Explanation of SIP Trunking and the ipLDK system

[jongemcf]

Hi,

In brief we have an ipLDK-100 and ISDN lines giving 4 CO lines. We have gradually expanded the system such that we now have 2 DTIB12s and an SLIB12 card giving us 36 internal extensions. In addition we invested in a VMIBE card and have just bought some more legacy handsets to go with our shiny new extensions. In short, there is a not insignificant investment in telephony equipment which works for us very well. (We are a UK non-profit so cost is often sensitive!)

I am however very interested in how we could utilise SIP to replace our ISDN CO lines. 4 channels would be quite sufficient but can some one indicate if the following scenario is correct:

- We purchase a VOIBE card which gives 12 channels of IP telephony
- We subscribe to a SIP trunking service which will route to/from the PSTN and also handle SIP-PSTN and PSTN-SIP.
- We set-up the ipLDK to connect to the SIP service and then presumably the current (non-IP enabled) DTKUs and SLIBs can access these as normal COs and receive calls in the same way as before.
- We use DISA and CCRs with voice announcements to route calls so presumably the networked COs can act in the same way as normal COs.

Of course in future we could buy IP enabled handsets and extend the number of internal extensions beyond those supported by the DTIBs and SLIB.

Comments on this? Have I got the gist of it right?

Another question is: If I select a SIP provider how do I reap the benefits of free VOIP-VOIP calls if I am dialling normal geographic numbers? Do I need to know their IP address to call??

Thanks in advance for reading this and hopefully taking the time to reply

jon.

 

[TryaReboot]

Hi Jon,

You got the gist ok..

On the iPECS system, the SIP channels (once programmed) are basically classed as DID lines and are programmed in the same way as you would with ISDN through PGM231 etc.

I am only assuming its the same for the ipLDK system whereby you have the details set up in the VOIB attributes with amount of IP channels etc and all your IP data from the ISP/Router etc and then, assign the VOIB in logical slot assignment with the amount of allocated lines in whatever group the ISDN was in prior.

Maybe someone else can confirm this but this is my understanding.


Hope this helps a bit.

Steve

 

[doktor]

Good report.

It is possible to use SIP instead of ISDN.
You should have the same services from the public network with SIP and ISDN, depending on your provider.
You can also add IP phones to your existing ipLDK-100.

///doktor

http://www.kandu.dk/spg153996.aspx

 

[jongemcf]

Thanks both for replies. I think that satisfies my main questions. Having read the VOIP part of the manual further I can see I can make calles by IP address but I assume this is only be using IP handsets on the system (with the VOIB/VOIBE card!).

I am keen to know how we reduce outgoing call costs overall bymaking sure that people we happen to call (we have no idea of course) are using a SIP/VOIP service. The only mechanisam I can see this happening by is the ENUM (e164) standard but it seems this is yet to be really adopted. I appreciate this is goin away from the scope of LG kit but any hints are appreciated!

Thanks
Jon.

 

[Chamvari]

If i am understanding you correctly do you want people from youR organisation to only use SIP service when calling OUT, in this case you can program LCR (Least Cost Routing) that is if you have other lines in the system which are not SIP.

Dial Tone shoul be There