I am at the start of putting together a proposal to move our entire telephony over to an Asterisk based solution and am keen to get some hardware & general network/topology advice.
Our existing physical setup:
We are a small company - total 7 employees.
We have a UK main office and two UK satellite offices.
We have one Italian satellite office.
We have one Estonian satellite office.
Each satellite office is staffed by only one person.
We are a brokerage and consequently both make and receive significant amounts of calls throughout the day, often international.
Our existing telephony setup:
We have a PBX at the UK main office (BT Versatility). This has 3 analogue lines attached to it (all share the same number). We also have one 8Mbit DSL broadband line which doubles as a fax line.
Each satellite office has a single analogue line with their own local number. These offices also have 2Mbit+ DSL broadband (attached to the analogue line).
Desired Voip setup:
Asterisk box (debian based) located at the main UK office, using a PBX trunk from www.gradwell.com to receive incoming calls and originate outgoing calls onto the PSTN network. We would migrate our head office number over to Gradwell.
All staff throughout the company are given Polycom SoundPoint IP 430 phones.
The Polycom phones and Asterisk box at the UK main office sit on their own internal LAN which is connected to a dedicated broadband line for connecting to the PBX trunk provided by Gradwell.
The Polycom phones at all the satellite offices are connected to the internet via DSL broadband and also connected to a standard PSTN analogue line.
Incoming calls to satellite offices originate on the analogue PSTN line, however, it must be possible for a call to then be transferred via SIP to any other office.
Outgoing calls from satellite offices can go out via the local analogue PSTN line, however, it would be nice if they could go out via SIP to the Asterisk box in the UK main office (for auditing) and then onto the PSTN network via the Gradwell trunk.
Further down the line we might get international numbers added to our Gradwell provided PBX trunk so that all incoming and outgoing PSTN calls from any office (main or satellite) are first routed via SIP through Gradwell and then onto either the PSTN network (if outbound) or to our Asterisk box (if inbound). In this case the local analogue PSTN lines (which are plugged into the Polycom phones) simply would act as redundancy in the event the PBX trunk or broadband goes down.
Questions:
As you no doubt will have guessed, I am very new to Voip and Asterisk - does the above even make sense? Is what I describe possible?
Is the SoundPoint IP 430 the right phone for the task or should I be using something else?
I presume that because all PSTN tasks will be handled by either Gradwell and their PBX trunk or the Polycom phone, it will not be necessary for us to fit our Asterisk box with any Digium cards i.e. we do not need to terminate any lines locally. Is this correct?
Is it wise to use a PBX trunking service like that of Gradwells or is it better to have multiple analogue or ISDN lines at the UK main office and handle everything ourselves?
From what I have read it would seem that a dedicated DSL line (8Mbits down / 512Kbps up) should be sufficient to handle 5 simultaneous calls. Is this realistic?
I am aware that Voip is renowned for echo problems and low volume. Will using a PBX trunk service alleviate these problems or is it better to do everything internally with decent hardware & cards. Call quality is paramount to us. Cost is a secondary issue.
That’s all (unless you spot some glaring omission I need to factor in/consider)!
I apologise if this has already been covered but I have been unable to find any FAQs or previous posts that specifically fit our scenario.
Regarding the soundpoint 430 phones, the spec page on Polyphone’s site mentions that it has two lines in addition to two 10/100 ethernet ports.
I had taken this to mean that it could be plugged into a normal PSTN analogue line as well as a LAN and thus be able to take/make calls via SIP or PSTN. Am I reading this wrong (very probable)?
Finally, you are correct in that the main UK office does use ADSL max. Therefore, if 5 simultaneous calls is not realistic, what would you say could be achieved - perhaps 3?
[quote]Regarding the soundpoint 430 phones, the spec page on Polyphone’s site mentions that it has two lines in addition to two 10/100 ethernet ports.
[/quote]
What is meant the description is that it has a 10/100 port for connection to the lan and one to connect a “PC” to.
The 2 lines mean 2 SIP lines, IE 2 accounts.
ADSL Max is a weird beast when used with VOIP it can be good or very bad. you need to budget on only 60% of the bandwidth being workable.
Use G729 and 5 calls wont be a problem. You will though need to think about having an external address for the server otherwise double nat will be a real issue.
You might want to look at Gradwells centrex service for the remote offices.
Thanks very much for clarifying the two lines re: the Soundpoint 430. In light of this I think what I am really looking for is a min-asterisk box at each satellite office in addition to a Soundpoint 430. This would allow the satellite offices to make/receive calls via their local PSTN and also be connected via SIP to the UK main office. As all calls will pass through this mini-asterisk box it also allows easy call auditing.
Hi
Depending on the number of lines at the remote offices you could look at PSTN to sip gateways such as vegastreams or Sipuras That way you just have the one central system.
I don’t think a central UK system will suit our model very well. The reason for this is that we have two remote offices abroad and the bulk of their calling will be to clients in their own country over PSTN. In which case I don’t think a Sipura or similar will do what we need.
The main reason for bringing the remote offices onto Voip is so that we can audit their calls (by logging into the mini-asterisk box) and also to give us the ability to transfer a call from say our Italian office to one of our offices in the UK via SIP (and vice versa). Moving all the offices over to voip will of course also allow “free” intra-office communication.
By using a fxo gateway at each site and a central server you will have better control of calls, all sites will be able to break out locally to what ever country they are calling they will be able to call and be called by all users and transfer calls to all users. You will also have one central point of configuration and monitoring/recording etc.
OK, I think I must have misunderstood something somewhere as what you describe is exactly the setup I am looking to achieve.
I read through the user manual of the Grandstream GX400 (which I believe is a Voip gateway like the Sipura) and it wasn’t obvious that this device does what you describe i.e. allows you to have a SIP phone (like the soundpoint 430) attached and then route calls locally over PSTN or via ethernet to a central asterisk box.
Most of the time calls in and out of remote offices (i.e. to/from customers) will not need to have anything to do with the central system in the UK. It is only very rarely that a customer call (i.e. PSTN) might need to be transferred over SIP to another office. Of course it is common for there to be a lot of inter-office calling between staff which ideally should always be over SIP.
Would a Sipura work with a Soundpoint 430 attached?
The other thing I notice (as I currently understand it) is that these gateways seem to be designed to be used with an analogue phone. This seems quite cumbersome as all commands have to be issued via the touch pad.
Ideally, I’d like to abstract away everything to do with voip as much as possible. As far as the staff are concerned they do not need to know how a particular call is being routed. For example:
If they want to transfer a caller to another office they simply go through a couple of menus on the phone screen, select the office/extension and hit the call transfer button.
If they want to make an outgoing call they just pick up the receiver, dial the number and that’s it i.e. the system decides if the call should be routed over IP or PSTN.
If they want to call a regular client they just navigate to the contact directory on the phone, select the person they want and hit the call contact button. This directory would ideally be centrally managed.
To achieve this integration and ease of use it seems the best bet is to fit everyone with fairly high end SIP phones that are connected to a local mini-asterisk box which in turn is connected to the local PSTN via FXO and also the central/mother asterisk box via the internet.
It means more configuration but to be honest - once a box is configured it is unlikely to change very much. Also, I would think the config for each remote office is going to be very similar apart from a few number changes?