combining 2 existing systems

[Flipmstr2]

I have a situation where an existig customer of mine has been bought by another company that also uses a Shoretel.

1) has anyone had issues changing the license?

2)the customer is actually (for some unknown reason) just wanting to connect the 2 systems via 4 SIP Tie lines. Is there going to be a "best" way of acomplishing this? Should I treat it just like combing a PBX via PRI? HOw are the TIE lines set up?

[Jason Learmouth]

I've never dealt with relicensing a system, that will probably cost money.

SIP tie lines work pretty well. The setup is very similar to a PRI tie, it's just Off System Extensions and all the same limitations apply. Unless they are on new switches which let you allocate 4 SIP trunks from the built in resources you might as well configure 5 since that's what you get from a port. SIP Trunk licenses will be required at both ends.

Just create the trunk groups and add trunks to them which point to the other SG switch hosting the trunks, mirror this on the other system and you should be up and running.

[ECook]

Good luck changing the licensing. We had the same situation and it took 10 months to get it all straigtened out. The biggest issue was ongoing support....what they actually had vs what ShoreTel showed they had.

[jmccumber]

We have two separate ShoreTel systems that we are trying to link via SIP. I used a similar setup to what you described and all we get is 5 digit dialing back and forth. The trunks hang on 'offering' for anything off system (in or outbound).

I assume if you have trunks at both sites this would easily take care of what you need.

[Flipmstr2]

Well I guess I won't need to worry about the license issue, they don't want to integrate the two systems. They will only be TIE lined together.

So here is my question.

The new company is a 6 site company.
How will line access work? Will I need to define a seperate routing table or a dial "8" access code?

[cburgy]

Shoretel actually leverages this in large installs (10,000+ seats) to scale up.


You'll setup your SIP trunks between the two and actually use OSE to define the ranges that exist on both systems. Is the dialplan the same length (i.e. 4 digit) on each system? Are there any overlapping ranges?

You'll create static sip trunks pointing to the opposing SG switch on each system providing SIP trunks. What version of Shoretel are each of the systems on?

[Flipmstr2]

Unfortunately I do not know any info about the multisite system.
Ours is 7.5.

My main question is about trunk routing, If I dial a number that is local to the other site, is it routable? Naturally if it is overly involved then they may be S.O.L.

[cburgy]

Unfortunately I do not know any info about the multisite system.
Ours is 7.5.

My main question is about trunk routing, If I dial a number that is local to the other site, is it routable? Naturally if it is overly involved then they may be S.O.L.

You mean least cost routing? As in seize their local PRI (for example) for their local area code?

[Flipmstr2]

Yes, least cost routing. Is it doable? I know that there are local area codes associated with the SIP trunks. Does the system utilize these for call routing purposes, or are they strictly for how calls are dialed.

[jmccumber]

Shoretel actually leverages this in large installs (10,000+ seats) to scale up.


You'll setup your SIP trunks between the two and actually use OSE to define the ranges that exist on both systems. Is the dialplan the same length (i.e. 4 digit) on each system? Are there any overlapping ranges?

You'll create static sip trunks pointing to the opposing SG switch on each system providing SIP trunks. What version of Shoretel are each of the systems on?

Chris, you seem to have a handle on this, maybe you can help me out as well. As I mentioned above we are trying to tie two systems together so we can test all aspects of 8.1 in a lab environment along side our production 7.5 version. They are both on 7.5 for now and they are both using 5 digit extensions. The numbers do overlap but they don't necessarily have to. We can dial between the two over the SIP trunks but can not get to or from the test system from outside?

I apologize for sneaking in on someone else's post but the problems are similar.

[Jason Learmouth]

Chris, you seem to have a handle on this, maybe you can help me out as well. As I mentioned above we are trying to tie two systems together so we can test all aspects of 8.1 in a lab environment along side our production 7.5 version. They are both on 7.5 for now and they are both using 5 digit extensions. The numbers do overlap but they don't necessarily have to. We can dial between the two over the SIP trunks but can not get to or from the test system from outside?

I apologize for sneaking in on someone else's post but the problems are similar.

You will need to allow the types of calls you want to route in the SIP trunk group on the "trunkless" system (i.e. local, LD, etc.) On the system connected to the real PSTN, you will need to put that systems trunk access code in the "Prepend Dial In Prefix" section of Tandem Trunking. Make sure that Tandem Trunking is enabled and that both trunk groups have a user group that has trunk to trunk enabled.

[jmccumber]

Thanks, unfortunately that is exactly how I have everything setup but still not working. The odd thing is at one point I could actually call out to a cell phone from the 'trunkless' system as long as I had the access code in the Tandem Trunking (without it, I at least got the AA on the other system). Then for no apparent reason it stopped working and has not worked since. I have considered doing a repair on the database thinking maybe one of my many changes did not get updated correctly. We did have to do a repair to clear an issue with an OSE that did not update in the database at one time. Thanks for your reply, we are actually ordering two T1 switches, a little much for our needs but it will at least more closely mimic our current set up when we test 8.1.

[Jason Learmouth]

You should ensure that the user group of the user making the call has access to the tie line trunk group, this isn't required for OSE calls but it is for this type of routing. You should have the DTMF option near the top of the SIP trunk group enabled for the tie line trunk group on both systems as well.

On the trunkless system, are there any other trunk groups setup? I'd have a look at the trunk test tool to ensure the call is going out the tie line, also have a look at trunk test tool on the other system to see what is coming in and were it is being routed back out.