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Queue_log to MySQL in 10.13.66 distro?

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Continuing the discussion from Queue log into MySQL database:

Continuing the discussion from Registering queue log in database:

Continuing the discussion from Registering queue log in database:

I am trying to get this working with the latest distro 10.13.66 and hopefully someone can steer me in the right direction. I have tried the above mentioned guide and scoured the internet.

In logger_general_custom.conf I have:
queue_adaptive_realtime = yes
queue_log_to_file = yes

I did not have the extconfig.conf file or the res_config_mysql.conf files, so I created them:

In extconfig.conf I have:

[settings]
queue_log => mysql,general

In res_config_mysql.conf I have:

[general]
dbhost = 127.0.0.1
dbname = asteriskcdrdb
dbuser = mydbusername
dbpass = mydbuserpassword
dbport = 3306
requirements=warn

Here is the structure of my queue_log table which I got from this link: http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

CREATE TABLE tablename (
id bigint(255) unsigned NOT NULL AUTO_INCREMENT,
time varchar(26) NOT NULL DEFAULT '',
callid varchar(40) NOT NULL DEFAULT '',
queuename varchar(20) NOT NULL DEFAULT '',
agent varchar(20) NOT NULL DEFAULT '',
event varchar(20) NOT NULL DEFAULT '',
data varchar(100) NOT NULL DEFAULT '',
data1 varchar(40) NOT NULL DEFAULT '',
data2 varchar(40) NOT NULL DEFAULT '',
data3 varchar(40) NOT NULL DEFAULT '',
data4 varchar(40) NOT NULL DEFAULT '',
data5 varchar(40) NOT NULL DEFAULT '',
created timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP,
PRIMARY KEY (id),
KEY queue (queuename),
KEY event (event)
) DEFAULT CHARSET=utf8;

Nothing seems to work. I also tried using the dsn,tablename instead of the [general] context for the extconfig and res_config_mysql files. I restarted MySQL and reloaded Asterisk afterward.
Does anyone know the trick to getting the queue_log into MySQL with the FreePBX distro? It would be really useful for queue status boards and reports.

Thanks,
Nowell


Voicemail no longer a destination?

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I restored voicemail.conf and now the are enabled and voicemail has reappeared as an option. :smiley:

[solved] EndPoint Manager not finding Cisco phone

Programming my polycom vvx310

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25 years is a life time ill be retired then LOL

Struggling with Digium T1 Card

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Hello,
I am trying to use an old Digium TE420 in a new Supermicro build.
When I boot up, the card has power and is recognized by Dahdi and shows up in the Dahdi config module but does not work. In the Dahdi Config module I get BLU/YEL/RED/LFA
The LEDs sequence indicating that it's not ready to be used.
In another system with the same FreePBX distro install it works. When idle on this system I only get RED/LFA and the LEDs are solid indicating they are waiting for a T1 to be connected.
I am pounding my head against the wall as to why it's not working. Any recommendations as to where to start other than use an old motherboard?

Yealink T-32G won't provision!

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Thanks for the response. I did as suggested. the tftpd entries in my messages file are these:

Feb 1 16:08:08 FreePBX in.tftpd[18949]: RRQ from 10.7.1.63 filename y000000000032.cfg
Feb 1 16:08:08 FreePBX in.tftpd[18949]: sending NAK (1, File not found) to 10.7.1.63
Feb 1 16:08:08 FreePBX in.tftpd[18950]: RRQ from 10.7.1.63 filename 001565831b55.cfg


That is it. I look in my tftpboot directory, and file 001565831b55.cfg is there with necessary rights and owner "asterisk".

What is going on?????

Power supply on my dell out of whack need a new supplier for freepbx

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yes, portability on the sky train / buss or by car so its form factor is really small for demo purposes. What about FXS out to analog PA systems? If I had a small valcom with speaker, they may be impressed :smile:

Power supply on my dell out of whack need a new supplier for freepbx

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covered by the 300* (pad 20 20 characters)


Networking issues

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@cyberdocwi

I have to say that I had planned to wipe and reload anyway. I was just hoping to get it to work first so I would have it narrowed down and could better handle any issues that may arise in the next load.

I have been looking at taking one of my little rack mountable servers and turning it into a Firewall/Router using Untangle. It seems quite powerful for the average home user like myself. Has all the features I would ever need in the base free distro and is simple to use. I am just learning and sometimes I have to ask very dumb questions in forums like these just to get by. I have to say if it as not for patient forum members I would never have learned anything about Linux, FreePBX, etc. I will perform the reload tonight and hope it all goes well. I will just set it to DHCP to start with and see where we go from there. If I can get to the Web UI from there I will try it that way then make changes using the FreePBX front end. Wish me luck!!

Power supply on my dell out of whack need a new supplier for freepbx

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What do you mean? BTW yes, totally forgot about the ATA adapters. I used to have one. Its been years but would need to have a fresh supplier of adapters.

Who is a reliable supplier of polycom phones in the US?

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So I have a nice set of Polycom vvx310s but they were picked up locally from a buyer who did not like the delayed voip signal. Looking for a reliable competitively prices re-seller in the states that sell polycom. Thanks

Who is a reliable supplier of polycom phones in the US?

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ohh and they need to be mid west or west cost to keep shipping times down to the Vancouver BC market.

Power supply on my dell out of whack need a new supplier for freepbx

Yealink T-32G won't provision!

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log into the phone and take a look at what got loaded. it looks like the config file is provisioning properly however something is not loading correctly. my guess is that the sip server address will be missing.
i would also compare the firmware level on the phone to the firmware level you have specified in the EPM. go to /tftpboot/yealink/1 (or whatever firmware slot you are using) and look at the file called version. it will list the firmware version by phone type. if the firmware levels are significantly different, then manually update the firmware on the phone by downloading it from the pbx to your laptop/pc and then log into the phone and file the upgrade screen. we have discovered that if the firmware on the phone is too old, that it will not provision from the tftp or ftp server properly.
if the firmware matches, then i would factory reset the phone (either log in and reset it) or simply hold the "ok" button until the phone factory resets and then provision the phone again.
my guess is that you will find that everything but the sip server address loaded correctly. if you manually enter it you will probably see the phone register.

Struggling with Digium T1 Card

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I'm coming to the conclusion that this card just isn't going to work in an i5 based motherboard, I think it needs to be Core Duo.

Is it any help or indicator that now I get the error in my terminal wc4xxp Interupts not detected?


Digium D70 and Rest Apps

Endpoint Manager and Cisco 9971, 9951

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I have a couple of Cisco 99x1's in for testing at the moment. I have them both registered against the server and working, by creating the SEP{mac} files manually. I did it manually because they are not detected by Endpoint Manager.

I suspect that the mac addresses of he phones are not in the 'known list' perhaps? The phones have mac addresses starting: ACA016 and B8BEBF

Is there a log file for EPM somewhere so I can see what is happening?
Also, the EPM seems not to support the firmware for these phones, I've got the files downloaded from Cisco, is there any way to have EPM manage the files if I put them in a special location or something?

I'm sure I can work this out, I just need a bit of a pointer for where to start,
Thanks!

Confused about Distro upgrades

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(Note I'm running FreePBX 13.0.54)

  1. I have my PBXes set to do Automatic Updates each month, but this last upgrade didn't happen.

  2. When I run the upgrade to go from 6.12.65-31 to 6.12.65-32 using the WebGUI it says the upgrade process successfully upgraded... but then it didn't. rebooted. same result.

•2016-02-02 12:41:31: (6.12.65-32) Starting stage 1
•2016-02-02 12:42:37: (6.12.65-32) Stage 1 complete
•2016-02-02 12:42:37: (6.12.65-32) Starting stage 2
•2016-02-02 12:43:17: (6.12.65-32) Upgrade process successfully completed!

  1. So then I ran the upgrade using the Linux CLI... and it looked nothing happened...
    but then it stopped responding and I guess it did upgrade, because I rebooted and I saw the following:

•2016-02-02 12:57:23: () Update sequence initialized
•2016-02-02 12:57:23: () Requesting updates
•2016-02-02 12:57:23: () Processing update
•2016-02-02 12:57:23: (6.12.65-32) Starting stage 1
•2016-02-02 12:58:02: (6.12.65-32) Stage 1 complete
•2016-02-02 12:58:02: (6.12.65-32) Starting stage 2
•2016-02-02 12:58:52: (6.12.65-32) Stage 2 complete
•2016-02-02 12:58:52: (6.12.65-32) Starting stage 3
•2016-02-02 12:58:52: (6.12.65-32) Stage 3 complete
•2016-02-02 12:58:52: (6.12.65-32) Starting stage 4
•2016-02-02 12:58:53: (6.12.65-32) Stage 4 complete
•2016-02-02 12:58:53: (6.12.65-32) Starting stage 5
•2016-02-02 12:58:53: (6.12.65-32) Stage 5 complete
•2016-02-02 12:58:53: (6.12.65-32) Starting stage 6
•2016-02-02 12:58:53: (6.12.65-32) Stage 6 complete
•2016-02-02 12:58:53: (6.12.65-32) Upgrade script complete

  1. Looking at the wiki (http://wiki.freepbx.org/display/PPS/FreePBX-Distro-6.12.65) it says on the last Distro: "FreePBX Distro 6.12.65-100 This will take your 6.12.65 version system to a 10.13.66-1 version and track. Please note 10.13.66 is the Current STABLE and 6.12.65 will be End of Life 12-31-15"
    Should I really be running 10.13.66 instead of 6.12.65?

Valcom VIP-201A Configuration

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I am trying to configure a Valcom VIP-201A SIP Paging Server for our FreePBX Distro system. I am having trouble getting the configuration correct so that the VIP-201A is registered to the PBX and is "Reachable". Here is where I am:

  • Valcom VIP-201A does not like long SIP passwords. I have used a shorter password and now I can get it to register.
  • When I issue a "sip show peers" at the asterisk command line, the device always says "UNREACHABLE".
  • After some period of time, the device no longer receives incoming calls and therefore, no paging through the Valcom device is available.

I have tried to use both chan_sip and chan_pjsip, with the same basic results. I just can't seem to find a configuration that works. In the Valcom VIP-102B IP Solutions Setup Tool, I have the SIP configuration set this way:

  • SIP Mode: Station
  • Phone Number: 1234 (The extension number)
  • Description: blank
  • Authentication Name: 1234 (The extension number)
  • Secret: abcd1234ab12 (alpha-numeric password, no special characters, 12 letters and numbers)
  • Realm: blank
  • SIP Servers:
    • Primary: 192.168.1.200 (PBX IP), Port 5061 (chan_sip port)
    • Backup 1, 2 , 3: blank
  • Register: Checked
  • DNS SRV: Not Checked
  • Outbound Proxy: blank
  • Outbound Port: 5060 (didn't change it since I'm not using a proxy)
  • SIP Port: 5061 (chan_sip port)
  • Idle Timeout (secs): 0
  • RTP Port: 20000
  • Max Call Timer (secs): 0
  • Night Ring: Not Checked
  • Pre-Announce Tone: Checked
  • Store and Play: Not Checked

Other settings that may be of interest:

  • Continuous Beacon: Checked
  • Audio Output Volume: -15

I appreciate any feedback anyone can provide!

Change language of conference module

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This is fixed in freepbx 13. You can select and change the language per conference

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