Addign VOIP to my Network

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VoIP
Hello all. I have been tasked with finding a way to save money in our telecommunications costs. VOIP can save us money in the long run, but must me implemented correctly or you risk headaches. I want to know if it is possible to get a VOIP solution that will let me keep my existing avaya partner pbx in our LA office and our Toshiba Strata here in NY?

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Unless you are HellBent to suffer the pains of pioneering, or want to spend a lot of bucks with Cisco, I would suggest contacting COVAD for your primary offices in NY and LA. They have a backbone of their own and much higher quality of service than the other BrandX guys. There is probably a 70% chance you can save money and even keep your vintage PBX system with their offering of PBXi. Then you can use that until you feel it valuable to go straight VOIP with new handsets. Good luck!

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  • Johnny95003
    What model of the Toshiba Strata do you have?
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  • Hedgehog
    Hi bklynbound, You should analyse your phone bill to see the % of intra-office calls. You will be surprised to discover that a large number of calls are between your offices. If that is the case and you have a semi-decent broadband connection on both offices, I suggest you install out-of-the-box VoIP Gateways to get free calls between your offices. You should look for one that offers you the option to plug in directly into your existing PBX. They appear as additional extensions and are automatically routed to the remote office. They are fairly simple to setup, you get good support from vendor or reseller and get to keep your existing kit for a few more years until you decide to go all-out for a VoIP PBX and sign up for VoIP providers as the 1st poster suggested. Check out the MultiVoIP gateways from the good ol' Multitech people (http://www.multitech.com) Another alternative is to setup SIP proxies on each end and use either softphones on the PCs or hardware VoIP phones in your LAN. There's obviously more configuration for you to do, but SIP is a recognised standard (as opposed to Skype!), there are several open-source SIP proxies and I am sure there must be some SIP Registrar in your area that will provide SIP services & numbers. Check out siproxd for FreeBSD, Linux, OpenBSD, OS X & Solaris on http://siproxd.sourceforge.net/ We use it here and it works great. Send me PM if you decide to go for a SIP implementation and want to test it. Good luck! Hedgehog
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  • Paul144hart
    The idea of using SIP/VoIP on your data network needs to be considered carefully - you'd want to define a QOS VLAN for the VoIP so the calls don't get packets lost due to congestion. Using the old PBXs like that would require a gateway such as the Cisco / Nortel , etc and maybe / probably also an Adtran or other box to bring the T1s to the PBX (you can split a T1 easily and allocate numbers of ports to your telephone provider - depends on your site size / config). Hopefully the PBX does have T1 open spots!! I think I've seen adding a T1 card to a PBX to be ~$3000 (was an old nortel pbx). The open source proxies are okay - but you don't get techynical support. There are quite a few companies out there that would give you a free quote if you want to a jump on what it may cost. For analyzing the cost you may want someone to do the financial judgements - you'll probably want either internal rate of return or a break even point to show that it actually will save money. Someone with an MBA... -Paul
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  • Bklynbound
    If this helps in anyway. I have a Toshiba Strata 280 key system (50 phones) here in NY (DKSUB280A, DKSUE280A) with a Voice T1 with DIDs and PRI (I think). We have a SDSL 1.5 MBPS connection to the net. In LA there is an Avaya Partner key system (ACS R7). They connect to us through a VPN tunnel between two netscreen devices. they have a SDSL connection to the net as well.
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  • Paul144hart
    For your netscreen boxzes, make sure they will allow the SIP or H.323 protocol and RTP packets. With 1.5M of SDSL, you could allocate about 200K of bandwidth for the VoIP traffic - many due to the fact you have a T1 PRI trunk. Not said if you are using all of that for calls. You can put in a couple softphones and a proxy to test connectivity out for not much money. Better SIP softphones is eConf or Pingtel.If you look at H323, there are free softphone s/w (http://www.sipfoundry.org) - founded by Pingtel. Remeber, RTP streams are 8KHz/8bit or 16KHz.. each stream is a voice channel.
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  • Mikel69
    I would recommend spending the bucks with Cisco, but of course I am biased :) But this will offer the most flexibility / compatibility with your current infrastructure.
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