3300 Clustering with Asterisk (SIP)

[jnfern]

Hi all,

We currently have 4 Mitel 3300 ICPs in a cluster. I can create a 4 digit extension on any ICP, and call this from a phone on any other ICPs.

I have created a SIP Trunk to an Asterisk server, currently I have 4 digit extensions on this, and I'm using ARS Digits Dialed form to say 'If I dial 8888xxxx' route this to Asterisk. This works fine, however not ideal as people need to know if an extension needs this prefix or not.

What I would like to do is be able to dial the 4 digit extension which is registered on Asterisk without any prefix, so a user doesn't need to know where another user's extension is registered.

I have tried to add the Asterisk Network Element to the Cluster Element Assignment form, however this wasn't identified as an existing element. If I created a new element within the Cluster form, and then configured it as a SIP Peer in the Network Element form it was automatically removed from the Cluster form.

I'm very new to the world of Mitel, so don't know if what I'm trying to do is possible or not, and if so am I on the right lines. Our current cluster does include a couple of non-Mitel entries (old analogue exchange) but I can't see how they were added when what I tried failed.

Any suggestions for a solution would be appreciated.

Cheers,
-Jeff

 

[slapin]

It is not possible to make it work in Mitel cluster. But is it possible to make it work through the cluster.

Include SIP extensions in protable directory and assign them to cluster element which has SIP trunks to the asterisk server. Then make this SIP trunk the last choice for prefixes pointing to xxxx block of extensions.

When you dial one of the extensions residing on Aserisk PBX, the call will be routed to the element which serves SIP trunks. Then cluster routing prefix will be stripped and just 4-digit extension will be presented to the call router. Call router will try to look for local extensions first, then it will check ARS table where it will find the route to the asterisk server. Then you can insert some extra digits as an indicator for the Asterisk server where this call comes from. After that it is easy to route this call to any desired destination with Asterisk BPX.

 

[jnfern]

Thanks for your post slapin. I understand what you mean about having the extensions residing on Asterisk routed the the ICP which has the SIP Trunk (Remote Directory Number Assignment), however I don't get what you mean by:

"Then make this SIP trunk the last choice for prefixes pointing to xxxx block of extensions."

Do you know which form this would be under in System Administration Tool? You go on to say when the local exchange can't find the extension it will check the ARS table, is it a case of setting (in ARS Digits Dialed): 4 digit extension as Digits Dialed, 0 to follow, Route, Termination number <whatever needed for SIP>.

To make matters slightly more confusing, our extensions seem to have been allocated randomly, so I don't think I'm going to have a single xxx-yyy range available on Asterisk. Would this then need EVERY extension to be added to the ARS form as above, or have I completely misunderstood.

Sorry if these are noddy questions, I'm still very new to the world of Mitel.

 

[slapin]

You understood me correctly. You need to make an ARS mask which will match every posible extension on Asterisk PBX.

For other members of your cluster they will appear to be located on transit node.

I have seen a document describing lookup order. Possible that it was a post on this forum... I remember that the call router will look for local extensions first, including hunt groups and other local objects, then remote directory and at last ARS. I use this method for our video phones integration.

Last choice means that you may have route lists in your configuration, so add this route at the end ot the list.

For example you have extensions 1000 - 8000 in the system and any of them can be connected to Asterisk

So you will have to add folowing ARS rules on the node, the Asterisk PBX is connected to:

1 +3 Route to SIP trunk
2 +3 Route to SIP trunk
3 +3 Route to SIP trunk
4 +3 Route to SIP trunk
5 +3 Route to SIP trunk
6 +3 Route to SIP trunk
7 +3 Route to SIP trunk
8 +3 Route to SIP trunk

 

[jnfern]

I understand ... although I'm not quite ready to try those ARS Digits Dialed just yet given what could happen if it doesn't work quite right. I might try it with an XXX + 1 route to SIP trunk so the most I can break is a set of 10 extensions - just as soon as I get a range of 10 digits I can try.

The other thought I had, and this is more me typing out loud. I cannot add Asterisk itself to the Cluster Element, however I could add a dummy entry to the Cluster, which will then allow entries to be added to the Remote Directory Number form (Via OPS Manager) referencing the Cluster Elemend ID of the dummy cluster entry. This cluster entry, although doesn't correspond to a configured Network Element, will have a Cluster Elemends ID digits, which corresponds to an ARS Digits Dialed. This ARS Digits Dialed will be set to route to the SIP trunk. The Network Element form will have 2 entries for Asterisk, one is the 'real' SIP trunk and the other is an empty element which was created by the Cluster Element form.

Actually, this seems like a more complete way to do it, as the Directory is then up to date, and knows what extensions are actually in use.

 

[slapin]

You will not be able to use OPS manager since it creates remote directory records based on real extensions assigned to network element.
Actually the idea itself is pretty good. I will definitely keep it in mind and try whenever I will have a chance. Maybe you will be able to use manual editor in OPS manager and assign directory records to this fake Cluster element. Asterisk will strip intra-cluser digits without any problems. The other issue is that you will need to maintain a list of prefixes for all clustered extensions on the Asterisk server.

 

[jnfern]

The reason why I said using it in OPS Manager is because the help page within the System Admin for the 3300 says the Remote Directory Number form should not be used to add details, only OPS Manager should be used.

I've not had time to look at this any more yet, but I will report back on if it works.

 

[slapin]

It would be nice to hear about your results, just don't put too much expectation on it.

You can edit Remote Directory manually as long as you know what are you doing. OPS manager is able to sort everything out automaticaly, but does not produce any magic.

 

[jnfern]

If by 'know what you are doing' means taking random guesses and seeing what happens then I'm an expert

 

[slapin]

This is an initial stage. Later on you will sort it out