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Astribank is a versatile and powerful channel bank that was specifically designed for the Asterisk IP-PBX. Astribank supports all the common telephony lines and trunks: FXS, FXO, BRI, E1/T1 PRI, T1 CAS and E1 R2. The Astribank driver is a part of the standard Asterisk distribution.
The aim here is to explain the relationship between the callgroup and pickup group settings in extension conf files of an Asterisk server.
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 etensions to this group and they can pickup calls ringing at members of the group. Obvious realy.
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 dont 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
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 whos pickup group includes 2 can pck 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 collegue 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.
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.
Named pickup groups are new with Asterisk 11. But are not yet supported in FreePBX upto and including 13, So be carefull and dont add them to your pickup/call group settings yet in Freepbx as they will not work eventhough it shows in the GUI.
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.
Configuration should be supported in several channel drivers, including:
pjsip.conf uses snake case:
When you switch to Gradwell, you can enjoy all the great benefits of our award-winning VoIP telephony service and keep your existing number. Becoming a Gradwell VoIP customer doesn’t mean losing your existing PSTN phone number. Our dedicated porting team can transfer your existing UK and international numbers within days of joining us to minimise any disruption to your business.
Cost: £20.00 one off fee to transfer your number, capped at £200 if you have a lot of numbers in the same sequence to move.
£20.00 each port rejection.
£10.00 for any date change/cancellation.
We are able to move your number from various telephony providers, including:
Full geographic porting:
Partial geographic porting agreements:
Full non-geographic (08) porting agreements:
We are always building on our porting agreements so if you have a number from a company not listed above, please let us know and we will see if it is possible to port.
To keep your number, you will need:
(Please note this is a guide and times are may be subject to delays)
Please note, any resubmissions or changes will be charged at £20 (excluding VAT) and will be subject to a new lead time. You will also be charged this amount for any numbers you port, but then do not use or cease. It may therefore be beneficial to cease the numbers with the current providers before putting through the port request to minimise cost.
For more information regarding porting your number, please visit the FAQs.
The 8180 SIP Audio Alerter is a loud ringing, paging and intercom device for use with SIP telephone systems. It has two main uses. The first is as a loud ringer for use in places such as warehouses. The second use is as a paging and intercom system. It can operate in both these modes at the same time by appearing as two different extensions on your phone system at the same time.
When registered with a SIP server, one endpoint will play an audio file from internal memory upon ring detection. The second endpoint will auto-answer for voice paging or full intercom (two way audio).
Equipped with a high efficiency integrated amplifier and tuned high quality loudspeaker, the 8180 is typically eight times louder than a telephone speaker. Several audio files are pre-loaded into the 8180 internal memory for ring sounds but users may also record or upload custom audio files, music, sound effects, or voice announcements.
The advanced features of the 8180 include SoundSureTM technology which automatically adjusts loud ring and loudspeaker volume to compensate for background ambient noise. Ideal for variable noise environments (restaurants, workshops, classrooms, etc.), SoundSureTM ensures that ringing or paging is always heard but not unnecessarily loud.
Please contact for pricing and avalibility