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Gradwell Approved Professional

A Gradwell Approved Professional (GAP) has a core set of knowledge required to succeed with the Gradwell product set and systems. To become a Gradwell Approved Professional partners must pass an examination and show an in depth knowledge of Gradwell Products and Voip Support.

This means you can be confident that they can:

  • Achieve consistent roll outs of reliable business telephony services.
  • Confidently resolve end user technical issues within an acceptable time frame.

To signup to Gradwell services with us CLICK HERE

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Peripherals Products

The Algo 8180 SIP Audio Alerter

The 8180 SIP Audio Alerter is a loud ringing, paging and intercom device for use with SIP telephone systems. It has two main uses. The first is as a loud ringer for use in places such as warehouses. The second use is as a paging and intercom system. It can operate in both these modes at the same time by appearing as two different extensions on your phone system at the same time.

When registered with a SIP server, one endpoint will play an audio file from internal memory upon ring detection. The second endpoint will auto-answer for voice paging or full intercom (two way audio).

Equipped with a high efficiency integrated amplifier and tuned high quality loudspeaker, the 8180 is typically eight times louder than a telephone speaker. Several audio files are pre-loaded into the 8180 internal memory for ring sounds but users may also record or upload custom audio files, music, sound effects, or voice announcements.

Ambient noise monitoring

The advanced features of the 8180 include SoundSureTM technology which automatically adjusts loud ring and loudspeaker volume to compensate for background ambient noise. Ideal for variable noise environments (restaurants, workshops, classrooms, etc.), SoundSureTM ensures that ringing or paging is always heard but not unnecessarily loud.

Applications / Usage scenario

  • Loud Ringer in noisy or variable noise environments (classroom, restaurant, machine shop)
  • Voice Paging (warehouse, workshop)
  • Outdoor ringing or paging
  • Multi-cast wide area notification

Easy to Install

  • Network managed SIP endpoint
  • Configuration is possible using the feature buttons or web interface
  • PoE eliminates local power supply

Features

  • Voice Paging with talkback capability
  • High efficiency and high output wideband speaker
  • Multicast receive or broadcast capability
  • Outputs for external speaker, slave amplifier, or visual alerter
  • Dual purpose loud ringing and/or talkback voice paging
  • Significantly louder (eight to twenty times) than typical telephones
  • Low frequency tones outperform traditional shrill electronic ringers
  • SoundSureTM ambient noise compensation adjusts output for noise level
  • Multicasting capability for wide area notification

Customisable

  • Pre-loaded with several ring tones including bell, chime, gong, buzzer, warble, and dogs
  • Supports custom uploaded WAV files or recorded messages
  • Selectable/customisable alert tones or announcements

Please contact for pricing and avalibility

Categories
Case Studies

VoIP User Connects 14 Million BT Customers To iNUM

VoipUser.org, the largest open-source VoIP network, has announced the interconnection of British Telecom 0870 range numbers to the VoIP peering iNUM network.

“VoIP remains restricted by the lack of interconnection between networks,” said Dean Elwood, founder of VoIP User. “By interconnecting to the BT 0870 range, we have enabled cross-network calling for 14 million users which is free of charge at off-peak times. What we have achieved today is the extension of our service into a new network of over a million VoIP users”.

British Telecom Plc, the UK’s largest telecom provider, announced that from 16th January 2009 it will make 0870 numbers free to call to subscribers on their Anytime call plan.

John Peter, managing Director of BT’s Consumer Business, said in a Press Release on Thursday “All of our 14 million Anytime customers have free calls included in their package and now all 14 million have free calls to 0870 at times that fit with their calling plan, which is something not offered by any of our competitors.”

Ian Plain, of technical consultancy cyber-cottage.co.uk, who architected the system for VoIP User, said “we created a mapping service for the 0870 iNUM interconnect. The system is accessed by users dialling 08700 68 58 48 from a BT landline and entering an iNUM number to contact. Details can also be stored meaning that the caller only ever has to enter the iNUM number to contact once.”

Tjardick van der Kraan, co-founder of VoIP User, said “The value add for customers of our services is the ability to call iNUM +883 range numbers, the so-called ‘Earth Area Code’, free of charge at off-peak times from a BT landline. This effectively connects British Telecom customers to Worldwide iNUM VoIP customers with no per minute charge.”

The iNUM Peering Network, run by Belgium based Voxbone S.A. operates using a standards based mechanism for traffic exchange between telecommunications networks.

Traditional dial codes are tied to a particular country or city. iNUM numbers avoid this limitation and users can therefore keep their number wherever they go in the world, being reachable on the same number for life.

Further details of the service are available at : VoIP iNUM

About VoIP User

VoIP User is located in the United Kingdom and provides hosted telephony services funded by the same community that uses these services. VoIP User’s business model is unique, based on inbound PSTN traffic revenue being used to purchase pre-wholesale bulk outbound routing. VoIP User

About Voxbone

Headquartered in Brussels, Belgium, Voxbone provides worldwide DID numbers and free phone numbers over a private intercontinental VoIP. The switchless architecture of the Voxbone network enables customers to realize the benefits of IP communications by rapidly deploying new services with local presence and simultaneously reducing costs. Voxbone

Categories
Blog

Sip attacks and what Data-centre operators can do

More and more we are seeing SIP brute force attacks from hosted servers. These aren’t really hacking attempts as in many cases they just try the same user and password Millions of times.

We block these attacks automatically on out servers but that doesn’t stop the traffic, They carry on till we get the Data-centre to shut-down the server. Which can be difficult.

We have seen attacks recently from Germany on the increase in particular one data-center based in Berlin. The staff here DO NOT respond in a timely manner to abuse reports and it has taken upto 4 days to get the servers shut down. They claim that if they shut the server down it infringes their customers rights. we have pointed out to them that they clearly state in their AUP(below) that the server cannot be used for this purpose.

“a. Utilize the Services to cause denial of service attacks against ***** or other network hosts or Internet users or to otherwise degrade or impair the operation of ******s servers and facilities or the servers and facilities of other network hosts or Internet users;”

And if they do, the server will be shut down. So why don’t some data-centres respond? This is an interesting one, At the data-centre in Berlin the attacks always started round the same time on a Sunday Morning on a clean dedicated server and had all the finger prints of  human not Bot activity, as with bots we see them try a few times and then give up. With these attacks they are started and keep going even when we are dropping all the packets, in this case the Bot moves on.

When the attack is finally stopped we get no explanation or in the case of it taking many days to stop , No apologies or explanations for taking so long.

I do think its time for Data-centre operators to take their AUP’s seriously and enforce what they say.

Categories
Case Studies Technical

Man In The Middle

Man In The Middle Server, What does he mean I hear you ask?

Well its the best term for it, as it describes what it is and does. Basicly we have an AsteriskServer sitting in between the incoming lines and the main  PBX. Idealy We would have liked to replace the existing PBX but in some cases this just isnt possible and we have to accept this.

So why may we need to do this, Well we have used this solution to allow customers to have follow the sun  support call centers.  Calls arrive to a dedicated DDI number that is for support in that country and depending on the time of day it is routed to the call center that is open in another part of the world over a voip network. This is done by Asterisk checking the called number of all calls and if a match is made the call is passed off to Asterisk to handle the call in its dialplan, all other unmatched calls are passed on to the main system.

We have also used this method of connection to play prompts to callers before the call is sent to the main system. Calls in this case can be identified by CallerID name or number. So for example calls from the BT operator, International payphones or even just a certain CallerID Number are played a specific message and then either forwarded on or passed on to a IVR (Interactive Voice Response menu) to be handled in a specific manner.

A another often used reason for this type of connection is when migrating from a legacy pbx to and Asterisk server. The line connections are the same and the call flow is controlled by the dialplan, Routing calls on teh Asterisk system to its dialplan and routing calls for extensions still on the old system to it.

Once all extensions are migrated to the new system the old system can be turned off and removed with no interuption to the users on the new system.

Categories
Knowledge Base

Asterisk Music on Hold

Music on hold is always an issue, We get asked many times “can we put such and such mp3 on as hold music” Well the answer is always the same, Yes and no.

Yes in that we can convert any mp3 to be played as MOH as long as you have the relevent PRS licence.

So the answer is then normaly No we wont do it then, Just look at the costs detailed here and you will see why.

So why are we mentioning it here.

Well Asterisk has Music files loaded for MOH and it was always assumed these were licenced correctly and required no additional licence or agreement. Well it seems that some countries are challenging this.

See the following from Digium..

Open Source Asterisk has had for quite some time the ability to play Music On Hold (MOH) to callers as an optionally configured call feature.  Of course, as soon as the code had the ability to play music, there was a general request and obvious concept that Asterisk should include a few default music-on-hold files.  At that point, several people within Digium looked around at the possible files we could use, but all of them had some type of license issues, which is understandable.  We found a company which sold rights to music, and we discussed in specific, painstaking detail what we wanted to do with the files and how they were going to be used.  They agreed that we could do what we wanted and distribute the files with Asterisk and that they were able to provide to us the appropriate license, so we paid our fee and proceeded to pick some likely music.  We then included them in Asterisk in the hopes that the community would find them useful as part of the system without having to search out selections which complied with various copyright issues.  This was a good-faith gesture on our part, and we had a quite reasonable expectation that the vendor from whom we purchased the license was authorized to provide to us a global right-to-use and redistribution capability to the Asterisk community for these sound files.

Apparently, that assumption is now being questioned.  In some nations (Australia and France, to pick two that have been brought to our attention) there are some who are claiming that we do not have the rights outlined above, and that our users therefore are in a similar situation where they may be in violation of license terms.

In the interests of space here I will not outline the exact organizations, laws, and claims in question.  Suffice it to say they are complex and unclear with a broad range of possible interpretations. Currently, at least two organizations disagree that we are complying with a set of license terms.  This is very far outside of Digium’s ability or interest to manage, nor do we wish to become involved in the protracted series of legal proceedings required to sort out this licensing issue.  So we have chosen another path that is more clear to us: we will eliminate the files of questionable license from Asterisk, and replace them with music that has  clearly defined and more acceptable licensing terms which are compatible with both the Asterisk license, and with any reasonable redistribution methods that might be used by others who re-package Asterisk.

So how can we be assured this won’t happen again?  The new music we’ve included is under the Creative Commons 2.5 license – which quite frankly didn’t have much of a following for media back when we first were looking for a set of MOH files.  Certainly, the selection of good-quality music files that would suffice did not exist in an easy-to-obtain fashion, or we would have gone this route in the first place.  Hopefully you’ll like the new music on hold, and will be customers of the artists who have so graciously given their work out under such a reasonable license.   We found the new music on Opsound for those of you who are looking for an even wider selection of freely available music.

We apologize for putting people through this aggravation – we sincerely wish that the recording industry would standardize license terms and avoid treating customers like enemies.  Perhaps there is a silver lining here –  this may be a good opportunity for you to freshen up your hold music – maybe “Calm River” was getting under people’s skin after the thousandth time you put them on hold.

The Gory Tech Details:

The new files can be found here: http://downloads.asterisk.org/pub/telephony/sounds/

The new files have names that contain “opsound” replacing the term “freeplay” in the file names.  It’s really that simple.  We’ve removed the old “freeplay” files and symlinked the old names to the new “opsound” versions, just in case.  The contents of the .tar archives are different, but Asterisk should “just work” when the file contents are put in place unless you’ve extensively modified your music on hold configurations.

If you have an existing Asterisk system it is our suggestion that you delete any and all copies of the existing “freeplay” music on hold set on your system(s) and replace them with the new “opsound” module sets.  The good news is that this is very simple – very little configuration is required on your system – just a simple file copy and then restart Asterisk to see the new files.

What happens if you don’t update your music on hold?  The state of the FreePlay license currently is in question, and even if it is determined that the Freeplay files were not provided under the necessary license terms,  it’s unlikely that enforcement actions will be brought against end-users for using unlicensed hold music.  However, we would encourage all administrators to update their instances of Asterisk to the new sound files.  This is really a distasteful process for all of us, and in order to protect ourselves from any future liability we’re going to suggest that everyone remove the older files entirely.

Categories
Knowledge Base

General Configuration Guide Skype for SIP and Asterisk

 

If you are new to SIP, Asterisk is a useful, open-source (GPL) platform with which to test and experiment with the Skype for SIP. This is a guide on how to install Skype for SIP on a system agnostic or “vanilla” Asterisk server.

 

To install Asterisk on your server, please see the Digium documentation here http://www.asterisk.org.

 

This configuration guide is based on Debian Linux (Lenny 64bit). With a basic installation of Debian you can install Asterisk by issuing the following APT command at the command line:-

apt-get install asterisk

 

 

Configuration Files for Vanilla Asterisk

 

In configuring Skype for SIP on a vanilla Asterisk system we are primarily concerned with two configuration files:-

 

  1. sip.conf (located in the /etc/asterisk/ directory)
    The sip.conf file holds the registration details for the Skype for SIP channel
  2. extensions.conf (located in the /etc/asterisk/ directory)The extensions.conf holds the dial plan telling Asterisk what to do with incoming and outgoing calls.-

 

Let’s do a walkthrough of the configuration steps.

 

Configuring the sip.conf File

 

Step 1

 

The sip.conf file has two sections that need to be completed. The “General” section (denoted in the file with the [general] heading) and peer section denoted in the file with the [peers] heading.

 

In the General section we need to add a “register” line. This tells Asterisk to register with Skype at the Skype local point of presence.

 

Add the following, under the “[general]” section in the file, substituting your 9905xxxx number and password with your actual credentials for the Skype for SIP profile you wish to use. Your SIP Profile details can be found in the Skype Business Control Panel (BCP):-

 

register => 99051000xxxxxx: PaSsW0rD@sip.skype.com /99051000xxxxxx

 

Step 2

To ensure that we also receive the callerID from Skype clients we also should add:-

 

trustrpid = no

sendrpid = yes

 

 

Step 3

Next, we add a section for the peer, in the “[peers]” section of the sip.conf file. Again we substitute the 9905xxxxx number and password with the SIP Profile credentials from the Skype Business Control Panel (BCP):-

 

[99051000xxxxxx]

type = peer

username = 99051000xxxxxx

fromdomain = sip.skype.com

fromuser = 99051000xxxxxx

realm = sip.skype.com

host = sip.skype.com

dtmfmode = rfc2833

secret = PaSsW0rD

nat = no ;This should be set to reflect your network NAT configuration

canreinvite = no

insecure = invite

qualify = yes

disallow = all

allow = alaw

allow = ulaw

;allow = g729 ; Uncomment this if you have G729 licences

amaflags = default

trustrpid = no

sendrpid = yes

context = skype_in

 

Please Note:

If your Asterisk PBX is behind a NAT device, you should set “nat = yes” in this section.

 

If your Asterisk PBX has a dedicated internet IP address, set this to “nat = no”.

 

Step 4

After setting these changes, reload the Asterisk’s SIP module by typing:-

 

asterisk -rx “reload chan_sip.so”

 

…….at the command line.

 

Step 5

After the SIP Module has reloaded enter asterisk -rx “sip show peers” at the command line, which should return:

 

pbx*CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status

99051000xxxxxx/99051000xx 193.120.218.68 5060 OK (52 ms)

 

Then enter asterisk -rx sip show registry” which should return:

 

pbx*CLI> sip show registry

Host Username Refresh State Reg.Time

sip.skype.com:5060 99051000xxxx 105 Registered day, dd mmm yyyy hh:mm:ss

 

If you see output similar to the above, then you are registered to the Skype SIP gateway and ready to make and receive calls.

 

We now need to setup the extensions.conf so that we have a dialplan setup and Asterisk knows how to deal with incoming and outgoing calls.

 

Configuring the extensions.conf File

 

The extensions.conf file requires a “context” and an “extension” to be added for incoming Skype calls, plus an extension to be added to the context that users use for outgoing calls.

 

Incoming “context”

 

Add the following lines to the [context] section of extensions.conf, substituting 9905xxxxxxx with the 9905 number for the SIP Profile. Again you can find the details of your Skype SIP Profiles in the Skype BCP:-

 

[skype_in]

exten => 99051xxxxxxxx,Noop(${CALLERID(name)} , ${CALLERID(num)})

exten => 99051xxxxxxxx,n,Dial(SIP/100,30,t,r)

exten => 99051xxxxxxxx,n,voicemail(100|u)

 

This is a simple “vanilla” context that shows us the callerID name and number, dials extension 100 for 30 seconds and finally, if unanswered, goes to voicemail. This sequence will need to be amended to suit your requirements. If you are planning on having many SIP Profiles or Online Numbers that all need to end up at the same destination, or the destination is decided by the Skype Business Account that the online number is registered against, a more complicated Dialplan can be used. For example:-

 

[skype_in]

exten => 99051xxxxxxxx,1,Noop(${CALLERID(name)} , ${CALLERID(num)})

exten => 99051xxxxxxxx,n,Queue(sfs|r|||40)

exten => 99051xxxxxxxx,n,voicemail(100|u)

 

 

Outgoing “Context”

 

The outgoing context must be included in the context for your user’s phones. Usual security measures apply. Do not include this in a context for incoming calls.

 

[skype_out]

 

exten => _90Z.,1,Set(CALLERID(num)= 99051xxxxxxxx)

exten => _90Z.,n,Dial(SIP/0044${EXTEN:2}@99051xxxxxxxx)

 

exten => _900.,1,Set(CALLERID(num)= 99051xxxxxxxx)

exten => _900.,n,Dial(SIP/${EXTEN:1}@99051xxxxxxxx)

 

 

In the sip.conf add the following to create user 100

 

[100]

secret=secret

mailbox=100

callerid=”myskypetrunk” <100>

type=friend

host=dynamic

context=international

;nat=no

nat=yes

canreinvite=no

dtmfmode=rfc2833

pickupgroup=1

callgroup=1

subscribecontext=default

notifyringing=yes

disallow=all

;allow=alaw

allow=ulaw

allow=gsm

 

in the extensiosn.conf add the following to the default context

 

exten => _XXX,1,Dial(SIP/${EXTEN},20)

 

Also create a context called international

 

[international]

include => default

include => skype_out

 

Categories
Case Studies

Asterisk install in a campus style school

In late 2005 We we approached to replace an aging Panasonic system for a British Public School.
The driving force for moving to voip was that the site was spread over a wide area and different buildings and to provide telephones to the remote buildings would prove too expensive.
The system was replaced with a central Asterisk server with nearly 80 extensions. The core LAN was upgraded to Netgear Layer 3 switches with Powerdsine POE midspans.
The remote buildings added extra complexity as one was on the other side of a public road.

To overcome this, this building was connected to the main site via a “Point to Point” Wifi link. The other building was closer and could be connected via a Fibre Link between the it and the main building.

Campus Site

The system was configured to use account codes in public areas, These handsets can only make emergency or internal calls unless a validated account code is entered.
The handsets used were a mix of Aastra 480i and 9133i because of build quality, reliability and BLF support.

The system has now been in place for over 2 and a half years now and in that time the only faults have been either ISDN failing or cable faults. This is a great demonstration of the reliabilty of Asterisk and Aastra handsets and voip in general.

UPDATE

We have since writing this post updated the system on new hardware as part of a refresh. They are now running on a newer version of Asterisk and have more buildings connected to the network.

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Products

Handsets,Gateways and Cards

We supply VoIP Hardware, Phone Systems, IP Phones & VoIP Equipment from all Major Manufacturers.

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Digium Openvox Patton
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Categories
Case Studies

Solicitors Group

We were approached to provide a Voip solution for a London Solicitors. They were moving from a Single office to two offices in different parts of the city.

At the Primary office we installed an Asterisk server that was connected to the ISDN30e link where all DDI numbers were delivered for both sites. This site also had its own VOip connection as well

At the main site the operators were based using the Asternic FlashOsPanel for displaying extension status for both sites.

A key feature of the system was to provide a flexible system for out of hours callers to contact the duty Solicitor. This was provided so that callers on calling out of hours  can leave a message for the next day or press an option to contact the Duty solicitor. The Duty solicitor can call into the system and change the contact number at will if for example they are busy with a client.

A SIP trunk connected both sites together over a 20Meg link that was used for both voice and data.

The new sites were not in the exchange area of the original phone number so this was ported to Gradwell.net so that it could be delivered via an IAX2 connection to the relevent server.

The handsets used were a mixture of Aastra 53i and 55i with an xml appliction to set DND status in the Flash ops panel and light the light on the handset.