Anyone know the best way to get SIP trunks onto a Definity?

Which Definity and load are we talking about??

I'm not certain I understand your question. Sorry.

I have a client with a Definity PBX that currently has 4 PRI's into their system. They would like to have SIP trunking in order to save on Intrastate calling. I am becoming familiar with Asterisk SIP servers and am seeking knowledge on how (if at all) the Asterisk server can be interfaced into the Definity PBX.

It would be best to configure for both in/out bound call(s).

I hope this provides clarity.

There are different models of the Definity PBX, and this could matter with "if and how" to implement SIP trunking. Mike is simply inquiring the specific Definity model and software release.

2 examples (G-Model Definity):
Definity G3r (Generic 3 w/ Risk chip)
Definity G3i (Generic 3 w/ Intel chip)

You will need something to convert the SIP traffic to TDM or analog based, since the G3 doesn't support it natively.

https://support.avaya.com/edoc/docs/def8_1/d81conn2.pdf

Starting on page 271 describes IP trunking.

The switch and release of software they are running will determine what options are available to them. They can type in this command for you and get both. "newterm" This will list the system and version of software they are running. Then go to the Avaya site above and see whats available.

Document only shows IP trunking, not SIP trunking. Also, you may need to look at an SES box to perform the proxy function

SIP trunks came out in CM3.0. Any software load less then that, it can't be done. You can due IP trunks with legacy definity R9.5 (or is it R10) and greater.

Awesome start. Thanks very much. Let me get some answers and I'll return with more questions I'm sure.

I understand some people have figured a way to get Asterisk interfaced with the Definity's as a means for (Native) SIP trunks.

This is what the customer has.

System: G3csiV6

Software: G3V6i.03.3.246.1

The CPU interface card is a TN798B if that help you at all.

A G3csiV6 is not even capable of doing IP telephones or trunks, let alone sip trunks as liquidvw stated above. The switch needs to be R9.5 or greater to do IP telephones and trunks.

What about utilizing a gateway for the SIP trunks? Anyone have experience with this? I've seen this accomplished with "lesser" quality switches.

Thanks,

You could trunk the Definity to something else then use SIP from there.

As in an Asterisk box...?

bestboards. This post made the hair stand up on the back of my neck. I DO NOT want this to sound mean or critical, but when you have a "client" and you don't even know what it means when asked for the software version they have... You may not be qualified to serve them. Granted the term "load" is a little, at least to me, in an Avaya environment. Sounds more like Nortel talk.

I will readily admit to knowing less rather than more about Avaya, but I do know my limitations. Not sure I would be counseling a client about a PBX based on what I've heard.

If I'm totally out of line for making these comments then ya'll just tell me to file it under shut the heck up.

Sincerely
Richard

That might do it.

I think the original answer to this was correct. You need a gateway to convert sip traffic to pstn lines (or whatever is currently connected to your PBX.) There are boxes that do this now. I found one while I was researching this (it was a little over $300) but never jumped off this cliff. I'm not an expert at this stuff by any stretch.

If you don't want to use a PC you could do it with an Adtran TA900 series IAD. They can generate POTS, channelized T1, or PRI from a SIP trunk.

What is the difference between IP trunks and SIP trunks? Advantage or disadvantage between the two. Advantage compared to using traditional T1 and PRIs I currently use?

As far as comparing IP and SIP, it's not so much a matter that there's a "difference" between IP trunks and SIP, but that SIP is actually a set of strict rules and standards (protocol) for the specific treatment of IP trunking. In addition to SIP (Session Initiation Protocol), some of the other VoIP protocols include H.323, H.248, and Media Gateway Control Protocol (MGCP). Like you, I'm learning the IP side of telephony myself, and I think SIP is easiest explained here.

If I'm not mistaken (and someone please correct me if I'm wrong): SIP also provides for IP trunks to directly interface to SIP-enabled switches and IP/hybrid PBXs without the requirement of an externally installed gateway device (such as a Lucent I-Merge Gateway, which is an externaly rack-mounted appliance that provides interface between TDM and IP trunks.)

Thanks Mike. That helped.

If a vendor is offering IP trunking, of whatever flavor, ... I'm guessing some of it would stay within that carriers network and some could go over the internet? Should I expect a carrier to be able to keep my IP voice traffic entirely on their network? Is it possible to ensure QOS over the internet?
Is it cheaper for the carrier to do it IP?