This is a short video on the setting up of extensions on the Elastix Asterisk based IPPX.
Tag: Asterisk
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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 |
| 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
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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
- 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
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
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.
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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
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
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MPF ADSL
Advanced Broadband will give you the fastest download speed you can get from your exchange, a generous bandwidth limit and direct connection to the Gradwell VoIP network, making it a perfect partner for Internet telephony.
- No BT Line Rental
- £25 pm
- Generous 100Gb Peak Data Transfer per month
- FREE Thomson Wireless Router
- Voice prioritisation to support VoIP users
- Get the fastest speed your line can give you, up to 16Mbps
- 12 month contract, rolling monthly thereafter (30 day notice period)
Premier Broadband has all the benefits of our Advanced Broadband package, and more. 200Gb peak bandwidth and assured throughput mean you have the optimum amount of bandwidth for up to 10 concurrent Internet telephony calls.
- No BT Line Rental
- £50 pm
- Enhanced Care and Service Level Agreement
- 200Gb Peak Data Transfer per month
- FREE Thomson Wireless Router
- Assured throughput of at least 512Kb/s
- Upload speeds of up to 1Mbps and download speed of up to 16Mbps
- 12 month contract, rolling monthly thereafter (30 day notice period)
Premier Plus With Enhanced Care, a Service Level Agreement, fantastic upload speeds and assured throughput, Premier Plus Broadband is perfect for the largest of small businesses.
- No BT Line Rental
- £60 pm
- Enhanced Care and Service Level Agreement
- 200Gb Peak Data Transfer per month
- FREE Thomson Wireless Router
- Assured throughput of at least 1Mb/s
- Upload speeds of up to 2.5Mb/s and download speed of up to 16Mb/s
- 12 month contract, rolling monthly thereafter (30 day notice period)
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Digium G100/G200
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for Asterisk connectivity.
The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI interface, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations.
The Digium G100 VoIP Gateway includes a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. It is built to support TDM-to-SIP, SIP-to-TDM and SIP-to-SIP (transcoding) applications. In a TDM-to-SIP deployment the VoIP gateway significantly reduces operating costs by connecting a legacy business phone system with dynamic SIP trunking services. SIP-to-TDM deployments use the VoIP gateway to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.
Applications
Digium VoIP Gateways are flexible solutions that fit a variety of communications applications. The applications listed below represent some of the most widely used, today. The flexible configurtation options and standards-based connectivity mean Digium’s gateway appliances can support a wide range of custom applications.
Public Switched Telephone Network T1/E1/PRI to VoIP:
VoIP Provider to Legacy PBX:
Specifications
Interfaces / Connections
Benefits
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Features
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Price excluding VAT : G100 £820.00 G200 £1370.00
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Gigaset Dect test mode
Setting test mode on the Gigaset handsets can be very useful. Detailed here is how to set a handset into test mode and what the numbers then mean. Once set go off on a walk round your site to find dead spots. Then change the base position to get the best coverage
The above screen Shows RX power at 100% , Frequency is 3 , TimeSlot is 02, Basestation code is 78 and finally Bit error rate is 100% (This means 100% Good not 100% error rate)
A short document on Setting test mode on Gigaset dect handsets for site surveys is available for download here. This shows you how to enable it and what each of the numbers mean.