One touch conference 1

do they do anything else like make outbound calls?

I can think of a few things, but you need to state your requirements.

Can they only speak to each other once all 5 answer? Or can the people who answered speak to each other immediately once they pickup regardless if the last person picks up?

Do you need all the phones to ring until all of them answer? Maybe in a teleworking scenario? Or are they always within earshot of each other so each phone doesn't have to ring until they're answered?

Out of curiosity, what's the use case?

So its a group of control rooms across a large plant and this is the emergency hotline, when the emergency phone receiver is picked up, it will automatically dial and connect all the emergency phones at other control rooms, setting up an immediate teleconference and all parties are able to talk at the same time. The information supplied to me is a bit limited, but I would expect that talking immediately once more than two parties enter the call would be best. The phones are manned 24/7 and only for emergency so all should be answered.

Group Paging lets you talk out a bunch of phones that can't talk back to you.

Intercom groups let you 2 way talk, but you only call 1 extension in the group at a time.


SA8893 is blast conference.

Do these phones do anything else? Call 911? If so, you can probably accommodate for it. But, as it stands, here's an idea:

-Change system special, enable 8893
-Change vector 1 - or whatever vector you want to use
-Set that vector for meet-me conference to Y
-Step 1 is wait 0 sec hearing silence
-Step 2 is route-to meetme
-Presuming extensions 2221-2226, add VDNs 3331-3336.
-They all point to vector 1
-Set meet-me conf on page 1
-Set conference type blast on p2.
-For VDN 3331, brigde in 2222,2223,2224,2225,2226
-For VDN 3332, bridge in 2221,2223,2224,2225,2226
-Etc
-Add analog X ported stations 4441-4446
-Change stations 4441-4446 and set abbreviated dialing list 1 to personal 1
-Change abb personal 4441
-Set item 1 as 3331
-Change station 4441
-Set the hotline as List 1, dial code 1
-Change station 2221 and make the first appearance abrdg-appr of 4441
-Have the next 3 keys as call-appr
-Set page 2 Select Last Used Appearance to Yes
-Make sure the phones have the first line for the abrdg-appr selected.
-Do the same thing for abb lists for 4442-4446 for 2222-2226 to call VDNs 3332-3336
-You can have 2221-2226 at "auto-answer all" if you want them brought in immediately, just know that if someone isn't at a desk, you'll have to hang it up, might hear noise, whatever


That's my best guess right now

Blast Conference it's definitely the feature you are describing. I think the post above is the answer you'll want - SA8893.

That's great an looks like exactly what I need thankyou

This works ok apart from it only involves the first 5 additional parties (conf restricted to 6 we need 8), i did try altering system-parameters features

Page 6
Conference Parties without Public Network Trunks: 12
Page 7
12-party Conferences? y

But still the same, we use only G450 and G430 and all DSP resource is in those gateways, is the restriction there and a media server required to go beyond 6 parties?

I don't think the special app would take advantage of the new 12 party conference. You need to use a looptyloop trunk to bring more than 6 in. Haven't tried it, but to be fair, you did say you only needed 6

And no, no media server is required. You're fine with G430/450. Just don't use SIP phones for it.

20. Conference Limits

There are fields on the system-parameters features form that control the number parties that can be on a conference call.
• “Public Network Trunks on Conference Call”
• “Conference Parties with Public Network Trunks” and
• “Conference Parties without Public Network Trunks”
These fields are ignored for Blast Conference calls.

Loop-around trunk group administration
o CM shall administer per system the trunk group to be used for Blast conferences of more than 6 parties. Notes: It's the customer's responsibility to: configure the associated hardware so that all of the trunk group's members loop around, back into CM and to make sure the trunk group has as many members as the maximum number of Blast Conferences of more than 6 parties likely to be in use simultaneously on this system.
• Loop-around destination administration
o CM shall administer per system an extension number that will be recognized as the incoming destination for Blast conference loop around trunks.

Not sure I quite understand, so you add a trunk group against "Loop-around Outgoing Trunk Group" is this a physical back to back trunk? So out on an analogue trunk port wired to an analogue station port for example? Or 2 PRI with crossover between them?

Could be a DID on a PRI. Never wired analog ports together, be careful It's just a way to make it a 2nd call that can have up to 6 participants. If you've got SM, you can just send 1234 out to SM, and have SM send 1234 back to CM and have that loop around destination be 1234, so when CM wants to add 7+, it'll make a new call out to SM to 1234, SM will send it to CM and when CM catches it on 1234, it'll just keep adding people.

Yes have an SM so what steps are required for instance want a blast conference with 1000-1008, 1000 initiates and 1001-1008 in the VDN (2000) config, set "Loop-around Outgoing Trunk Group" to the SM SIP Trunk group 10. what is the config to send the 1234 to the SM and when it comes back to the CM what's it setup as?

No idea. Never tried.

I'm presuming that because the blast conference doesn't have an extension on its own, that the setting of 'internal conference extension' is the number it would dial out the loop around trunk.

In sm you'd just have a dial pattern for 1234 pick the routing policy to CM.

TraceSM and see?

So config the "Internal Conference Extension" and a trace shows that the set called the Internal Conference Extension this then sent an invite on the sip but with no Called number, Calling number is the VDN. As I don't get a called number at the SM how do you turn it back to CM.

Just see this at start of invite in SM

INVITE sip:abc.com SIP/2.0
From: "VDN_Name" <sip:2000@abc.com>;tag=4cf987a42ddc41eb8780c29xxxxx
To: <sip:abc.com>

so as the To doesn't have any digits how can I do an adaptation on it to get some digits to route back to CM

Try having a dedicated Sig group with domain blast.customer.com and do a domain based pattern for x. Maybe a regex to match 'nothing' would work too.

Otherwise 8.1 has source number routing available as well, so you can route based on what number it's coming from.

how do you do a domain based pattern? As I am not managing to get these looped around.

when you make a dial pattern, it's for a domain. You can pick ALL, or a specific domain.

Are you on 8.1?

It honestly looks like it was meant for an analog loop around trunk if it's not sending called party number. You'd probably need a regex dial pattern match that matches "anything" because any dial pattern in SM is based on digits of which you must have a minimum of 1 - which this type of call clearly does not have!

Yes on 8.1.2 and like you say it looks like it is just expecting a call on the trunk assigned to Blast carrying the CLID of the originating VDN that way it knows what Meet Me its for. As it says in the document its expecting the customer to be responsible for the trunk so maybe just back to back PRI etc. Like you say would normally need at least a digit, but as it is configured in CM in the System Parameters it does not pass through any routing so cant add some digits there. I did wonder about doing it back and forth between 2 CM's (SIP) if i could do it there but again how do you route an inbound call with no digits, there is Night DN but not the Digit treatment prompt that I could see.

Trunks do have an incoming destination - but I can't remember if that's just analog.

I figure they probably intended that to be the case and instead of burning 2 PRI cards back to back, I think they meant to burn 2 analog ports.

I've just never plugged 2 analog trunk ports into each other. Not sure if that's a good idea - I never was great with that FXO/FXS stuff!