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Junior Linux Systems Administrator, Manchester, UK

DMC are an open source telecommunications services provider, based in Manchester, UK.

We are looking for a dynamic and pro-active person to join our technical team and help support our growing client base. You must be technically able to support the Linux operating system and troubleshoot hardware and network connectivity issues. An interest in telecommunications technologies is essential.

Full time, 37.5 hours per week, permanent.
Start date: ASAP
Closing date: 2013-08-16
Salary: £21,000 p.a.

Key responsibilities

  • Answer the help desk telephone line and manage support tickets in our request tracking system,
  • Give first line technical support as required and ensure that cases are escalated to the correct level in a timely fashion,
  • Regularly check, action and log requests into the help desk system for cases logged by email, web or voice mail,
  • Consistently provide excellent customer service.

Key Skills required

  • Knowledge of Linux operating system,
  • Experience of troubleshooting network problems,
  • Excellent customer service skills,
  • Confident, friendly and professional personality with a proactive and enthusiastic approach to work,
  • Strong organisation skills especially the ability to multitask and prioritise workload,
  • Ability to be self-motivated and work under minimal supervision,
  • Good problem solving abilities and analytical skills,
  • Full UK driving licence.

Direct applications only. Please send your CV and covering letter to careers@dmcip.com.

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Custom ring tones on the GXP 1450 / 2100

GXP 1450

Previous models of Grandstream IP phones (namely GXP 1200 / 2000 / 2010 & 2020) allowed Asterisk to select which ring tone to use by sending the Alert-Info SIP Header. This was done from the Asterisk dialplan using the SIPAddHeader application;

exten => s,n,Set(ALERT_INFO=Alert-Info:\;info=r2)
exten => s,n,SIPAddHeader(${ALERT_INFO})

… where r2 is the value of the ‘Distinctive Ring Tone; Custom ring tone 2, used if incoming caller ID is’ setting in the Advanced Settings page of the Web GUI.

However, the new range of GXP phones (i.e. 1450 / 2100) do not ring if the Alert-Info SIP Header is sent as above. We assume that this is a bug or intolerance in the SIP stack on the phone as the SIP INVITE from Asterisk to the phone that contains the custom header is never acknowledged. The solution is to format the Alert-Info header as follows;

Alert-Info: \;info=r2

For example;

exten => s,n,Set(ALERT_INFO=Alert-Info: <sip://127.0.0.1>\;info=r2)
exten => s,n,SIPAddHeader(${ALERT_INFO})

The good news is that the new format appears to work on the old models although you may need to upgrade the firmware on the new model range to version 1.0.1.66.

Posted in Asterisk, Grandstream | Leave a comment

HOWTO compile and install DAHDI 2.3 with OSLEC enabled

One common issue that persists through all telecoms related business is echo.

In the early days of the telephone, telecoms companies spent a lot of money going to great lengths trying to find a solution to the problem which occurred frequently when the traditional telephone network was first deployed.

We have to deal with this issue fairly frequently during consultancy on customer sites with their Asterisk PBX’s.

The issue almost always occurs with calls over PSTN/ISDN hardware and rarely appears to affect VoIP (probably due to upstream trunking providers use of echo cancellation software).

There are several approaches we can take:

  • We can enable DAHDI’s echo cancellation (“mg2″) and tweak various settings to try and make it work.
  • We can pay serious money to Digium et al. for a hardware module that will cancel the echo on the card itself.
  • We can compile DAHDI with OSLEC.

OSLECOpen Source Line Echo Canceller – is pretty much what it says on the tin. It’s an echo canceller, written by the legendary David Rowe which is far superior to other open source line echo cancellors. Unfortunately, as one of Digium’s revenue streams is the sale of hardware echo cancellation modules, getting it working feels like a bit of a hack and isn’t fantastically documented.

I thought we’d share how we did it…

Firstly, the guys at BSM Development deserve a pat on the back – their article on OSLEC is what got us going and really helped us get it up and working.

First we switch to our source directory:
dmc-pbx:~# cd /usr/src/
Let’s now download BSM Development’s lovely dahdi-oslec tar.gz with the modification you need to make to the plain DADHI source:

dmc-pbx:/usr/src# wget -c http://www.bsmdev.com/Downloads/dahdi-linux-oslec-2.3.0.1+2.3.0.tar.gz

Let’s now download the complete dahdi-linux tar.gz from Digium
dmc-pbx:/usr/src# wget -c http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz
Untar everything:
dmc-pbx:/usr/src# tar xvf dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz
dmc-pbx:/usr/src# tar xvf dahdi-linux-oslec-2.3.0.1+2.3.0.tar.gz

Now lets just copy the OSLEC source into the correct part of the DAHDI source
dmc-pbx:/usr/src# cp dahdi-linux-oslec-2.3.0.1+2.3.0/* dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi

From here you need to edit one line in this file:
dmc-pbx:/usr/src# sensible-editor /usr/src/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/Kbuild

Search for OSLEC. Uncomment, the first line that mentions it, save and exit.

Now just change the root of the source package…
dmc-pbx:/usr/src# cd dahdi-linux-complete-2.3.0.1+2.3.0
…and compile, install and install startup scripts:
dmc-pbx:/usr/src/dahdi-linux-complete-2.3.0.1+2.3.0# make all;make
install;make config

then reboot
dmc-pbx:~# reboot

When it’s back up, run
dmc-pbx:~# dahdi_cfg -v
to configure the dahdi interface

and then
dmc-pbx:~# sensible-editor /etc/dahdi/system.conf
and replace any mentions of “mg2″ with “oslec”.

After that you just need to restart dahdi
dmc-pbx:~# /etc/init.d/dahdi restart
and there you have it; a working OSLEC installation!

Fire up Asterisk and you are away and can start enjoying great quality echo cancellation without expensive hardware!

Do feel free to let us know how you get on in the comments! :)

Posted in Asterisk, Open Source | Tagged , , , , | 5 Comments

*CS: An Asterisk Call Server for business

Since 2006, DMC have been providing Asterisk based telephone systems to businesses of all shapes and sizes across the United Kingdom and European Union.

Over the last 2 years, we have developed *CS, a complete business telephone system that comprises an optimised Web GUI (Graphical User Interface) and dialplan.

The *CS dialplan embraces the best features of the Asterisk open source PBX and other complementary tools and technologies. The simplified multi-role GUI allows system administrators, supervisors and end-users to easily configure and manage the system and access call details and recordings.

A suite of CTI (Computer Telephony Integration) software tools enhance the solution further making *CS a complete telephone system for the SoHo, SME and Enterprise market.

Fully featured PBX

*CS is a fully featured PBX system that is not limited by user or line licences. The following features are available as standard;

  • ACD (Automatic Call Distribution)
  • Call Queues and Ring Groups
  • Caller Display / Alpha tagging
  • Music or Information-on-hold
  • Call Recording
  • Live call monitoring or snooping
  • Voice Mail
  • IVR (Interactive Voice Responders)
  • Conference Bridging

Lines and Numbers

* CS can connect to any type of Trunk Line whether it’s POTS, ISDN, VoIP, GSM or even Skype. Least Cost Routing ensures that outbound calls are routed over the lowest cost trunk line (e.g. international calls over VoIP, mobile calls over GSM).

We can provide Telephone Numbers for any geographic area code and non-georaphic and toll-free numbers. Numbers may be Ported from BT or Virgin Media to a VoIP service provider if your business moves away for the local exchange or simply wants to benefit from lower cost line rentals.

DDI numbers may be configured to route calls directly to Agent’s desk phones or to different Ring Groups and Queues.

Hot desking

Users may log in to any telephone handset or softphone with their unique extension number and secret PIN. Their incoming direct calls, voicemail and calls to queues or ring groups that they are a member of, will follow them wherever they are.

Call detail records (CDR’s) for calls they make or receive are attributed to their user account regardless of the telephone handset they are logged into.

Daily stats report

A comprehensive daily report of Key Performance Indicators (KPI) and usage statistics may be e-mailed to an administrator or manager.

Call recording and playback

All calls or calls from selected agents may be recorded on the server. An intuitive click-to-listen feature allows users to find personal call recordings through their web browser and play them back on their telephone handset.

Supervisors may be given extended access to this feature so they are able to review call recordings from a group of users.

Users and Supervisors may add notes and tags to call recordings so they may be easily retrieved at a later date.

Supervisor module

Supervisors may listen in, undetected, to live telephone calls from the GUI. A Whisper mode allows supervisors to whisper instructions or hints to an Agent as they handle a call. This feature is ideal for training new telephone operators.

Advanced management

*CS includes an optimised GUI for User, endpoint and queue management including the following features;

  • Bulk update of users, queues, incoming numbers and SIP accounts.
  • Check Queue / Ring group membership from the User profile page and dynamically add users to queues.
  • Endpoint Manager with customisable templates.

Contacts database

*CS can automatically log every number that called or was called to a central contacts database. The numbers may then be associated with a contact name, company name or third party data source so next time they call relevant information about the caller may be displayed on the telephone screen.

From the contacts database, numbers may be blacklisted or given higher or lower priority in the call queues.

The contacts database may also be used for click2dial.

Scalability and High Availability

*CS is available as an onsite PBX system or as a hosted service. The entire solution is scalable and will grow or shrink as your business changes.

Resillience and Redundancy may be built into the system to ensure a High Availablility service.

Further Information

For further information about *CS, please call 0161 850 4001 or e-mail sales at dmcip.com.

Posted in *CS, Asterisk | 1 Comment

DMC enters the Blogosphere

Re-entry

Since 2006, DMC have been providing open source communications solutions
to businesses across the UK and Europe and pioneering VoIP telephony so
we have always been at the leading edge of technology.

One thing that we didn’t get round to, at least until now, is blogging
about our endeavours in the Digital World.

So, here goes.

This blog will contain posts about our company – DMC, Asterisk, VoIP,
Open Source and other related topics. If you would like to receive
updates to the blog, then please follow our twitter account @dmcip.

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