Best Asterisk Deployment Options

We are looking to deploy asterisk for different clients ranging from 50 to 80 phone users. We are trying to make a decision but and would like everyone’s opinions;

  1. Should we get a PRI-T1 line or integrated T1 line and DiDs from carriers like voicepulse?
  2. Should we get a data T1 line and get DiDs?

or is there a better way to go about such deployments?

Thanx guys.


I hesitate to say that there is a ‘better’ option. That’s like saying do you want to heat your house with gas or oil, there are ups and downs to both (although many will probably have other opinions).

If you get a PRI T1 and run the phones over it, you will be limited to 23 concurrant calls at any one time. You will get your phone service from a local telco. It will probably include unlimited local calling with a slightly steeper than voip fee for long distance. You can generally get DID service; that is you buy a number of phone numbers (DIDs) and the line signals which one the call is coming in for. Pricing on this depends on your local telco or whoever you buy the T1 from (not always a telco). Reliability is good unless you have a crappy provider.

If you get a data line and run VoIP over it, this can also work very well. Using G.729a codec, I have heard of getting 100 concurrant calls (channels) across a single 1.5mbit full data T1. G.729 is of course a compressed codec, so such a setup won’t sound as nice as the T1.

If you go the datalink/VoIP route, you might save some money. There are a few things to keep in mind tho-

  1. YOU NEED QOS CONTROLS ON YOUR WAN LINK this is not optional. If you do not have good quality of service control on the wan link you will get choppy audio whenever someone downloads something. The ONLY time you don’t need qos is when VoIP is the ONLY thing hooked up to the link.
  2. your phone service is only as good as your Internet connection. If it sucks, well, you get the idea.
  3. make sure you have enough bandwidth. using ulaw/alaw codec, a single voice call uses around 80kbit/sec including overhead. g.726 (adpcm) reduces that to 40 or 50, gsm/ilbc/g.729 put it in the 20s. The more you compress the audio, the less good it will sound. Most people don’t mind g.729 or GSM, but you can’t fax over it.
  4. You can’t fax over a compressed link. You need ulaw/alaw and even then sometimes it doesn’t work. You CAN fax over a T1 line’s phone channel.
  5. get a provider that doesn’t suck. Voicepulse doesn’t suck. Remember that if your provider sucks, you may be unable to port your DIDs out of them.
  6. stay away from hosted pbx services. They often don’t let you port DIDs and you will lose extension dialing if your Internet goes down.

Thanx a lot for the insightful response. The pros and cons are well highlighted. I lean more toward g711 for the best audio and to reduce CPU usage for the server/pbx itself. So in summary it is better to go with data T1 for more concurrent calls as opposed to a max of 23 max with PRI T1 plus more expensive rates.

Iron hit the nail right on the head with his post.

However i make comment though on the Codec selection.

Currently in our Setup we have Dual Xeon 3.2GHz Servers with the latest Intel processors, and i can tell you with this CoDec even with such power it chews the ass out of the CPU, the difference is that with the latest Xeons you don’t loose voice quality until you reach about 97% usage of that CPU. So in saying that best of to avoid this CoDec unless you absolutely need to use it.


These are the better of the CoDec choices, however whilst they do very little damage to CPU load, they do however chew the ass out of the bandwidth, and once your link reach 85%-90% Saturation they may produce really choppy voice quality, even with a non contended 1-1 link (Which is what we use) this may still happen and most likely wont be avoided.

If your not thinking of doing any faxing over the data link and are most likely going to have a lonely old POTS line handling the fax, then i would also recommend the G.726 CoDec, whilst we are still running testing on it and the Jury is still out, this CoDec has so far shown it can really preform and in most cases it has even equaled uLAW/aLAW voice capabilities. However we personally ourselves are having a damn hard time getting our Tier-1 Carriers to use that CoDec, they just don’t like to play ball, every time we ask they keep telling us to only use G.723 or G.729 which i am not a big fan of myself.

Now i don’t know exactly what costs are for Bandwidth in the states, but if you can and if it is possible, try your best to get a non-contended 1-1 service or if not try and make sure it’s contention ratio is not greater then say 20-1 this way you can be sure that you can achieve best quality results of the data packets.

Also keep in mind, that no matter what you should have at least 2-4 POTS lines spare if you can, phones with businesses are mission critical, and generally pots lines do not fail, where as Internet Links have so many points of failures they still are not reliable enough to run your business of it solely (This can be argued, but it is just my opinion), also what ever DID’s you get, make sure the ITSP can do re-routing of calls to those DID’s to some back up lines, either the spare POTS lines or something else, this way your ass is covered.

Hope this helps.



just to clarify a bit- a T1 can give you 23 usable voice channels or around 1.544mbit/sec of data.

Now if you use it in data mode, and running VoIP over it your results will vary based on codec. If you use g.729, preferably over iax2 in trunk mode (reduces overhead), you can probably get 100 or more calls going over that 1.544mbit/sec. OTOH if you use ulaw/alaw you will get quite a bit LESS than 23 calls, because each call uses around 90kbit/sec including overhead. Keep this in mind.