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This is the support site for Andrews & Arnold Ltd, a UK Internet provider. Information on these pages is generally for our customers but may be useful to others, enjoy!

VoIP SIP Trunks: Difference between revisions

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Our VoIP system can simple 'trunk' calls to/from your own PBX - it is perfectly normal for you to have your own office PBX, such as a FireBrick or an Asterisk server for example and for your equipment to provide you will all the 'PBX' type functionally that you require. However, you can also have SIP phones register directly against our servers and use the PBX functionality that we provide. Or, indeed, have a mix of the 2 - e.g. we can route calls to you by SIP (trunk), but we can also record calls.
Our VoIP system can simple 'trunk' calls to/from your own PBX - it is perfectly normal for you to have your own office PBX, such as a FireBrick or an Asterisk server for example and for your equipment to provide you will all the 'PBX' type functionally that you require. However, you can also have SIP phones register directly against our servers and use the PBX functionality that we provide. Or, indeed, have a mix of the 2 - e.g. we can route calls to you by SIP (trunk), but we can also record calls.


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=Outbound SIP Trunks - you to us=
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In practice all VoIP Numbers from AAISP can be used as a SIP Trunk, in that the SIP account can be used to make multiple outbound/inbound calls.


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There is no limit on the number of calls you can make with an AAISP VoIP account, simply use the username/password and the server in your PBX config.
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=Outbound SIP Trunks=

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SIP trunks are available from AAISP. In practice all VoIP accounts from AAISP can be used as a SIP Trunk, in that the SIP account can be used to make multiple outbound calls.

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There is no limit on the number of outbound calls you can make with an AAISP VoIP account, simply use the username/password and the server in your PBX config.

If you'd like SIP trunks from AAISP, then please do contact the Sales department.


For your server config, you'd use the same details as that of a normal SIP phone:
For your server config, you'd use the same details as that of a normal SIP phone:


*'''Server''': voiceless.aa.net.uk
*'''Server''': voiceless.aa.net.uk
*'''Authentication Username''': Your Number in International format, e.g. +441234567890
*'''Username''': Your Number in International format, e.g. +441234567890
*'''Password''': Your SIP Password ([[VoIP_Password|which needs to be set via the control pages]])
*'''Password''': Your SIP Password ([[VoIP_Password|which needs to be set via the control pages]])


=Inbound SIP Trunks=
=Inbound SIP Trunks - us to you=


Inbound calls can be routed to a registered SIP phone, redirected to another number or routed to your own SIP server. To help where a customer may have a block of numbers the AAISP Control Pages have a concept of grouping numbers together to use a common number as their settings.
Inbound calls can be routed to a registered SIP phone, redirected to another number or routed to your own SIP server.


Configuration of our side, to send calls to your server, is set on the Control pages. You'd want the 'SIP to your Server' option.
Configuration of our side, to send calls to your server, is set on the Control pages. You'd want the 'SIP to your Server' option.
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In some number ranges (e.g. 033) we offer DDI (direct dial in) blocks. These are blocks of 10, 100, 1000 or even 10000 numbers routed to one destination. You could have several blocks.
In some number ranges (e.g. 033) we offer DDI (direct dial in) blocks. These are blocks of 10, 100, 1000 or even 10000 numbers routed to one destination. You could have several blocks.

To help where a customer may have a block of numbers the AAISP Control Pages have a concept of grouping numbers together to use a common number as their settings.


To manage the blocks each block has a base number in our control pages e.g. 01234567890 for a block of 10, or 01234567800 for a block of 100. This allows you to control where the numbers are sent (e.g. SIP to your server, registered SIP phone). Incoming calls to any of the numbers is delivered as per this setting.
To manage the blocks each block has a base number in our control pages e.g. 01234567890 for a block of 10, or 01234567800 for a block of 100. This allows you to control where the numbers are sent (e.g. SIP to your server, registered SIP phone). Incoming calls to any of the numbers is delivered as per this setting.
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