Linking 406 and 403 using PRI/T1

[amitiu]

I need to tie a 406 into a 403 for a demo using the PRI modules, but I am having a hard time getting the two to talk. I have tried all sorts of settings for both boxes, and am stumped.

Does anyone know the settings required for the lines to get them to work, and also does this require a crossover cable or a straight cable? Thanks.

 

[BuckWeet]

you need an x-over cable for sure

pins 1+2 swamped with 4+5

also, you might need a CSU because I don't think the IPO can provide clocking...

BuckWeet

 

[IPGuru]

You will also need to set one end to qsig A & the other to Qsig B.
If this is for Demo purposes it is worth remembering that Qsig calls DO NOT go through Incomming CalL Routes

 

[amitiu]

Thanks for the replies... one more question though. I thought the PRI module had a CSU built into it? Is that sufficient or do you think I need to get one?

 

[BuckWeet]

It has a built in CSU, but I don't think you can tell that CSU to provide clocking for the link that is where you would need the external CSU to do clocking..


BuckWeet

 

[gee4u]

Further to this thread...did you manage to get this going??

Regards,
Graham

 

[lgasdia]

If this is for demo purposes, I think that it is best to connect these together as a small community via IP Trunking, LAN port to LAN port with a standard CAT 5 cable.

To the best of my knowledge, connecting two units together via digital trunks only gives you limited (Q.Sig) feature transparency.

For customers that have had full T1's between locations, I would recommend this also.

 

[gregarican]

lgasdia, that's what I am pondering. Right now I am just scoping out the suitability of upgrading my Merlin tandem networking endpoints to IPO. I would have a full point to point DS1 to work with and was thinking that just using this as a WAN data link would be a "better" fit than trying to run as PRI. I know about all of the inherent PRI call control and feature advantages versus T1 but would I have similar advantages just using WAN IP?

 

[lgasdia]

The advantage of IP Trunking is that you will have better utilization of the P2P T1, with G.729 compression ("toll equivalent&quot, you will more than double the number of concurrent calls, I tell customers ~30K per call with G.729, but if you bet on ~20K you should be fine. Traditional T1 channels are 64K.

If you have a dedicated T1 for voice, and want "toll quality" with VoIP trunks, setup your compression to g.711, which is the 64K, and DO NOT check Silent suppression - I have had more quality complaints from silence suppression than compression method.

The IPO small community network requires IP Connectivity, so the features/functionality with the IP Trunking is night and day. Check out the small community networking features - which are available with IP connectivity.

Equipment side, make sure you have the VCM resources!

 

[gregarican]

See my other thread about my setup --> thread940-741546 I have a HQ with 40 phones that links to a satellite location that has 10 phones. The HQ has a PRI for PSTN access and the satellite location uses this PSTN access with only a handful of loop start lines locally for backup purposes. I was thinking that a VCM 30 for the HQ and a VCM 10 for the satellite location would suffice.

Is this a correct assumption?

 

[lgasdia]

IP to IP will not consume the VCM resource for the length of the call, only during call setup.

The VCM resources will be consumed for the length of the call when going from the TDM to packet bus.

So ... digital set to IP Trunk = 1 VCM resource
IP Phone to IP Trunk = momentary VCM resource for call setup

Main site digital set to remote site digital set = 1 VCM resource at the main AND 1 VCM resource at remote for the entire call. If you have IP Stations, then these resources will only be consumed for the length of the call if you are going out over PSTN lines.

NOTE: even thought the Voicemail is on a PC, it is still on the TDM bus - so IP Stations checking voicemail will consume the resource as well.

This is kind of hard to explain in righting, I hope this made sense! Did you end up going with digital or IP sets, that will be the determining factor with the VCM modules.

 

[gregarican]

I haven't actually implemented anything yet, just researching what would be involved. I was originally thinking all IP sets.

 

[lgasdia]

With all IP sets, match up the number of local lines with the VCM resources (T1 = 24 trunks = VCM 30, all trunks can be accessed - and you have a couple more for the Voicemail)

Main site would need the 30, remote would need the 10.

With Digital sets, match up the VCM with the IP Trunks, the number of calls you think will occur between locations.

Both sites would match resources, and I don't think you would need more than 10 IP trunks for the number of users you're talking about, so VCM 10 at both sites, (only takes up 20% of the T1 with g.729!!)

You would have teh same 10 IP Trunks with IP Sets, but you won't need the extra VCM for these necessarily.

Depending on your reason for deploying IP Sets, ROI is more evident in the IP trunking/long distance - So I would recommend the digital sets for initial setup, and if you add more extensions later on and run out of digital interfaces, throw in some IP sets then.