Categories
Handsets Products

RTX8630 IP DECT Multicell solution

The RTX8630 is complete cordless telephony solution offering a great scalability. The system is expandable and can grow with the business; from 1 to 40 bases and up to 200 users. The RXT8630 offers seamless call handover and repeater support. There is a choice of two different RTX DECT handsets, both with a high quality colour screen and wideband audio.

System features

  • Up to 200 users (200 handsets registered)
  • Scalable from 1 to 40 bases, with seamless handover
  • Allows up to 10 x simultaneous calls per base station (Expandable up to 400 calls per system)
  • Power over Ethernet (PoE): IEEE 802.3af Class 2
  • Range: upto 50m indoor and 300m outdoor per base
  • Repeaters supported
  • Bases are wall mountable using optional mounting kit (RTX8630Mount)
  • Choice of two handsets: RTX8430 and RTX8630
RTX8630 IP DECT Multicell solution
RTX8630 IP DECT Multicell solution
  • RTX8430 Entry level handset
    • 1.44″ TFT display
    • Local phone book with 50 entries (1 number/name)
    • Headset connector (3.5mm)
    • Battery life: Up to 8 hours talk time and up to 75 hours standby
  • RTX8630
    • 2″ TFT display
    • Local phone book with 100 entries (200 central entries)
    • Vibrate mode
    • Headset connector (3.5mm)
    • Battery life: Up to 18 hours talk time and up to 200 hours standby

Pricing:

RTX8630 Base RRP: £189.00+VAT

RTX8430 Handset RRP: £99.00+VAT

RTX8630 Handset RRP: £140.00+VAT

Call for availability and project pricing

Categories
Elastix Support Knowledge Base Support

CallerID in Elastix systems.

We get calls regularly on where to set the callerid in Elastix IPPBX systems. There are 3 places it can be entered for external caller ID and some can overide others but not all. so here is a simple explanation

Firstly, You can set it in the Extension, Trunk and the Route, In the route there is a check  box as well.

1. If you set it in the Trunk and no where else it will send this out as the CLI.

2. If you set it in the Trunk and in the Extension it will send out the Extension external cli as the CLI.

3. If you set it in the Route, Extension and the Trunk and don’t tick the override it will send the Routes CLI

4.  If you set it in the route and the Extension and the Trunk and tick the override it will send the extensions CLI

5.  If you set it in the Route and the Trunk and tick the override it will still send the Routes CLI

I hope that makes sense :-)

Categories
Elastix Support Knowledge Base

Elastix Custom Extensions.

This Short Video shows you how to setup custom extensions in Elastix and FreePBX

These can be used for calling mobiles or other external numbers that you want users to be able to dial as if they were extensions.

 

 

 

Categories
Cards

Digium Cards

digium_cards

Not only was Digium the first vendor of telephony interface cards built specifically for Asterisk, but it has always been the market leader, with over 50% of the world’s board business.

Analogue Cards

Digium analogue telephony cards are high-performance, highly reliable and cost-effective interfaces for POTS lines to your Asterisk solution. Multiple applications can be created to satisfy the business needs of any organization when using Digium analogue cards in concert with Asterisk software, the Linux® operating system and standard PC/server platforms.

Digital Cards

Digium’s super-reliable digital line cards connect Asterisk-based communication systems to T1, E1, J1 and ISDN-BRI interfaces.

Hybrid Cards

The Digium Hx8 Series are high-performance, cost-effective hybrid analogue and BRI telephony interface cards providing the capability to seamlessly integrate mixed-mode environments in a single device. Use the telephony card selector to identify the card that fits your requirements.

  • RoHS compliant
  • Manufactured in an ISO 9001:2001 certified facility in the United States
  • Maintain an MTBF greater than one (1) million hours
  • 5-year hardware warranty

 

For configuration and pricing please email or call us. We always want to speak to customers buying cards to make sure that it will be compatible with their server hardware.

Categories
Elastix Support Knowledge Base

Setting up timed call flow in Elastix

Screenshot from 2013-06-19 14:50:45If 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.

Categories
Case Studies

Multi-Site Multi-Country Asterisk network

UPDATE

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.

Globe

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 Openvox  Asterisk solutions.  .

xe2000-xe3000

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.

 

Categories
Knowledge Base

Elmeg IP290 Configuration

elmeg_290_large

 

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.

  1. The phones dont seem to support tftp download, Just http and https
  2. They dont support sub directories. So files must be in the root directory of your webserver.

Firstly you need to configure your dhcpd.conf

Add the following to the general section

option snom-setting code 66 = string;
option snom-bootfile code 67 = string;

Then the following to the subnet

class “snom-phones” {
match if substring(hardware,1,3) = 00:09:4f;
option snom-setting “https://SERVERIP”;
}

Theb the following are the two files you need to create .

elmegIP290.htm

<html>

<pre>

# example snom general setting file

# After each setting (before the colon) you can set a flag

# General language and time configuration parameter

language: English

web_language: English

timezone: GBR-0

time_server: pool.ntp.org

ntp_server: pool.ntp.org

date_us_format: off

time_24_format: on

user_host1: SERVERIP

user_host2: SERVERIP

tone_scheme: GBR

</pre>

</html>

elmegIP290-00094FMACADDR.htm

<html>

<pre>

# example snom specific setting file

# After each setting (before the colon) you can set a flag

user_name1: 345

user_pass1: PASSWORD

user_name1: 345

user_realname1: 345

user_host1: SERVERIP

user_srtp1: off

user_dp_str1: !([^#]%2b)#!sip:1@d!d

# You may add up to 4 (snom300/ 12 (snom320,snom360,snom370) accounts

# set 1st account to active outgoing identity

active_line: 1

</pre>

</html>

 

Categories
Knowledge Base Technical

Q931 Clear Cause definitions

This provides definitions to the Clear causes in Q931 messages.

Normal class
Cause #1 “unallocated (unassigned) number”
This cause indicates that the destination requested by the calling user cannot be reached
because, although the number is in a valid format, it is not currently assigned (allocated).

Cause #2 “no route to specified transit network”
This cause indicates that the equipment sending this cause has received a request to route
the call through a particular transit network which it does not recognise. The equipment
sending this cause does not recognise the transit network either because the transit network does not exist or because that particular transit network, while it does exist, does not service the equipment which is sending this cause.

This cause is supported on a network-dependent basis.

Cause #3 “no route to destination”
This cause indicates that the called user cannot be reached because the network through
which the call has been routed does not serve the destination desired.
This cause is supported on a network-dependent basis.

Cause #6 “channel unacceptable”
This cause indicates the channel most recently identified is not acceptable to the sending
entity for use in this call.

Cause #7 “call awarded and being delivered in an established channeP’
This cause indicates that the user has been awarded the incoming call, and that the incoming call is being connected to a channel already established to that user for similar calls (e.g. packet-mode X.25 virtual calls).

Cause #16 “normal call clearing”
This cause indicates that the call is being cleared because one of the users involved in the
call has requested that the call be cleared.
Under normal situations, the source of this cause is not the network.

Cause #17 “user busy”
This cause is used when the called user has indicated the inability to accept another call.
It is noted that the user equipment is compatible with the call.

Cause #18 “no user responding”
This cause is used when a user does not respond to a call establishment message with
either an alerting or connect indication within the prescribed period of time allocated
(defined in ETS 300 102-1 by the expiry of either timer T3 03 or T3 10).

Cause #19 “no answer from user (user alerted)”
This cause is used when a user has provided an alerting indication but has not provided a
connect indication within a prescribed period of time.
NOTE:This cause is not necessarily generated by ETS 300 102-1 procedures but may be
generated by internal network timers.

Cause #21 “call rejected”
This cause indicates that the equipment sending this cause does not wish to accept this
call, although it could have accepted the call because the equipment sending this cause is
neither busy nor incompatible.

Cause #22 “number changed”
This cause is returned to a calling user when the called party number indicated by the
calling user is no longer assigned. The new called party number may optionally be included in the diagnostic field. If a network does not support this capability, cause #1 “unallocated (unassigned) number” shall be used.

Cause #27 “destination out of order”
This cause indicates that the destination indicated by the user cannot be reached because
the interface to the destination is not functioning correctly. The term “not functioning
correctly” indicates that a signalling message was unable to be delivered to the remote user; e.g. a physical layer or data link layer failure at the remote user, user equipment off-line, etc.
Cause #29 “facility rejected”
This cause is returned when a facility requested by the user cannot be provided by
the network.

Cause #30 “response to status enquiry”
This cause is included in the STATUS message when the reason for generating the
STATUS message was the prior receipt of a STATUS ENQUIRY message.

Cause #31 “normal, unspecified”
This cause is used to report a normal event only when no other cause in the normal class
applies.
Resource unavailable class

Cause #34 “no circuit/channel available”
This cause indicates that there is no appropriate circuit/channel presently available to handle
the call.

Cause #38 “network out of order”
This cause indicates that the network is not functioning correctly and that the condition is
likely to last a relatively long period of time; e.g. immediately re-attempting the call is not
likely to be successful.

Cause #41 “temporary failure”
This cause indicates that the network is not functioning correctly and that the condition is not likely to last a long period of time; e.g. the user may wish to try another call attempt almost immediately.

Cause #42 “switching equipment congestion”
This cause indicates that the switching equipment generating this cause is experiencing a
period of high traffic.

Cause #43 “access information discarded”
This cause indicates that the network could not deliver access information to the remote
user as requested; i.e. a user-to-user information, low layer compatibility, high layer
compatibility, or subaddress as indicated in the diagnostic.
It is noted that the particular type of access information discarded is optionally included in
the diagnostic.

Cause #44 “requested circuit/channel not available”
This cause is returned when the circuit or channel indicated by the requesting entity cannot
be provided by the other side of the interface.

Cause #47 “resource unavailable, unspecified”
This cause is used to report a resource unavailable event only when no other cause in the
resource unavailable class applies.
Service or option not available class

Cause #49 “quality of service not available”
This cause is used to report that the requested quality of service, as defined in CCITT
Recommendation X.213, cannot be provided, (e.g. throughput or transit delay cannot be
supported).

Cause #50 “requested facility not subscribed”
This cause indicates that the requested supplementary service could not be provided by the network because the user has not completed the necessary administrative arrangements with its supporting network.

Cause #57 “bearer capability not authorised”
This cause indicates that the user has requested a bearer capability which is implemented
by the equipment which generated this cause but the user is not authorised to use.

Cause #58 “bearer capability not presently available”
This cause indicates that he user has requested a bearer capability which is implemented by the equipment which generated this cause but which is not available at this time.

Cause #63 “service or option not available “unspeciried”
This cause is used to report a service or option not available event only when no other
cause in the service or option not available class applies. Service or option not implemented class

Cause #65 “bearer capability not implemented”
This cause indicates that the equipment sending this cause does not support the bearer
capability requested.

Cause #66 “channel type not implemented”
This cause indicates that the equipment sending this cause does not support the channel
type requested.

Cause #69 “requested facility not implemented”
This cause indicates that the equipment sending this cause does not support the requested
supplementary service.

Cause #70 “only restricted digital information bearer capability is available”
This cause indicates that one equipment has requested an unrestricted bearer service but
that the equipment sending this cause only supports the restricted version of the requested
bearer capability.

Cause #79 “service or option not implemented, unspecified”

This cause is used to report a service or option not implemented event only when no other
cause in the service or option not implemented class applies.
Invalid message (e.g. parameter out of range) class

Cause #81 “invalid call reference value”
This cause indicates that the equipment sending this cause has received a message with a
call reference which is not currently in use on the usernetwork interface.

Cause #82 9dentified channel does not exist”
This cause indicates that the equipment sending this cause has received a request to use a
channel not activated on the interface for a call. For example, if a user has subscribed to
those channels on a primary rate interface numbered from 1 to 12 and the user equipment
or the network attempts to use channels 13 through 23, this cause is generated.

Cause #83 “a suspended call exists, but this call identity does noV
This cause indicates that a call resume has been attempted with a call identity which differs from that in use for any presently suspended call(s).

Cause #84 “call identity in use”
This cause indicates that the network has received a call suspend request. The call suspend
request contained a call identity (including the null call identity) which is already in use for a suspended call within the domain of interfaces over which the call might be resumed.

Cause #85 “no call suspended”
This cause indicates that the network has received a call resume request. The call resume
request contained a Call identity information element which presently does not indicate any
suspended call within the domain of interfaces over which calls may be resumed.

Cause #86 “call having the requested call identity has been cleared”
This cause indicates that the network has received a call resume request. The call resume
request contained a Call identity information element which once indicated a suspended
call; however, that suspended call was cleared while suspended (either by network timeout
or by the remote user).

Cause #88 “incompatible destination”
This cause indicates that the equipment sending this cause has received a request to
establish a call which has low layer compatibility, high layer compatibility, or other
compatibility attributes (e.g. data rate) which cannot be accommodated.

Cause #90 “destination address missing or incomplete” This cause indicates that he
called user cannot be reached because the called party number is not in a valid format or is
not complete.

Cause #91 “invalid transit network selection”
This cause indicates that a transit network identification was received which is of an
incorrect format as defined in Annex C.

Cause #95 “invalid message, unspecified”
This cause is used to report an invalid message event only when no other cause in the
invalid message class applies.
Protocol error (e.g. unknown message) class

Cause #96 “mandatory information element is missing”
This cause indicates that the equipment sending this cause has received a message which
is missing an information element which must be present in the message before that
message can be processed.

Cause “97 “message type non-existent or not implemented”
This cause indicates that the equipment sending this cause has received a message with a
message type it does not recognise either because this is a message not defined or defined
but not implemented by the equipment sending this cause.

Cause #98 “message not compatible with call state or messages type
non-existent or not implemented”
This cause indicates that the equipment sending this cause has received a message such
that the procedures do not indicate that this is a permissible message to receive while in the call state, or a STATUS message was received indicating an incompatible call state.

Cause #99 “information element non-existent or not implemented”
This cause indicates that the equipment sending this cause has received a message which
includes information elements not recognised because the information element identifier is not defined or it is defined but not implemented by the equipment sending the cause.
However, the information element is not required to be present in the message in order for
the equipment sending the cause to process the message.

Cause #100 “invalid information element contents”
This cause indicates that the equipment sending this cause has received an information
element which it has implemented; however, one or more of the fields in the information
element are coded in such a way which has not been implemented by the equipment
sending this cause.

Cause #101 “message not compatible with call state”
This cause indicates that a message has been received which is incompatible with the call
state.

Cause #111 “protocol error, unspecified”
This cause is used to report a protocol error event only when no other cause in the protocol
error class applies.Interworking class

Cause #127 interworking, unspecified”
This cause indicates that there has been interworking with a network which does not provide causes for actions it takes; thus, the precise cause for a message which is being sent cannot be ascertained.

Categories
Blog

A new patent troll.

SIP Trunking

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.

More to follow on this I’m sure….

Categories
IPPBXs Products Services

Multi User Hosted PBX

Use the Internet to make calls – it’s simple, cost effective, and perfect for small businesses and call centres.

Voice over Internet telephony reduces telephony bills; connects mobile, remote and office workers; and gives a consolidated impression of your business – same number no matter where employees are located.

From 2 to 10 people wanting to stay in touch, Our feature-rich packages can offer a local or International number from which to operate – or you can bring your old number with you (number porting). There are also a wide range of add-ons that offer inclusive minutes. We can go through setting it up for you or you can do it yourself. To signup follow this link or call us on 01225580025

  • Multi User VoIP £8.00 per month
  • 4000 UK landline minutes £ 20.00 per month Lower amounts available
  • 4000 UK & International landline minutes £25.00 per month
  • 500 UK mobile minutes £30.00 per month

*Prices exclude VAT.

Key features

An online customer control panel allows you to manage your own account, and you can expand the system when ever you need to. All you need to get started is broadband, a router, and an adapter or a VoIP phone.

  • Make immediate savings: Free internal calls. Competitively priced calls and inclusive landline and mobile minutes package add-ons.
  • Quick and easy to set up: No difficult installations.
  • Excellent call quality: With no compromising on functionality.
  • Never be out of touch: Call forwarding available.
  • Keep your old number: Seamless transition with ‘number porting’.
  • Global presence: International numbers available.
  • Stay in control: Online customer administration, call logs and invoicing.
  • Voicemail and voicemail notification
  • Call forwarding to any number including mobiles
  • Online contacts directory, call logs and invoicing
  • Customised CLI (caller line identity)
  • Time of day routing

 

Features

  • Set up £4.99
  • Monthly £8.00
  • Included phone number UK and International*
  • Concurrent calls per number 2
  • Internal extensions 10
  • Call packages FREE VoIP-to-VoIP
  • 999 Emergency Services access YES
  • Minimum contract length 12 months
  • Voicemail YES
  • Voicemail notification SMS or Email
  • Call forwarding YES
  • Codecs supported G729a, G711u, G711a
  • Online call logs and invoicing YES
  • Online contact directory YES
  • Customised CLI (caller line ID) YES
  • Time of day routing YES
  • Audio call conferencing YES
  • IVR/Auto-attendant YES
  • Music on hold YES
  • Hunt call groups YES
  • 4000 UK landline minutes add-on £20.00 per month
  • 4000 UK & Int. min. add-on £25.00 per month
  • 500 UK Mobile minutes add-on £30.00 per month
  • Additional Number^^ £3.00 per month
  • Personal Number** £10 setup, £10 per month

* Surcharges apply for International numbering.

^ Subject to fair use check at 4000 minutes per month.

^^ Can only point at a Gradwell VoIP number on the same account, they can act as a mainline phone number however they must take an existing route. Please note that they cannot be used to increase concurrent calls.

** Can only point directly at an extension number. You cannot direct these numbers towards hunt groups, call queues or any other type of functionality. Does not provide an additional line.

All phone services (inc Unlimited packages) are subject to Terms and Conditions and standard call charges. All prices exclude VAT. Range of hardware and accessories available.