What do I need to set up an Asterisk service?

Question

I've got a rather in-depth request for anyone who has jumped through all the hoops.

What do we need to make and receive calls to/from VoIP & PSTN phones worldwide, and what would the cost be?

We intend to use our existing Cisco 7940 / 7960 IP phones.

What specs should the server have for good call quality? we will have 5 offices with 8 users (perhaps 4 concurrent), and 1 with 20 users (perhaps 10 concurrent).

With the server located in one site and all our IP phones/stations located throughout the United States, what kind of internet will be needed at both the server and IP phone ends? (asymmetric/symemetric? need packet prioritization via the switch to ensure voice call quality? will cable internet be ok?)

Do we need to pay a service to convert our IP calls to/from the regular PSTN network? what about long-distance PSTN?

What support, installation, troubleshooting, fixing solutions does Digium offer and how much does it run? We need to ensure that we won't lose service.

Answer 1

First of all, Asterisk is a more or less freeware attempt at IP telephony. If you allready have Cisco 7940 and 60 series phones then you probably allready have a Cisco Call Manager, and ip telephony working. If you want to get rid of Call Manager and migrate to Asterisk, you'll first need to check and make sure your 7940 and 60 series phones allready support, or can be upgraded to use SIP, rather than the sccp protocol they are "probably" currently using. I don't honestly remember whether they are upgradeable or not. Those models have been replaced with the 7941 and 7961 respectively. A call to your Cisco SE or if you have support, a TAC case will clarify that for you.

Your proposed operation is relativel tiny, so your Call Manager server requirements will be relatively minor. It will depend to a great extent, upon how many bells and whistles the Call Manager is going to be asked to support. Cisco uses a point system to determine server size based upon the services required and number of ip telephony endpoints.

Bandwidth usage depends upon the CODEC you are using, but figure that you will need about 80K per call, so if you have 4 concurrent users at a site, you will need to have at least 320K of bandwidth guarenteed as available for ip telephony, at each end site.

If you desire phone service to maintain the same quality and reliability that you currently receive from the phone company, you will need guarenteed bandwidth and quality of service enabled on and end to end basis. IE, the path that any ip telephony endpoint may take to reach any other ip telephony endpoint, must be QoS enabled.

If you want quality and reliability in your system, cable connections to your end sites will not work. Typicaly, cable companies do not offer gurenteed point to point bandwidth and quality of service, so unless you can find a cable providor who will offer that service, cable is probably not a good choice for ip telephony enabled end sites. Otherwise, using cable connections, you will be at the mercy of everyone elses traffic patterns. If the path is good on Monday you'll have good calls. If the path is overloaded on Tuesday you'll get tons of droped calls.

Yes, you will need a PSTN gateway connected to your ip telephony system, or you will not be able to make calls to anywhere other than to your own ip telephony endpoints :-)

Where you put your PSTN (public switched telephone network) connections will depend upon your calling patterns, to a great extent, and to another great extent, depending upon how much surviveability you want your remote end sites to have, should they lose their WAN (Wide Area Network) connection. If you want to insure that your remote sites still have the ability to dial out and receive calls when they lose their WAN connection, then you will need remote site surviveability at each end site, and will need to have some sort of dial back up installed (typicaly located at or in your remote router) to insure that remote systems can maintain phone system connectivity when they WAN is down.

Personaly I would not reccomend any ip telephony solution for the sake of ip telephony itself. It takes a great deal of research and specific knowledge of your coroporate operation before you can make a truly informed decision as to whether ip telephony offers any real savings or improvement over standard or digital centrex or PBX services at all.

You must also consider the 800 pound support gorilla that you will create when you implement ip telephony (Voip). You will be taking on a great many of the support and repair duties, once having implemented ip telephony, that previously you were able to shove off on the phone company to accomplish. This will add costs to your operation, and additional pressure on your it staff to perform, if the phone system goes down.

While what you are wanting to accomplish is perhaps not as difficult as I have made it sound above, I would strongly reccomend that you obtain the servies of a reputable vendor with a ton of experiance in ip telephony implementation, to explore your options and present you with real implementation and operational costs. IP telephony encompasses many skill sets and levels of understanding to implement reliably. Especialy, if you attempt to implement a multi-vendor system as you describe above (Cisco Phones - Asterisk Call Manager), and you will probably thank yourself later for getting qualified help from the outset, to avoid what can be very costly mistakes later.

Answer 2

i could heavily evaluate going with a hosted pbx voip service and changing the analog telephones with VoIP telephones particularly of utilising an ATA... some thing like polycom 550's or 650's for table telephones, or 330's for wall telephones, or telephones that aren't getting heavy use.. in case you want to apply an analog PBX then you definately are in all risk unlikely to be utilising your latest analog telephones, you will would desire to get propriatary telephones that for the period of high-quality condition with the PBX which you get.. for some thing undemanding you will get your self a panasonic 4x4 pbx in case you purely want to apply 4 line analog telephones and pass the full telephone gadget/pbx element, then you definately can get some telephones that have intergrated telephone gadget effective properties... you will get one at&t 1080 which has a equipped in vehicle-attendent (in accordance to the documentation) then use 1080's, 1070's or 1040's because of the fact the extra extensions... some funds extra, yet once you're going the analog course, why not purely use a single SPA8000 (8 line ATA) particularly of the two PAP2 or SPA2102 which you are going to be utilising

Answer 3

I think these questions are too specific... only a service provider who provides these types of services can answer these questions... any way the sales reps can give you this information free of charge... Why dont you just goole some providers out and find out some information from each one? and form an openion?

check out these links anyway...