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TTS Engine Custom - Amazon Polly - 24 languages

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I am trying AWS polly out, but have come to a brick wall.
When I execute the script the wav file is not created.
and no obvious error message.
Is there a way to enable a debug reporting ?

Any help appreciated.

Scripts as above
extract from full log as follows

[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] pbx.c: Executing [1@ext-tts:4] AGI("SIP/Sip_61731230612-0000008e", "propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node") in new stack
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/propolys-tts.agi
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: TTS AGI Started
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: Generated WAV file: /var/lib/asterisk/sounds/tts/aws-polly-tts-df04e889735f84e9670a11fbe67d89b9.wav
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: TXT file: /var/lib/asterisk/sounds/tts/aws-polly-tts-df04e889735f84e9670a11fbe67d89b9.txt
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: Text to speech wave file doesnt exist, lets create it.
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: Executing aws-polly
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: File was not created!
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: propolys-tts.agi,"this is a test of aws Polly",aws-polly,/usr/bin/node: TTS AGI end
[2017-10-09 14:08:25] VERBOSE[4758][C-00000092] res_agi.c: AGI Script propolys-tts.agi completed, returning 0

Permissions seem ok too


Panasonic KX-TGP600 dont reconnect after Asterisk Restart

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I will give you feedback when i know if a lower value worked.

Today i came across that the phone with asterisk state "UNKNOWN" or "UNREACHABLE" called me on the phone.

So it is a RTP issue with external phones. The RTP Ports are "0" and the phones are not registred anymore.
But the basestation by the customer had a green lamp and signaled that everything is fine.

The latest firmware is already installed.

TTS Engine Custom - Amazon Polly - 24 languages

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Belay my last post
Word of warning for others following this thread.
When you cut and paste your AWS access keys do not lose the commas at the end of the line.
:grin:

Setup SNMP v3 on FreePBX 14

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After running into a lot of incomplete information on the web and here in the forums on how to properly setup SNMP to monitor resources on a FreePBX installation I've decided to put together a guide on what worked for me here for anyone that would like to setup the same.

1. Setup the snmpd agent:


# yum install net-snmp asterisk-snmp

The above command will install the snmpd agent and the latest version of the asterisk snmp module onto the machine. However there is a caveat, if you don't have the latest version of asterisk installed on the machine that you are attempting to setup snmp on the module will fail to load later on in the process. There are a couple of workarounds to this. First you could run yum update to update all the packages on your machine to the latest version automatically or you could install the version of asterisk-snmp that matches the specific installed asterisk version that's currently running on your machine.

To find out what version of asterisk is currently installed and to install the proper version of the asterisk-snmp module you can run the following commands.

# rpm -qa | grep asterisk | grep core
asterisk13-addons-core-13.17.2-1.sng7.x86_64

To install the exact version of the asterisk-snmp package that would match your installed version of asterisk your yum install command would actually look like this:

# yum install net-snmp asterisk13-snmp-13.17.2-1.sng7.x86_64

If the installation was successful you can move on to configuring the snmp agent and asterisk snmp module to properly collect and report data for you.

First, we'll modify the /etc/snmp/snmpd.conf file to configure a couple of things. First we'll setup a snmp v3 user that's allowed to connect to the service and receive data. We'll make this a read-only user since we are not really using snmp to modify any settings on this box (I am not even sure that you can even use snmp to modify any asterisk settings). Second we'll configure AgentX since that's how asterisk provides the snmpd agent with informaiton.

To accomplish both add the following lines to your snmpd.conf:

# Asterisk SNMPv3 user
createUser AsteriskMon MD5 [CHANGEME] DES
rouser AsteriskMon priv

# I am also monitoring to ensure that at least one asterisk process is running on this machine at all times.
proc asterisk
sysObjectID .1.3.6.1.4.1.22736.1

# Enable AgentX support
master agentx
agentXSocket /var/agentx/master
# Set permissions on AgentX socket such that process in group 'asterisk' will be able to connect
agentXPerms 0660 0550 nobody asterisk

Make sure to modify the [CHANGEME] block above and provide the system with what you want to be your password and encryption pass phrase.

Second, we'll modify the /etc/asterisk/res_snmp.conf file to enable snmp within asterisk:

[general]
; We run as a subagent per default -- to run as a full agent
; we must run as root (to be able to bind to port 161)
subagent = yes
; SNMP must be explicitly enabled to be active
enabled = yes

Third, we'll enable snmpd to start automatically on boot, start the service and ensure that the asterisk module is properly loaded:

# chkconfig snmpd on
# service snmpd start

To load the asterisk module and ensure it is running properly get into the asterisk CLI and run the following commands:

# asterisk -r
pbx*CLI> module load res_snmp.so
pbx*CLI> module show like snmp
Module                         Description                          Use Count  Status         Support Level
res_snmp.so                    SNMP [Sub]Agent for Asterisk         0          Running        extended
1 modules loaded

Lastly, the /var/agentx directory doesn't automatically get the correct permissions assigned for asterisk to be able to write to the master socket file so we need to make this one time change as the last step in the configuration process:

# chmod 755 /var/agentx

2. Setup the monitoring tools:


Normally you'll want to setup a tool on another host to access and store the snmp information from your phone system (Zabbix in my case). However, if you'd like to test your configuration after the above setup procedure you can do the following.

First, we'll need to install the snmp utilities package which contains tools like snmpwalk that will allow us to do our testing.

# yum install net-snmp-utils

Second, we'll need to load the appropriate MIB files that are provided by asterisk right here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+MIB+Definitions

You'll want to save the ASTERISK-MIB.txt and DIGIUM-MIB.txt to /usr/share/snmp/mibs/

Lastly, we'll need to modify the snmp.conf file to use the MIBs that we've just added. Modify the /etc/snmp/snmp.conf file and add the following line:

mibs +ASTERISK-MIB

Notice that the above file is snmp.conf and NOT snmpd.conf. That's also the only line in my snmp.conf file.

Now you should be able to run the following command and get information spat back out at you:

# snmpwalk -v 3 -u AsteriskMon -l authPriv -a MD5 -A [CHANGEME] -x DES -X [CHANGEME] localhost asterisk

3. Sample interesting information and wish list


Following are a few examples of interesting data that I am collecting using snmp from my phone system.

1: Total currently active calls

ASTERISK-MIB::astConfigCallsActive
OID: .1.3.6.1.4.1.22736.1.2.5.0

# snmpwalk -v 3 -u AsteriskMon -l authPriv -a MD5 -A [CHANGEME] -x DES -X [CHANGEME] localhost ASTERISK-MIB::astConfigCallsActive

2: Total calls processed since asterisk was last restarted

ASTERISK-MIB::astConfigCallsProcessed
OID: .1.3.6.1.4.1.22736.1.2.6.0

# snmpwalk -v 3 -u AsteriskMon -l authPriv -a MD5 -A [CHANGEME] -x DES -X [CHANGEME] localhost ASTERISK-MIB::astConfigCallsProcessed

3: Asterisk up time

ASTERISK-MIB::astConfigUpTime
OID: .1.3.6.1.4.1.22736.1.2.1.0

# snmpwalk -v 3 -u AsteriskMon -l authPriv -a MD5 -A [CHANGEME] -x DES -X [CHANGEME] localhost ASTERISK-MIB::astConfigUpTime

Now I really wish it was possible to collect additional info through snmp that currently doesn't seem to be an option. I would love to be able to get the latency in ms of all peers that are setup in asterisk. I would be primarily interested in knowing when a SIP trunk is lagging or disconnected, however knowing how individual peers are doing would be a nice bonus as well. It would also be nice if I could find out the status of registered trunks directly through SNMP.

The above is certainly not an exhaustive list of information that would be nice for me to be able to retrieve from asterisk through snmp but it would make me ecstatic if support for those was added in some day.

I realize this wish list has nothing to do with FreePBX but my hope is that somebody that can do something about adding additional functionality to snmp for asterisk will read this and consider extending the functionality of the snmp implementation for asterisk some day. A boy can dream.

Enable SNMP in FreePBX Distro

Confused about context/rules

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Found it this weekend. It was so obvious, that I overlooked it for hours.
The "Outbound CID" on the Trunk was set. It seems that this overrides any manual setting.

Now there is only one problem remaining. If I get an incoming call everything is working fine, if I make an outgoing call both parties can not hear each other.

But: If I create a extension and call any external number (outgoing to ISP) its working, if I call the SfB trunk its working.
When I call from SfB to Mobile RTP debug does not show incoming RTP packets, only outgoing.

Run JAR at boot complete

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Hi all,

I am using FreePBX 14.0.1.3 version. When PC boot complete, I want to run the JAR file.

For that I had followed below ways, but it was not working,

  1. I had put the command in /etc/rc.local file
    su - root -c "java -jar /home/test/testing.jar"

  2. I had put the command in /etc/rc.local file
    java -jar file.jar &
    exit 0

Suggest me the right way, Thanks in advance !

Issues after upgrade to FreePBX14 - LoadLicenseIfExists - Incompatible File Format

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ok, I've bailed on the upgrade and re-installed 13. Do you still want the logs?


Wanpipe installation error

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why i'm getting when i install wanpipe .

/usr/include/wanpipe/wanpipe_common.h: In function wan_netif_set_ticks: //usr/include/wanpipe/wanpipe_common.h:2646:5: error: struct net_device has no member named trans_start dev->trans_start = ticks; ^ make[2]: *** [/usr/src/wanpipe-7.0.20/kdrvtmp/sdladrv_src.o] Error 1 make[1]: *** [_module_/usr/src/wanpipe-7.0.20/kdrvtmp] Error 2 make[1]: Leaving directory `/usr/src/kernels/3.10.0-693.2.2.el7.x86_64' make: *** [all] Error 2

OS : centos 7 64 bit
kernel : 3.10.0-693.2.2.el7

REST API documentation

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I'm looking for some basic documentation on the REST API module in FreePBX. Simple things like clicking a button and dialing out. I can't seem to find documentation on this module. Can anyone point me in the right direction?

REST API documentation

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There unfortunately is no documentation on it and what you want to do doesn’t not exist in it

Issues after upgrade to FreePBX14 - LoadLicenseIfExists - Incompatible File Format

REST API documentation

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Is this more of an Asterisk feature?

REST API documentation

Run JAR at boot complete

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rc.local should work. You probably need to put the full path to the java command. If you are stuck then check on Stack Overflow / Server Fault sites because this is not really a FreePBX question.


Issues after upgrade to FreePBX14 - LoadLicenseIfExists - Incompatible File Format

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Sorry, I’m also passed the point of no return. I had to restore to the pre-upgrade snapshot because it was causing too much downtime.

Softphones, External Clients, VPN, Etc

View "Trunk" traffic?

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Is there a way to view the traffic on trunks (like Wireshark).

I have a SiP Station on a PBX that is connecting great. Now I want to pass DTMF-SMDI so I need to se just what the system is sending. The hope is to replace a dieing voice mail

Whoops error after install

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Hi I am getting the following error. Please help me to solve this.

Whoops \ Exception \ ErrorException (E_ERROR)
Uncaught exception 'Whoops\Exception\ErrorException' with message 'file_get_contents(/etc/freepbx.conf): failed to open stream: Permission denied' in /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Exception/Frame.php:127 Stack trace: #0 internal function: Whoops\Run->handleError(2, 'file_get_conten...', '/var/www/html/a...', 127, Array) #1 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Exception/Frame.php(127): file_get_contents('/etc/freepbx.co...') #2 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Exception/Frame.php(204): Whoops\Exception\Frame->getFileContents() #3 /var/www/html/admin/views/whoops/views/frame_code.html.php(24): Whoops\Exception\Frame->getFileLines(-11, 40) #4 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Util/TemplateHelper.php(249): require('/var/www/html/a...') #5 Array) #6 /var/www/html/admin/libraries/Co

Error[6641] chan_sip.c:4273 Serious Network Trouble; __sip_xmit returns error for pkt data

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Well, I have relized that the errors are originated from the same IPs. They are allways the same and they are connected via VPN, like other extensions.

Regards

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