Categories
Asterisk Support Elastix Support Knowledge Base Technical

Running a Macro on answer for Asterisk queues.

asteriskThe 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

  1. Log in to mysql:   mysql -u root -p
  2. Enter password
  3. mysql> use asterisk
  4. mysql> ALTER TABLE `queues_config` ADD `qmacro` VARCHAR( 255 ) NULL;
  5. 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

queuemacro

In this box you put the macro name you wish to run when a member answers a call.

For example:-

[macro-logit]
exten => s,1,Noop( capsule crm intergration ${crminfo} ${CALLERID(all)})
exten => s,n,Set(foo=${CURL(http://127.0.0.1/directory/capsual.php?strCallid=${crminfo})})
exten => s,n,Noop(${foo})
exten => s,n,Hangup()

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;
?>

Have fun

 

Categories
Knowledge Base Technical

Flushing your sendmail queue.

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’.

You can set further options in /etc/sendmail.cf.

Categories
Handsets

Digium Handsets

The Only Phones Built Specifically For Asterisk

  • asteriskEasy provisioning from Asterisk or AsteriskNOW
  • Integrated with Asterisk voicemail, directory, parking, call recordings, call queues and more
  • Build custom phone apps with a simple JavaScript API

Digium’s family of IP Phones are the first on the market built specifically for use with Asterisk and Asterisk-based systems. All models include HD audio and plug-and-play deployment at a price that fits any budget. With multiple line appearances, context-aware soft keys, and advanced applications that integrate directly with Asterisk features, the Digium phones offer a better user experience than any other phone on the market.

Asterisk Phone Features

Smart Software

Access to information is the key to productivity in today’s business environment. The integrated applications that come standard with all Digium phones put critical information at your fingertips. With voicemail, call log, contacts, phone status, user presence, parking, call recording and call queue interface, the Digium phones provide simple, intuitive access to a wealth of information, saving valuable time.

 Simplified Provisioning

Standards-based IP phones have a reputation for being difficult to install and configure. Most systems require changes to network configurations or additional components to facilitate deployment. Digium phones support plug-and-play provisioning. Simply plug in the phone and it will automatically discover Asterisk systems on the network. Select the user you want to assign to the phone and the proper configuration is instantly loaded. For larger deployments you can pre-assign phones by tying a MAC address to an Asterisk user. It’s that simple.

Custom Applications

Most desktop phones come with a fixed feature-set that is determined exclusively by the manufacturer. Digium phones are different. All models include the Digium app engine, an innovative feature that makes it remarkably simple to build and deploy custom apps. All of the productivity apps that ship with a Digium Phone are written with the JavaScript API that is used by the app engine. A BETA version of the phone firmware with app development tools is available at phones.digium.com, along with documentation for developing your custom apps.

Getting Started With Digium Phones

Get AsteriskDigium phones will work with any version of Asterisk. However, we’ve added some compelling features that are only available today in Asterisk 11 or in special branches of Asterisk 1.8 (seeCertified Asterisk) and Asterisk 10 (the -digiumphones branch). To take advantage of simple provisioning, integrated applications and the app engine, you will need to use one of these versions

Call or email for sales enquiries 

Categories
Elastix Support Knowledge Base

Setting up extensions in Elastix

Elastix Asterisk IPPXThis is a short video on the setting up of extensions on the Elastix Asterisk based IPPX.

 

Categories
Knowledge Base

VoIP – Per Call Bandwidth

These protocol header assumptions are used for the calculations:

  • 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers.
  • Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
  • 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.
  • 1 byte for the end-of-frame flag on MP and Frame Relay frames.
  • 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC).

Note: This table only contains calculations for the default voice payload

Codec Information Bandwidth Calculations
Codec & Bit Rate (Kbps) Codec Sample Size (Bytes) Codec Sample Interval (ms) Mean Opinion Score (MOS) Voice Payload Size (Bytes) Voice Payload Size (ms) Packets Per Second (PPS) Bandwidth MP or FRF.12 (Kbps) Bandwidth w/cRTP MP or FRF.12 (Kbps) Bandwidth Ethernet (Kbps)
G.711 (64 Kbps) 80 Bytes 10 ms 4.1 160 Bytes 20 ms 50 82.8 Kbps 67.6 Kbps 87.2 Kbps
G.729 (8 Kbps) 10 Bytes 10 ms 3.92 20 Bytes 20 ms 50 26.8 Kbps 11.6 Kbps 31.2 Kbps
G.723.1 (6.3 Kbps) 24 Bytes 30 ms 3.9 24 Bytes 30 ms 33.3 18.9 Kbps 8.8 Kbps 21.9 Kbps
G.723.1 (5.3 Kbps) 20 Bytes 30 ms 3.8 20 Bytes 30 ms 33.3 17.9 Kbps 7.7 Kbps 20.8 Kbps
G.726 (32 Kbps) 20 Bytes 5 ms 3.85 80 Bytes 20 ms 50 50.8 Kbps 35.6 Kbps 55.2 Kbps
G.726 (24 Kbps) 15 Bytes 5 ms 60 Bytes 20 ms 50 42.8 Kbps 27.6 Kbps 47.2 Kbps
G.728 (16 Kbps) 10 Bytes 5 ms 3.61 60 Bytes 30 ms 33.3 28.5 Kbps 18.4 Kbps 31.5 Kbps
G722_64k(64 Kbps) 80 Bytes 10 ms 4.13 160 Bytes 20 ms 50 82.8 Kbps 67.6Kbps 87.2 Kbps
ilbc_mode_20(15.2Kbps) 38 Bytes 20 ms NA 38 Bytes 20 ms 50 34.0Kbps 18.8 Kbps 38.4Kbps
ilbc_mode_30(13.33Kbps) 50 Bytes 30 ms NA 50 Bytes 30 ms 33.3 25.867 Kbps 15.73Kbps 28.8 Kbps

Explanation of Terms

Codec Bit Rate (Kbps) Based on the codec, this is the number of bits per second that need to be transmitted to deliver a voice call. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Size (Bytes) Based on the codec, this is the number of bytes captured by the Digital Signal Processor (DSP) at each codec sample interval. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Interval (ms) This is the sample interval at which the codec operates. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
MOS MOS is a system of grading the voice quality of telephone connections. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one (bad) to five (excellent). The scores are averaged to provide the MOS for the codec.
Voice Payload Size (Bytes) The voice payload size represents the number of bytes (or bits) that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size.
Voice Payload Size (ms) The voice payload size can also be represented in terms of the codec samples. For example, a G.729 voice payload size of 20 ms (two 10 ms codec samples) represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]
PPS PPS represents the number of packets that need to be transmitted every second in order to deliver the codec bit rate. For example, for a G.729 call with voice payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted every second [50 pps = (8 Kbps) / (160 bits per packet) ]

 

Bandwidth Calculation Formulas

These calculations are used:

  • Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
  • PPS = (codec bit rate) / (voice payload size)
  • Bandwidth = total packet size * PPS
Categories
Blog

Yealink T20P the new entry level IP phone

Yealink T20P is the entry level phone, of the Yealink VoIP desk phone range.

The Yealink T20P provides an entry level phone for the Yealink range of VoIP phones. The T20P boasts features beyond what you may expect from the lowest specification of the range. Ideal for use for the home or as a small office phone.

The T20P telephone offers power over ethernet (PoE), two SIP accounts, two line keys and a clear 2x 16 line LCD screen. A cost effective entry level enterprise IP phone with 2 lines.

Strong provisioning is in place for the Yealink range, making the models perfect for ITSP’s or large scale deployments.

Features

T20P-large

  • 2 SIP Accounts
  • 2 Line, 2×16 LCD Display
  • 2 Programmable Keys
  • 5 Hard Function Keys
  • Power over Ethernet (PoE)
  • 3 Way Conference Calls
  • Speakerphone
  • Call Hold, Waiting and Transfer.
  • Wall mountable
  • Compatible with a range of corded headsets:

 

RRP £69.99 +vat , (Contact us for volume and Special pricing)

Categories
Knowledge Base Support

24×7 Asterisk server monitoring with Nagios.

We offer an economical solution for end users and resellers to monitor their Asterisk and Linux servers.

Our platform monitors servers 24 hours a day 7 days a week. Hosted in a state of the art US based data centre with connections to major UK data centres and multiple connections to the internet.

We offer different levels of monitoring from simple uptime and email alerts to system load, disk space and channel usage with email and SMS notification. Web panel and firefox/Chrome plugin available to all levels to view system status.

The service is primarily aimed at Asterisk based IPPBX server but we can monitor other Linux based servers and Mitel systems as well. Our checks on Asterisk servers were customised by us to allow easy and secure deployment as we only require SSH access to make checks and this is secured by server keys. 

Nagios monitor screen

 

Service levels

Silver Level £10 setup – £2.50 per month £25.00 per year

  • Single Server, 4 services from list below & email alerts.
  • Ping test
  • SIP/IAX Peer availability
  • Asterisk channels
  • ISDN availability
  • Disk Space
  • System Load
  • Heartbeat Status
  • SIP/IAX2 registration status
  • Mitel SNMP Alarm status

Gold Level £10 setup per server – £5.00 per month £50.00 per year

  • Upto 2 Servers, 4 services per server, email and SMS alerts by subscription

In addition to the silver list:-

  • Asterisk Database status
  • Custom checks, (cost for design may be inured)

Additional options.

SMS alerts by arrangement, if using Gradwell Numbers and outbound we can integrate with the SMS API

Extra contact £5 setup

Extra server £10 setup £2.50 per month £25 per year

Extra service £5 setup £0.50 per month £5 per year

Partner options are available, Please contact us for details.  Pdf  download cymon 

Categories
FIrmware releases

Gigaset N300 IP, N300A IP, N510 IP PRO – Firmware update 12/2012 (version 075) released

New version of firmware released for N300 bases, Upgrade to this if on 072 firmware to fix instability issues

– Problem of instability, which occurred only very sporadically with version 72, and reset of base station after intensive usage solved

– Problem with call transfer of an external party to an external target behind Cisco Manager solved

– de telefoongids (Netherlands): online phonebook search is working again

– Security:

· Password is masked in VOIP Wizard, no longer visible in clear text

· PIN entry delayed if user repeatedly enters wrong PIN

– S68H handset: CLIP presentation is working again

– Blind Call Transfer problem solved with Telavox.se and Firmix.at

– URI dialling: Problem with added international/local area codes fixed

– Problem with consultation call and “Use Area Code Numbers for Calls via VoIP” setting fixed
Categories
FIrmware releases Knowledge Base

Yealink release V70 firmware for their T2X Sets

Yealink has announced the release of the latest Firmware V70 for its award winning IP phone SIP-T2X series.

The key feature of this new Firmware V70 is “M7”, also known as the “unified auto-provision template”. With Firmware V70, the configuration files and the deployment methods of T2X, T3X and VP530 have now been unified.

With the deployment of “M7”, end users now no longer need to maintain different templates of T2X, T3x or VP530. In other words, it lowers the learning curve and increases the business efficiency remarkably.

End users can easily convert their old templates of Yealink IP Phone T2X series and T3X series to “M7” through Yealink Configuration Conversion Tool (CCT). Firmware V70 is now available for download free of charge at www.yealink.com.

Download release notes here

 

Categories
Gateways

Redfone FoneBRIDGE

The foneBRIDGE is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black box appliance designed to streamline installation and enable redundant design of open source telephony based VoIP systems such as Asterisk®, Elastix, Trixbox, FreeSwitch and others.

Features

Failover and HA Enabled
• Flexible Configuration
• Highly reliable design
• Simple Install
• Form factor independent
• Single, Dual, Quad and Octal Port models

Applications

• T1/E1 PRI Trunk termination
• Legacy PBX-to-Asterisk integration
• Simple, single server installs
• Complex, HA Asterisk clusters
• Channel Bank connectivity
• Mixed telephony environments (T1 and E1)
• Blade Servers where PCI slots are not available

Download Datasheet here

Price excluding VAT:

  • Non ec Single e1 £430
  • Non ec Dual e1 £850
  • Non ec Quad e1 £1200
  • ec Single e1 £640
  • ec Dual e1 £1200
  • ec Quad e1 £1700
*ec = Hardware echo cancelation