This is a short video on the setting up of extensions on the Elastix Asterisk based IPPX.
Tag: freePBX
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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)
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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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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
|
Features
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Price excluding VAT : G100 £820.00 G200 £1370.00
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Restaurant Booking Solution.
We have been working with a client on a Hosted restaurant booking solution, providing the CTI and call tracking systems. This was complicated by the simple fact that the booking system was a closed system by another supplier.
The system works by the restaurant diverting their line to a DDI number we provide that sends the call to the system with the destination matching a defined number for that restaurant when the call enters the system the relevant settings are looked up in a database and audio message file, IVR options and CallerID name are set and the call is passed to the IVR. The caller then chooses their preferred option, The call can be depending of the time of day be passed to the call centre for a booking to be taken on the restaurants behalf or the call is passed to the restaurant where in many a Hosted Gigaset Dect handset is provided for them to take the booking or call the call centre for free.
The system has changed and evolved over time and by using Asterisk has meant that we can accommodate most requests for changes, Most recently we updated the statistics package to Asternic Pro
Statistics. This has allowed much more detailed reports to be created for queues and agents.
The calls are delivered to the platform over a EFM circuit from Gradwell.net providing quality and reliability combined with increased capacity over the original ISDN30 circuit.
For disaster recovery we provide a backup system in a data-centre that is kept in sync with the office system so in the case of power outage or system failure calls can be diverted to this system and calls take on mobile phones. Switching to back-up system is completed by the single click of a button on a web-page that instigates the diversion of the lines and starts the backup system automatically.
Currently we are migrating the database services off to a separate VMware server with 3 VMs, one for each of the core web or mysql servers. This will allow the service to scale as there are now over 1 million records per datatbase and it is showing no sign of slowing.
The current firmware in the Digium G series gateways have a quirk that if they don’t receive caller ID name they move the caller Id number to be the Caller Id name but don’t leave the Caller Id number in place. The relies on you setting “trustrpid=yes” in teh sip trunk configuration.
We have produced a short document on settings for using the gateway with any freePBX based asterisk solution. It can be downloaded here
We were recently approached by a Bristol IT company to replace their ageing Avaya system. The proposed solution was an Elastix 2.3 solution running on Vmware 5.1, with their ISDN30e line connected via a Digium G100 gateway.
A key requirement was the ability to recharge usage to tenants in the building and replace an aging and expensive Oak call logger, This was simple with the Elastix solution as this option is included free of charge in the system and just required the uploading of a rates table.
The customer decided on Yealink T28 handsets for the office, utilising the BLF to have visibility of who is engaged on calls.
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Outdoor pursuits company.
We were approached to supply a hosted PBX solution to a Southwest England based Outdoor pursuits company replacing their on site AsteriskNow system with a more manageable and streamlined hosted Elastix based solution.
This gave them the ability to have handsets located where ever they were needed and add handsets easily with the user friendly Elastix interface. It also gave them increased viability of call usage with the inbuilt reporting and FOP2 let them see users engaged on calls.
We ported their existing BT Analogue lines to Gradwell VoIP trunks, This gave them an immediate increase in call capacity and combined with inclusive landline and mobile call packages meant call costs were kept under control.
The aim here is to explain the relationship between the callgroup and pickup group settings in extension conf files of an Asterisk server and named pickup in freepbx, we will use numbers but not names (see explanation below).
Call Pickup is the abilty to pickup a ringing phone from another phone.
The ability to do this is defined in the extensions conf file.
In many systems there is only on setting to do this normally “pickup group” you add extensions to this group and they can pickup calls ringing at members of the group. Obvious really.
Now Asterisk goes one better. You can define the callgroup and pickup group, This way you define who you can pickup and who can pickup you. This is very useful for operators, who for example don’t want calls picked up of them but do want to pickup calls from all other users.
So how do you define it.
In our example we will have 4 phones defined as follows
Callgroup Pickupgroup
201 2 1-2
202 1-4 1-4
203 2,4 2,4
204 1 1
And who can do what when trying t pickup is as follows
Ringing Phones attempting Pickup
Call to 201 204 PU failed 203 PU Passed
Call to 202 201 PU passed 203 PU Passed
Call to 203 201 PU passed 204 PU failed
Call to 204 201 PU passed 203 PU failed
So from this we can see that its the Pickupgroup that defines what callgroup can be picked up.
So because 201 has a callgroup of 2 Only sets who’s pickup group includes 2 can pick up the call. whereas as 201 has a pickupgroup of 1-2 it can pickup calls from callgroups 1-2.
For example you may have 6 pickup groups defined with users only allowed to pickup their own group members except an operato who wishes to be able to pick everyone up and a PA who has a college who she wants to be able to pickup
So all normal users would have their pickup and callgroup the same. The PA would have the pickupgroup defined with both the group numbers but only its own call group. And finally the operator would have a callgroup of 0 and its pickupgroup of 1-6.
Named call pickup groups
Named pickup groups are new with Asterisk 11. And are now supported in FreePBX , But be careful even though the ‘hint’ says they can be numeric or names the just use the named variable.
namedcallgroup=office,home,1 namedpickupgroup=office,home
As above we have a namedcallgroup as 1 but this is not the same as callgroup 1
A named callgroup and pickupgroup can be set to a comma separated list of case sensitive name strings. The number of named groups is unlimited. The number of named groups you can specify at once is limited by the line length supported.
SYNTAX
namedcallgroup=[name[,name[,...]]]namedpickupgroup=[name[,name[,...]]] |
- namedcallgroup – specifies which named pickup groups that this channel is a member.
- namedpickupgroup – specifies which named pickup groups this channel can pickup.
Configuration Example
namedcallgroup=engineering,sales,netgroup,protgroupnamedpickupgroup=sales |
Configuration should be supported in several channel drivers, including:
- chan_dahdi.conf
- misdn.conf
- sip.conf
- pjsip.conf
pjsip.conf uses snake case:
named_call_group=engineering,sales,netgroup,protgroupnamed_pickup_group=sales |
You can use named pickup groups in parallel with numeric pickup groups. For example, the named pickup group ‘4’ is not the same as the numeric pickup group ‘4’.
Numeric call pickup groups
(obsolete use named groups)
A numeric callgroup and pickupgroup can be set to a comma separated list of ranges (e.g., 1-4) or numbers that can have a value of 0 to 63. There can be a maximum of 64 numeric groups. This is important to note as Freepbx does not sanity check what you put in there, So you can put 70 in the Gui and it will show 70 but do a sip show peer or a pjsip show endpoint and you will see its not set.