If you want to set up timed call flow in Elastix but still have the ability to override for holidays and when the office is open late you have a few extra steps to add.
We will assume you have your queues and extensions setup for this video. If you havent set your extensions up see our other video on setting up extensions.
We have used 2 day/night modes, One at before the call enters the time condition, This means that you can override day service for holidays etc and another at the end that means the call can be forced to go to a night queue instead of voicemail.
I hope you found this useful and keep coming back for more.
We have recently added the 5th system to the customers international VoIP network. This system was for their Polish office and is linked to their Tokyo, Sydney, Singapore and London office systems.
For this site a Sangoma FREEPBX 60 system was chosen for ease of remote deployment and reliability. The Tokyo & Sydney offices already has a Xorcom XR2000 systems whilst the London and Singapore offices have a OpenvoxAsterisk solutions. .
For the New International offices FreePBX systems were chosen as they provide a full turnkey system that can be sent out to the office plugged in. The systems initially obtain their IP address by DHCP and once a port is forwarded through the firewall to this address a fixed IP address is assigned and the customer firewall updated. Access to The GUI is by a SSH tunnel so that other than a random port for SSH and a port for IAX2 no other ports need to be opened on the customer firewall. Endpoint manager makes the deployment of handsets on the remote systems a simple and reliable process.
All systems have been linked by IAX2 trunks and the dial-plan configured so that desk to desk calls can be made between all offices and outgoing calls break out of the closest geographic system, for example a user in Sydney making a call to a UK number will have the call originate from the London system and the same goes for Tokyo, Singapore and Polish users calling UK or international numbers.
The network of systems is key to the support of the customers 24×7 support service. This is controlled by a dial-plan that is complicated by the fact that Japan does not have “Daylight saving” so even though the calls land on the UK system we had to configure the dial-plan to take into account local time in Tokyo and not base routing solely on UK time. This has proved reliable and very successful.
All systems on the network are monitored 24×7 by our Nagios monitoring platform, Not only monitoring Asterisk but also monitoring the status of the international IAX2 trunks.
The Elmeg IP290 are a clone of the Old Snom 190 Sets. and these did support auto configuration. We hoped that it was a simple change to get the files working with the Elmeg.
It turns out that it wasn’t, There are a few gotchas.
The phones dont seem to support tftp download, Just http and https
They dont support sub directories. So files must be in the root directory of your webserver.
In the last few weeks a large sleeping troll has come out of hibernation and seems set on disrupting the whole voip market.
Quote
“BT is engaged in licensing an extensive range of standards related patents that address the key features of SIP Trunking providers and VOIP operators providers.
BT’s Patents address a wide range of fundamental capabilities now in widespread deployment, such as:
Setting up a call
Breaking out to other networks
Managing resources efficiently
Registering terminal to a network
Cost effective call completion
Monitoring and alerting of IP call quality”
Well that pretty much covers all of the workings of a SIP network. A full list of the patents is here .
But its not Just BT, AT&T also have claims over SIP as well see here for a list.
It seems that some of the major patent holders see more money in the licencing of the now ubiquitous SIP protocol than maybe supplying it to customer. Which is a shame as the only ones who will make any money will be the Lawyers in the end.
The Asterisk Queue application has an option that will run a macro on answer, This can be very useful when integrating with CRM such as Capsule or call centre applications.
This option isnt included in freepbx, Though this can be hand coded it isn’t best to do this when using Elastix, AsteriskNoW or any other freepbx based system.
To add this option We have written a couple of patched versions of the relevant freepbx pages that can be downloaded here , You will also need to add a extra field to the mysql database as follows
Log in to mysql: mysql -u root -p
Enter password
mysql> use asterisk
mysql> ALTER TABLE `queues_config` ADD `qmacro` VARCHAR( 255 ) NULL;
mysql> describe queues_config;
You should now have something like this:- | qmacro | varchar(255) | YES | | NULL | | as the last line of the table.
Now download the tar file and unpack it. then copy the two files to the /var/www/html/admin/modules/queues directory.
On loading the queue page in freepbx you will now have the “Queue macro on answer” box
In this box you put the macro name you wish to run when a member answers a call.
This a simple dialplan that runs a php script to log calls to the capsule crm
capsual.php
<?php
$today = date(“F j, Y, g:i a”);
$duedate1 = date(“Y-m-d”);
$duedate2 = date(“H:i:s”);
$Token = ‘YOUR CAPSUAL API CODE’;
$number = $_GET[‘strCallid’];
$datetime = $today;
$duedate = “$duedate1″.”T”.”$duedate2″.”Z”;
echo $duedate;
$myxml=”<?xml version=”1.0″ encoding=”UTF-8″?>n
<task>n
<description>Call recieved from $number at $datetime. Please update and assign this task if required</description>n
<dueDateTime>$duedate</dueDateTime>n
<category>incoming call</category>n
</task>”;
// The URL to connect with (note the /api/ that’s needed and note it’s person rather than party)
// SEE: http://capsulecrm.com/help/page/api_gettingstarted/
$capsulepage = “https://youraccount.capsulecrm.com/api/task”;
echo $capsulepage;
echo $number;
// Initialise the session and return a cURL handle to pass to other cURL functions.
$ch = curl_init($capsulepage);
// set appropriate options NB these are the minimum necessary to achieve a post with a useful response
// …can and should add more in a real application such as
// timeout CURLOPT_CONNECTTIMEOUT
// and useragent CURLOPT_USERAGENT
$options = array(CURLOPT_USERPWD => “$Token:x”,
CURLOPT_HTTPHEADER => array(‘Content-Type: application/xml’),
CURLOPT_HEADER => true,
CURLOPT_RETURNTRANSFER => true,
CURLOPT_POST => true,
CURLOPT_POSTFIELDS => $myxml
);
curl_setopt_array($ch, $options);
// Do the POST and collect the response for future printing etc then close the session
$response = curl_exec($ch);
$responseInfo = curl_getinfo($ch);
curl_close($ch);
echo $responseInfo;
echo $response;
?>
Whenever sendmail has to deliver mails to other hosts which cannot be reached at that time, the messages are kept in the queue and are marked as “Deferred: Connection timed out”. Although the other hosts could be reached again and you want to tell sendmail to flush the mail queue, the command
sendmail -q -v
does not really try to reconnect to these hosts and still assumes that the connection timed out. The reason is that the hoststatus is cached, per default for a period of 30 minutes. Using
sendmail -OTimeout.hoststatus=0m -q -v
you can re-run the mail queue and force sendmail to reconnect to the hosts. You may want to define an alias for that, say, ‘sendmail-flush-timeouts’.
When using Skype for SIP trunks with Asterisk a simple an neat way to enable DDI calling for the skype names is to use the “extension” option.
This means that the ‘To’ in in the sip header is set to what you set.
This can then be picked out with a simple little bit of dialplan
tar zxvf dahdi-linux-complete*
tar zxvf libpri*
tar zxvf asterisk*
For the next set of commands it is important to follow the proper order: DAHDI first, then libpri, then Asterisk.
Install DAHDI.
cd /usr/src/dahdi-linux-complete*
make && make install && make config
Install libpri.
cd /usr/src/libpri*
make && make install
Change to the Asterisk directory.
cd /usr/src/asterisk*
In the next step, running the “configure” script will vary depending on whether your system is 32-bit or 64-bit. (Watch the video for more details.) When the menuselect command runs, select your options, then choose “Save and Exit” and the install will continue.
Use this command if you are installing Asterisk on 32bit CentOS.
./configure && make menuselect && make && make install
Use this command if you are installing Asterisk on 64bit CentOS.
./configure --libdir=/usr/lib64 && make menuselect && make && make install
Optional: If you ran into errors you will want to clean the install directory before recompiling.
make clean && make distclean
Once you have an error-free install, copy the sample files from the configs subdirectory into /etc/asterisk.
make samples
Then add the Asterisk start script to the /etc/init.d/ directory
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