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
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
Hourly rate Monday to Friday 9.00am – 5.30pm £45 Per hour or part hour Hourly rate Monday to Friday 5.30pm – 9.00am £62 Per hour or part hour
Day Rate Monday to Friday 9.00am – 5.30pm £360 Per day (This rate includes all travel expenses.)
RATES for the all other Areas Including Greater London
Day Rate Monday to Friday 9.00am – 5.30pm £460 Per day then £87 ph after 5.30pm
Day Rate Weekends 9.00am – 5.30pm £690 Per day then £130 ph after 5.30pm
(This rate includes travel expenses for sites upto 120 miles from Bath, For sites in excess of 120 additional charges will apply as well as costs for Hotel accommodation for sites over 180 miles (3 hours) from Bath.) Dependant on location half day rate is available on request. Please ask for details.
RATES for offsite configuration work, technical advice and Rmats access
Monday to Friday 9.00am – 5.30pm £45.00 first hour then £22.50 per 30 mins
Monday to Friday 5.30pm – 9.00am and W/E £62 first hour then £31.00 per 30 mins
All prices are based on a discount of 20% for payment within 14 days of attendance at site or completion of requested work. For payment terms in excess of 14 days please adjust prices to exclude 20% discount
All Prices exclude VAT. Vat will be charged at the prevailing rate at the time of purchase All prices are valid for 30 days from the date of this notice.
Elastix.org have announced the release of 2.4 stable.
Key changes are:
Changes in Elastix Framework:
The instalation of the Elastix system now its much cleaner.
The Migration to Privileged Scripts its completed. Now, there its no need to use commands such as /bin/touch, /bin/chmod, etc.
We improve readability on blackmin theme.
Fixed readout of FreePBX database password.
The internal jQuery was updated to 1.8.3 .
Some minor bug fixes for the Elastix Framework.
Changes in Elastix Addons :
Correction for Postgresql repo in ARM architecture.
Some minor bug fixes for Elastix-Addons.
Changes in Elastix Firstboot :
Make an update of password in manager.conf more robust in the case it falls out of sync with elastix.conf file
The Cancel option that used to appear in the dialog_password was removed, because if someone pressed, it no allows to continue configuring passwords. Now only appears the Cancel option after the firstboot if its necesary to change the password already seted.
Some minor bug fixes for Elastix-Firstboot.
Changes in Elastix Email_Admin :
Change of files owners for more security in the web path. Creation of new helper scripts (s
pamconfig,remotesmtp,mailman_conig,relayconfig).
Was made changes in the module email_account in order to better interaction at moment to create a new email account.
Some minor bug fixes for Elastix-Email_Admin.
Changes in Elastix Fax :
NEW MODULE Fax Queue.
Now errors are displayed when the fax job failed to submit and do not ignore them.
Remove useless code that could potentially error out the module.
Implementation of fax job cancelation.
Changes in Elastix PBX :
Add support and features to following phones: Elastix LXP200, Yealink model SIP-T38G, VP530 model, Alcatel Temporis IP800, Escene 620, Fanvil C62, Damall D3310 and Grandstream model GXV280.
Modified the way of displaying Reasons for Status in module weak keys.
In module Control Planel was made changes in function showChannel in order to fix bugs in wich the call made through a sip trunk have not been displayed in control panel.Some minor bug fixes for Elastix-PBX.
Changes in Elastix Security:
The instalation of this module now its much cleaner.
Change of files owners for more security int he path web path.
Some bug fixes for Elastix-Security.
Changes in Elastix System :
Reimplementation of GUI backup and restore operations on top of backupengine.
Add options to active o inactive services when reboot system in Process Status Applet.
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)
The ergonomically designed Yealink range of advanced SIP Handsets allows user to access the full capabilities of SIP-based VoIP platforms, delivering outstanding High Definition (HD) voice quality and advanced features, with PoE as standard that make communication much more efficient, productive and cost effective
T48GN £225
EXP40 £90
T46GN £170
T40 PN £86.00
T41PN £98.00
T42GN £120
T21P £67.99 +Vat
Yealink phones are compatible with many hosted VoIP services, IP PBXs and Traditional PBXs. The list is constantly growing as there are many new small hosted and IP PBX companies springing up all the time.
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
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)
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.
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
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Easy provisioning from Asterisk or AsteriskNOW
The ergonomically designed Yealink range of advanced SIP Handsets allows user to access the full capabilities of SIP-based VoIP platforms, delivering outstanding High Definition (HD) voice quality and advanced features, with PoE as standard that make communication much more efficient, productive and cost effective
