialed
Ring Strategy = rrmemory
Autofill = No
Skip Busy Agents = No
Join Empty = No
Leave Empty = Yes
I am not even sure if that would be an Asterisk or FreePBX issue.
Please let us know if you need any further information.
Thank you
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How to disable "Mark Answered Elsewhere"
Hi,
I have a freepbx 15.0.17.24 setup with 2 IP phones (Grandstream GRP2616) in my office. Both extensions are members of a ring group with “ring all” setting.I want to get a “missed call” when the other extension is picking up an incoming call.
I have tried setting “Mark Answered Elsewhere” to “No”, but its not working. I also tried Queues instead of ring group (again with “Mark Answered Elsewhere” disabled), but its not working either.
I have spend more than 4 days searching all over the forum, but I cant find any solution on this. Does anybody has any idea how to fix this?
Thanks
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Queue Agents Invalid when there is a single agent
This was filed as an issue in the Asterisk issue tracker[1] and a community member submitted a fix for it. The fix is in 16.17.0 and 18.3.0
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Queue Agents Invalid when there is a single agent
Neither of those versions have been published for the Distro yet but we have a ticket on it to get 'er done.
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Queue Agents Invalid when there is a single agent
That’s good news! Thanks to both of you.
@lgaetz I tried doing a quick search on JIRA but I was unable to find that ticket. Perhaps it is an internal one? If it is public and you don’t mind posting the ticket here, I’d appreciate it. I want to follow it so I can update the PBX as soon as it is out.
Thank again
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