Manage many internal call

we are planing to implement a system that should manage 900 internal call. and i’d like to now wich interfaces to use.

you are going to need to be a hell of a lot more specific than that. does 900 internal call mean 900 concurrant calls, 900 total users?

Also what kind of business is it and what is your usage pattern line? IE, if you have many departments and much of the call volume is between users in the same department, it might be easier to deploy several * servers, one for each department serving 100-300 users, which would all link back to a main server using iax2, main server providing PSTN interfaces, IVR, etc. OTOH, if you are running a call center type operation, 99% of your traffic is going to be external, and this becomes less efficient, not more.

It also depends on how your network is set up. 900 users by either definition is a hell of a lot, ie a lot that can overwhelm a standard 10/100 network infrastructure if its not dedicated.

Also what if anything are you upgrading from? do you need all 900 seats now? will you grow to it? answer these type of questions and provide much detail, and you will get a far more helpful answer.

Until then, a few things to think about-

  1. for ‘which interface’ it depends on what phones you are using.

IP phones- if you have the network for it, this is almost certainly the way to go.
If it’s a call center, you can usually get by with softphones (PC software and headset) and that will be cheap, $10-30/port. (one port = one user/phone/seat). If you want hardphones or it’s an office, check out something like the AAstra 9133i ($160) or 9112i ($130 IIRC) if you don’t need PoE. AAstra phones kick ass. SNOM rocks too, and has a full product line. Grandstream is okay if you are on a budget, but their remote configuration thing isn’t nearly as good. They have the GXP2000 (good office phone, $80-90 and the Budgetone series (no alpha display, feels cheaply made, crap speakerphone, $50-60)

POTSphones- avoid if possible, IMHO. More likely to cause issues with echo, bad wiring, etc. Might consider this as a budget option if your building has no ethernet but does have POTS.
To do this you have a few options- load up servers with Digium cards (TDM2400 type), channel banks (PRI-many POTS breakout device), or ATAs / media gateways (POTS-SIP adapters). You are generally paying about $60-90/port, not including the wiring or the phone that goes on the other end. That phone can be anything from a $10 radio shack wall phone to a $1200 polycom conference room set. Anything that needs a ‘phone line’ will work with this.

  1. can your network take it?
    If you are using IP phones, consider putting the entire VoIP system on a VLAN. Program your switches to prioritize this VLAN over everything else. Also consider what codec you are using, depending on what codec you use an IP call can take anywhere from 15kbit/sec to 80kbit/sec including overhead. Upgrades to your network may be required. If you are going to use IP phones, consider PoE (Power over Ethernet, IEEE 802.3af). IF a PoE compatible phone is plugged into a powered switch (they are more expensive than normal switches), it will draw power over the Ethernet jack from the switch. This means the phones do not require wall wart adapters (plug in transformers) and also if the switch is on a UPS, all phones will continue to operate in the event of a power failure. This is of course more expensive, as you need PoE switches and IP phones. However it also can add flexibility, as in the event of an emergency configuration change you can quickly reboot everybody’s phone by resetting the powered switch :smile:

  2. who is providing service?
    How are these 900 people going to dial out? If you are going to use VoIP, what codec? Probably you will want GSM or G.729 codecs, which both use around 15kbit/sec. G.729 requires a licensing fee per channel, whcih may be expensive. G.711 ulaw/alaw is the most bandwidth intensive, 64kbit/sec+overhead=around 80kbit/sec. It is however usually acceptable for faxing. You will need to figure out how much bandwidth is needed for VoIP and order an appropriate link. You WILL need to set up a QoS (quality of service) control on this link, to prioritize voice packets over other data.
    Alternatively, your service may be provided via PRI (T1/E1/J1) lines, which give you about 23 (T1) or 31 (E1, as i recall) voice channels per circuit.
    Whichever option you choose, you should get quotes from many providers to find the best one. Also keep an account active with an alternate carrier (perhaps a pre pay carrier) so if the first one is down, you can at least send some calls somewhere.

Hope that helps. If you give more info, i can help more.