MOVIN' ON UP-- LEGEND TO ASTERISK VOIP MIGRATION 1

[rapples]

Hi-

I have a client that's just about max'd out their old Legend system and we want to get into this century by migrating the system to voip-- in stages.

Here's what I'm thinking.

Phase-1
Install Vonage/Cisco ATA-186 adapter on a couple of the first outgoing lines-- leave everything else the same. I want to see if people complain about the quality before I get any further into my plan. Potential savings-- $250/month.

Phase-2 Build Asterisk server with Digium t1 card, add additional t1 card to legend, connect legend to asterisk server via t1. Trunk calls over ip, possibly send internet calls over legend voice t1 circuit. Cost $4500 done right.

Phase-3 Replace Merlin mail. Add SIP phones.

Phase-4 Ditch Legend system, buy replacement phones.

Phase-5 Ditch Voice T1 connected to Legend. Potential Savings $750/month.

Here's why I posted this.. I figure with some contribution a plan can be written and quicksand might be avoided. If you have any interest in this project let me know.

Thanks,
Richard

 

[franke]

rapples,
You failed to mention a main point: Why do they need VoIP?
Are they a large call center that require 'screen-pops', do they have a large number of off-site empoyees that need 'virtual' phones and PBX features?

Has anyone done a traffic study and network study to determine if it is even feasable for voice and data to coincide.

Do they have a dedicated data department that is familiar with the requirements of VoIP (primarily QOS). Are they willing to take on the additional work load or will the job still be 'telephone-man vs. data-man'. If so, how will trouble cases be resolved- if a call fails, is it a problem with the Internet Provider, the router, the gateway, the server, bandwidth problems, etc.

How much will they have to spend up front on infra-structure? When (if ever) was their data wiring certified? Will they require new workstations or OS's, network cards, servers, routers, gateways, etc.?

Is a better solution a 'hybrid' system that incorporates both a 'standard' PBX with VoIP capabilities (that would also include expansion capabilities) such as the Definity line (Prologix, ECS, etc.)

There are a lot of questions that need to be address before taking that step into 'this century'. I'd be interested in your and others opinions on this subject, as I'm sure most of us are being asked more and more about this topic.

franke

 

[jinxs]

I am new to the whole VoIP. I have yet to go to school at AVAYA U for Ip office. From what I was told so far, like franke said, all your equipment has to be able to handle VoIP or there will be consequences. We have an IP phone test set in our office, the quality is there as far as I am concerned. Then again, it is only one on our whole network. Again, more things to consider, UPS capabilities, maybe not a big deal but still an expence adding more equipment to the data rack. As well ,the IP phones require local power at the workstation, so if power is lost, no phones, no computer.

As well, like franke said, read the posts in the partner forum, IP office is not yet compatible with Micro 2003 server. They are still working on the patches for it. Will your ip equipment be up to snuff?

VoIP has come a long way since the later 90's, and according to the experts is "ready for the everyday market". Yet, there are a lot of things to consider.

 

[rapples]

Howdy-- I thought I'd get past all that and right to the technical issues.. but I'll answer it so you get a background.

You failed to mention a main point: Why do they need VoIP?
Are they a large call center that require 'screen-pops', do they have a large number of off-site employees that need 'virtual' phones and PBX features?

a: The old system is max'd out-- and we need want to get away from traditional telephony and move towards voip for costs savings. I calculated a 75% savings if we can dump our dedicated voice t1.

Has anyone done a traffic study and network study to determine if it is even feasible for voice and data to coincide.

a: Yes, we just doubled the size of our dedicated pipe to accommodate the extra traffic. The upgrade was free because of a newly renegotiated contract. (not doubling the pipe would have resulted in no cost reduction)

Do they have a dedicated data department that is familiar with the requirements of VoIP (primarily QOS). Are they willing to take on the additional work load or will the job still be 'telephone-man vs. data-man'. If so, how will trouble cases be resolved- if a call fails, is it a problem with the Internet Provider, the router, the gateway, the server, bandwidth problems, etc.

a: Yes, that is me. we're running 99.9975% uptime.

How much will they have to spend up front on infra-structure? When (if ever) was their data wiring certified? Will they require new workstations or OS's, network cards, servers, routers, gateways, etc.?

a: All the cabling is Enhanced cat 5, 1150 drops. The workstations all get scrapped at the end of next year anyway-- but we're not looking for any kind of integration. The data network will run next to a new separate data network specifically for the phones.

Is a better solution a 'hybrid' system that incorporates both a 'standard' PBX with VoIP capabilities (that would also include expansion capabilities) such as the Definity line (Prologix, ECS, etc.)

a: We've put 60 grand into what we have, the service contract is pricey, the phones are now eight years old. It's time to go this direction-- it's been decided-- we have about 200 computers that we're going to push for one more year, then they're all getting scrapped in June 2005.

There are a lot of questions that need to be address before taking that step into 'this century'. I'd be interested in your and others opinions on this subject, as I'm sure most of us are being asked more and more about this topic.

a: so assume that we've decided to do this, now let's focus on what is needed from a technical standpoint.

Thanks!

 

[franke]

I suppose, from a technical standpoint, it would depend on who's IP solution you end up using, whether it's Avaya, Cisco, 3-Com, etc. A trip to there respective websites should give you a rough idea as to what each needs to integrate with your current (or planned) environment. After narrowing down the field, have a rep stop by to give a deeper presentation. Many of them will even provide an 'on-site' visit to a customer currently using their solution. Figure on roughly a two to three month lead time before the actual install and at least two weeks after that to iron out the bugs.

And of course, there's always Tek-Tips

franke

 

[rapples]

Thanks for your reply-- Maybe I wasn't clear enough.. we're investigating moving to Asterisk- a GNU PBX package that runs on Linux. LEGEND TO ASTERISK VOIP MIGRATION was the title of the thread. I think most of it is straight forward, but I'm not quite sure about the interconnection between the two systems-- how the T1 needs to be configured and such.

Are there any Lucent Legend and Asterisk experts out there?

Rich

 

[Ghostin]

Are you wanting to switch the all the equipment, or are you just wanting to change your T-1 service from standard T-1 to Voip? You can't connect a Legend to a non Avaya switch and expect it to work correctly.

 

[rapples]

First step is to produce some pots lines using voip fxo adapters such as those from Cisco and Motorola. Test the waters.

2nd step is to add an additional t1 card to the legend, and have the asterisk pbx with digium t1 card provide additional channels to the legend.

then, I'd try to figure out if I could access the asterisk's voicemail across the t1 that ties the two switches together, or add fxo cards to the asterisk and send the voicemail out some jacks on an 016 card perhaps. I'll need to come up with some way to transfer calls to voip phones connected via Ethernet to the asterisk pbx-- I might be able to do something with a phantom extension with remote forwarding.

Anything ringing a bell?

 

[rapples]

Now that I think about it... it sounds much easier to connect the legend to the asterisk pbx using 800 cards, and connecting the asterisk's t1 to a channel bank..

what do you think about that?

 

[jomaha]

I think that is a poor solution (since I have systems set up like this). Analog ports carry both haves of a conversation (local talker and remote talker). That is OK when there is virtually no delay (pure analog) or small delay as in digital telecom curcuits and switching. When using VoIP, you introduce larger (and potentially more complex) delays - allowing echo to become a issue. Asterisk has echo cancellation algrithms - but you are far better off if you keep the conversation halves separated by avoiding analog lines until the handset cord of a digital phone thus you are better off with a T.

I cannot yet recommend connecting asterisk to a legend T1 interface. I have set up an asterisk box with a T1 port configured as the network side of a PRI curcuit but have not found a setup where the PRI works well. Is the Merlin implementation if PRI subject to trouble? This is on thread...

http://www.tek-tips.com/viewthread.cfm?SQID=878435&SPID=689&newpid=689&page=1

 

[Pepperz]

If you are goping VoIP office to office on a dedicated FR, you may have some success. If you are going out over the internet, it will fail, pure and simple. All venders of equipment and network services tout the fact that it can be done, and it can. I've done it, but there is a catch. It's called latency and QOS, quality of service. When out over the internet, on an e-mail it doesn't matter if it gets sent in segments. It all arrives, and it works. On a voice conversation, if part gets send with a little delay, and part gets a lot of delay during peak hours, you drop the call. If you don't need to make calls duringpeak hours you MAY be OK. Over the internet there is NO quality of service.
If you go VoIP, you can only "call" an IP number, so you'll need another service provider to get the IP voice traffic converted to local dialtone. There is another expense.
VoIP will definitely work for your only if YOU have TOTAL CONTROL of the network, and if I understand it right, you're talking about a full bore application over the internet. Not a good idea just yet.
There is also the issue of voice mail. Each conversion of form provides some loss of audio quality. With the compression and decompression, the analog to digital and back again, you can hit poor cell phone quality really fast.

Pepperz@charter.net

 

[merlinman]

IF you made this work, and no longer need the Legend, send it to me.

Thanx,