@voipsecurity wrote:

Hey guys,

We are using FreePBX 13 with Asterisk 13.21 certified and started getting a lot of call quality issues. We are doing a 3-way conference using AMI originate on asterisk, everything works great but sometimes when we have a good bit of calls, audio getts so choppy and we believe asterisk or our OS starts dropping UDP packets.

We are using the ULAW period for all trunks and calls. And one thing we noticed that recently we started getting this error message on asterisk console ->
chan_sip.c:23007 func_header_read: This function can only be used on SIP channels.

And every time we get this message, our network traffic on asterisk spikes up for a few seconds. We are just using chan_sip not pjsip on this server.

Please help us to identify the cause of this issue. Any suggestions will be highly appreciated.

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@briang70 wrote:

Hello,

So i have been doing some test calls in my new installation and for some reason DTMF is not detected by my PBXact in most of the calls but the strange thing is that it does detects DTMF in some random calls. When i reach my IVR and i dial my Option is does work in very few occasions.

On this log you can see that DTMF is being detected:

DTMF.jpg1108×204 102 KB

But in most of the calls the logs doesn’t show anything for DTMF.

Any idea whats happening here?

Thanks you All!

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@jcadman wrote:

I have an app_rtsp.c file i would like to install to asterisk so that I can monitor my IP Cameras through my SIP Phones. What are the steps to install a .c file on freepbx/asterisk?

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@pierredh wrote:

Hello, I try to call my freepbx with a sip url (ex: xxx@sip.12voip.com) and i don’t underdstand this error

[2020-05-04 14:03:06] NOTICE[24454][C-00000072] chan_sip.c: Call from ‘2126’ (85.201.XX.xx:64637) to extension ’ xxx@sip.12voip.com’ rejected because extension not found in context ‘from-internal’.

how to solve it ?

And in the diap plan, how can i do to say if the sip is contain @sip.12voip.com use this trunk ?

Thanks a lot

Pierre

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@microchipmatt wrote:

Hello, all, I would have tagged this off if my old post, but it is now closed. We’ve been converting all of our sites/phones to VoIP and freepbx, and that has been great. However our security panels have been left our of the equation in this conversion. I bought some HT801’s & HT802’s for experimentation. I went to put one in place, and before I did, the current, Alarm “Secuirty” guy said that they wouldn’t work, because the system he put in needs to sense, “Low Line Voltage”…I thought these HT801’s and HT802 would do that as well…Does anyone know if they do? Any help would be greatly appreciated.

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@PitzKey wrote:

Hi guys,

Now with more office users working from home, we got a complain from a user who does not have FollowMe call confirmation enabled on their extension: That on some calls the PBX does prompt to press 1, however, when they press 1, the “incoming-call-no-longer-avail” greeting plays as soon as they press 1.

Looking through the logs I see that this behavior only happens on calls that was transferred from another follow me user who does have call confirmation enabled.

Example:
Ext 101 FollowMe w/Confirmation
Ext 102 FollowMe w/o Confirmation

101 answers a call on their follow me number, they perform a DTMF blind transfer to 102, then is when the issue happens.

I tested this on two different PBXs, 1 is Ver13 Asterisk 13, the other is Ver14 Asterisk 16.

Is anyone else seeing the same behavior?

Thanks

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@jsmith3899 wrote:

Hello I am trying to register voiper on my cell phone to my freepbx. I can register all local phones just fine what all is entailed in registering a client from public IP can I get a cheklist to go throgh? Do I have to use STUN server and if so how do I use it? Please help getting 408 time out error message.

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@aboggs wrote:

This happened Friday after upgrading from version 13 to 14.0.13.28.

PBX will automatically transfer our calls to the answering service at 5 pm.

Now when the calls are transferred there is no audio either direction. The call rings the answering service. They answer but the caller cannot hear them and they cannot hear the caller.

Nothing has changed with our SIP service, router, or phone system otherwise.

Any suggestions would be appreciated,
-Adam

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@jmfield wrote:

So we have been testing this for weeks. We thought we had it solved but at this point are back to square one.

ringall - works with extension and having it CF to users cell.
Once we add another extension below that or mix an external number with # at the end it seems to break the entire group

hunt and memoryhunt don’t work at all like they are supposed to. We have messed with ring times from 5 seconds to 300 seconds. No matter what we have as ring times in this group it seems to say all users are busy around a 25 second mark. Almost like something somewhere else is overriding what we have in the ring group ring time setting.

Ideally we’d like to have
cellnumber#
ext A
ext B
ext C

ring cell 1st, then cell+A, cell+A+B+C but so far and weeks of testing this just never works. We were told to tray asterisk build 16 but it so far has not made a difference.

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@rsarceno wrote:

I using FreePBX 13.0.197.22
I’m only using 27% disk space
Before I reboot the server I run fwconsole stop.
The server reboot with the following error

Exception: SQLSTATE[HY000] [2002] Can’t connect to local MySQL server through socket ‘/var/lib/mysql/mysql.sock’ (111)::SQLSTATE[HY000] [2002] Can’t connect to local MySQL server through socket ‘/var/lib/mysql/mysql.sock’ (111) in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:

1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:134
3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:131
4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:131
5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:72
6. FreePBX->__construct() /var/www/html/admin/bootstrap.php:145
7. require_once() /etc/freepbx.conf:9
8. include_once() /var/lib/asterisk/bin/fwconsole:12

I tried service mysqld start
MySQL Daemon failed to start.
Starting mysqld: [FAILED]

Content of my Mysql.log
190207 23:35:01 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql
190207 23:35:22 InnoDB: Initializing buffer pool, size = 8.0M
190207 23:35:22 InnoDB: Completed initialization of buffer pool
InnoDB: The log sequence number in ibdata files does not match
InnoDB: the log sequence number in the ib_logfiles!
190207 23:35:23 InnoDB: Database was not shut down normally!
InnoDB: Starting crash recovery.
InnoDB: Reading tablespace information from the .ibd files…
InnoDB: Restoring possible half-written data pages from the doublewrite
InnoDB: buffer…
190207 23:35:30 InnoDB: Started; log sequence number 0 8772053
190207 23:35:34 [Note] Event Scheduler: Loaded 0 events
190207 23:35:34 [Note] /usr/libexec/mysqld: ready for connections.
Version: ‘5.1.73’ socket: ‘/var/lib/mysql/mysql.sock’ port: 3306 Source distribution
190504 7:27:25 [ERROR] /usr/libexec/mysqld: Got error 134 from storage engine
190504 7:27:25 [ERROR] /usr/libexec/mysqld: Sort aborted
200504 20:38:50 [Note] /usr/libexec/mysqld: Normal shutdown

200504 20:38:50 [Note] Event Scheduler: Purging the queue. 0 events
200504 20:38:50 InnoDB: Starting shutdown…
200504 20:39:21 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql
200504 20:39:21 InnoDB: Initializing buffer pool, size = 8.0M
200504 20:39:21 InnoDB: Completed initialization of buffer pool
200504 20:39:22 InnoDB: Started; log sequence number 0 9459016
200504 20:39:22 [ERROR] /usr/libexec/mysqld: Can’t create/write to file ‘/var/run/mysqld/mysqld.pid’ (Errcode: 2)
200504 20:39:22 [ERROR] Can’t start server: can’t create PID file: No such file or directory
200504 20:39:22 mysqld_safe mysqld from pid file /var/run/mysqld/mysqld.pid ended
200504 20:43:21 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql
200504 20:43:22 InnoDB: Initializing buffer pool, size = 8.0M
200504 20:43:22 InnoDB: Completed initialization of buffer pool
200504 20:43:22 InnoDB: Started; log sequence number 0 9459016
200504 20:43:22 [ERROR] /usr/libexec/mysqld: Can’t create/write to file ‘/var/run/mysqld/mysqld.pid’ (Errcode: 2)
200504 20:43:22 [ERROR] Can’t start server: can’t create PID file: No such file or directory
200504 20:43:22 mysqld_safe mysqld from pid file /var/run/mysqld/mysqld.pid ended
200504 21:26:05 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql
200504 21:26:06 InnoDB: Initializing buffer pool, size = 8.0M
200504 21:26:06 InnoDB: Completed initialization of buffer pool
200504 21:26:06 InnoDB: Started; log sequence number 0 9459016
200504 21:26:06 [ERROR] /usr/libexec/mysqld: Can’t create/write to file ‘/var/run/mysqld/mysqld.pid’ (Errcode: 2)
200504 21:26:06 [ERROR] Can’t start server: can’t create PID file: No such file or directory
200504 21:26:06 mysqld_safe mysqld from pid file /var/run/mysqld/mysqld.pid ended
200504 21:37:21 mysqld_safe Starting mysqld daemon with databases from /var/lib/mysql
200504 21:37:21 InnoDB: Initializing buffer pool, size = 8.0M
200504 21:37:21 InnoDB: Completed initialization of buffer pool
200504 21:37:22 InnoDB: Started; log sequence number 0 9459016
200504 21:37:22 [ERROR] /usr/libexec/mysqld: Can’t create/write to file ‘/var/run/mysqld/mysqld.pid’ (Errcode: 2)
200504 21:37:22 [ERROR] Can’t start server: can’t create PID file: No such file or directory
200504 21:37:22 mysqld_safe mysqld from pid file /var/run/mysqld/mysqld.pid ended

I appreciate any help.

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@mikeaph wrote:

I have a PBXact at my office. It is local on the network with everything configured appropriately. I would like to be able to access the GUI from my home.

As I understand it, one option is to change the SSH port to something random on the PBX, then set my router’s firewall at the office to allow only traffic from my home public IP on that port. Then, in the PBX firewall, I would need to allow SSH traffic on the Internet Zone. Typically, that would be unwise because you are exposing yourself to attacks via SSH. However, am I correct in understanding that since I am blocking all traffic at my router, it should be secure since I am only allowing traffic from my home IP? With this setup, I would be able to SSH into my PBX and, at the same time, create a tunnel so that I can access the GUI.

Another option, I believe, is to add my home public IP to the trusted, local or other firewall zone and then allow access to the GUI in whichever of those zones I choose? Then, of course, I would still block all traffic at the firewall level except for my home public IP. Would that still be secure? And if that is a viable option, which zone would be advisable?

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@marcocentr wrote:

Hello,
i try to configure freepbx whit voicemail when i have ring group as destination in inbound routes.

i have 3 extension telephone
701
702
703

the schema is:
inbound route—>destination:ring group ( 10)---->Destination if no answer ---->voicemail701 (busy message)

when i call the phone ring… and i can talk… but when i not respond the voicemail not start.

can help me?

thank you

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@vaibhav wrote:

I have created a simple test script that will dial an extension and deliver a message. Its working fine. It dials an extension and plays an audio file.

Can someone point me to a guide or commands which will dial an extension and instead of playing an audio file, it will read out a message using TTS – something like “Extension 1001 is not reachable”. I need something basic, nothing fancy. This is for notifications & alerts.

My current script is as below.

<?php
//https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+AMI+Events

$socket = fsockopen("127.0.0.1","5038", $errno, $errstr, 10);
if (!$socket)
{
	echo "$errstr ($errno)\n";
}
else
{
	fputs($socket, "action: login\r\n");
	fputs($socket, "username: admin\r\n");
	fputs($socket, "secret: xxxxxxxxxxxxx\r\n\r\n");

	fputs($socket, "Action: Originate\r\n");
	fputs($socket, "Channel: PJSIP/1022\r\n");
	fputs($socket, "Application: Playback\r\n");
	fputs($socket, "Data: silence/1&tt-weasels\r\n\r\n");

	$wrets = fgets($socket,128);

	fputs($socket, "Action: Logoff\r\n\r\n");
}

Thanks!!

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@jefbuan wrote:

I enable and load channel pjsip to transition my sip extensions to use pjsip.but and I enable
tcp - 0.0.0.0 - All in pjsip setttings…now I cant start asterisk , in restart loop now…hope someone can help me
[2020-05-05 09:20:24] ERROR[5421]: res_pjsip/config_transport.c:652 transport_apply: Transport ‘0.0.0.0-tcp’ could not be started: Address already in use
[2020-05-05 09:20:24] ERROR[5421]: res_sorcery_config.c:307 sorcery_config_internal_load: Could not create an object of type ‘transport’ with id ‘0.0.0.0-tcp’ from configuration file ‘pjsip.conf’

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@faisalkhan wrote:

how to detect if someone from softphone is auto answering the calls from the CDRs or any other way to detect if someone set’s his phone on auto answering.

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@omarara wrote:

Hi All,
I need some help as i am new to this system.
I have USB dongles around 30,I am sending from my server several prefixes exactly to each port"dongle" like this 001,002,003 ,… etc. ,so that the call goes to specific dongle
so i don’t know how can i do this! , should i add each dongle as a trunk ? or should i add them as Extensions ?
How can i make routes for that system based on prefixes per dongle !
Thanks in Advance

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@andrewpaes wrote:

Hi,

I would like to know where can I find (if it does exist) the video file for a call with video.

I already have all the things running properly:

• Call recording enabled for extensions as Force
• Call recording enabled for routes as Force
• Video Support enabled
• Video Codecs checked
• 3CX Softphone calling with video
• 3CX recording the video locally
• WAV file available on FreePBX CRD Reports

Now I would like to have/find the video file (MPEG, H264, etc).

Am I missing some config or it can’t be done?

I’m running on FreePBX 13.0.194.2 and Asterisk 13.

Tnx in advance,

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@dobrosavljevic wrote:

I’ve got a client that extensively uses the Call History panel in the UCP to listen to past call recordings. It’s worked great for the past two years, however as the call history has gotten bigger and bigger as they use it it’s taken longer and longer for the information to load or to perform a search in it.

I suspect because it performs the function on the entire call history over the lifetime of the phone system.

Is there a way to truncate that call history down to only say the last month of calls to speed up the performance as they don’t really care or need access to the history past that.

Is this maybe something I should setup a support ticket on?

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@maskicom wrote:

Hi all,

I setuped a FreePBX 14 instance running asterisk 13.32.I also configured a few Conference Rooms so we can have conferences since we are all confined and remote working. When one join the conference from external line, audio is fine. When they join from their Mitel 6867i SIP phone (pjsip) directly through the room’s 4 digits number, it gets really choppy. All the other calls from these remote workers are perfectly fine so I don’t think it’s a bandwith problem on their side. My system is running in ESXi 6.7 and I assigned 2 vCPUs and 4gb of RAM. We never really get more than 8 participants in the rooms altogether. It really seems like there are some kind of transcoding when using the conf. room with the SIP Phones that plague the audio quality. I read stuff from years ago that conferencing was too intensive for virtualization but it was like 5 or 6 years ago…

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@stappaf wrote:

Hello freepbx -ers

I have a problem sharing a phonebook on my Gigasets.

I use Gigasets N720 Dect Manager PRO and some mobile devices, and FreePBX with extensions for phone numbers.

Can someone help me to get a common contact lists on all mobiles the one if int(ernal) button and the other if phonebook button is pressed on the Gigasets?

Best would be if there is a way to share the freepbx contact list, but if a other list is easier i am open for all ideas.

I am a newbee with linux and would need help in easy words

Thank you!
Stefan

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