Auto Answer (Exist? If not - New Feature!!)

[Gymratz]

Thinking....
On my phone if I put it into "Custom" mode I could have it set to Auto-Answer a call. Better still, I could choose whether or not to automatically add the call coming in to the current call that I am on.

As I don't think this is available directly in the ST system - is there anyone who has done anything similar with 3rd party software that might interface with PCM to accomplish this?

Reason:
When a conference call is started in one of our conference rooms we could set the phone to "Custom" mode and allow the first 5 people to call it to be added into a make-me conference.

[Dr_Klahn]

Well, what you just described is a conference bridge.

[Gymratz]

That would be the hope for what it would be used for - without spending 10k and only getting 12 ports.
I was hoping someone had a way around that though - it's not like the software/hardware we currently have can't support what we want to do - it's just ST saying "If you want to do that, spend 10k first."

[OldPhoneGuy]

That would be the hope for what it would be used for - without spending 10k and only getting 12 ports.

You could take a spare server, load a flavor of Asterisk on it (such as PBX in a Flash, which is free), tie it to ShoreTel using sip trunks, and use the Meet Me Conferencing capabilities of Asterisk. We've done it this way. It's a very basic Meet Me Conference unit but it works very well for us.

[Gymratz]

Now that is the kind of solution I was looking for :-).
Thanks OldPhoneGuy! I'll take a look at that software and see if I can understand/figure it out!

[Gymratz]

I have a box running with PBX in a Flash and a couple test extensions on the system that, using a free SIP Soft Phone, can call each other.
I've entered in inbound and outbound settings from a few other forum topics...

I have some high level (and some detailed) questions though for anyone who has already set this up.

How many SIP extensions are needed? I haven't set this up yet as I don't have SIP enabled and will need to restart the system to do that.

Here is my goal (if it helps answer the previous question).
Primarily it will be used internally. We often have conference calls of 4-8 people, MOST being internally on ShoreTel but usually at least 1 or 2 on cell phones. The goal would be to easily conference in all 8 people. Ideally, there would be enough "room" to have a couple of these conferences going at once (perhaps 24 conference ports).
Does that mean I need 24 SIP trunks?

Also, for the life of me I can't find anything related to conferencing in the PBX web admin - any pointers there? Once I set up the SIP trunks, what else will I need to do in ShoreTel to do this? I'm guessing I need a few conference "extensions" on PiaF, and a way to tell ShoreTel that extensions 7500-7505 are on the PiaF system...

Thanks,

[Gymratz]

I guess the first time I tried to "update" all the new modules it didn't take (when I try to do all at once I get a page not found, so installing certain ones at a time). I now have conference capabilities within PiaF... All other questions still elude me though.

[OldPhoneGuy]

We often have conference calls of 4-8 people, MOST being internally on ShoreTel but usually at least 1 or 2 on cell phones. The goal would be to easily conference in all 8 people. Ideally, there would be enough "room" to have a couple of these conferences going at once (perhaps 24 conference ports).
Does that mean I need 24 SIP trunks?

Once I set up the SIP trunks, what else will I need to do in ShoreTel to do this? I'm guessing I need a few conference "extensions" on PiaF, and a way to tell ShoreTel that extensions 7500-7505 are on the PiaF system...

Thanks,

You will need one sip trunk between ShoreTel and PIAF for each conferee, i.e., 8 conferees = eight sip trunks, 24 conferees = 24 sip trunks. These trunks are placed into their own trunk group. You will need to check Enable SIP Info for G.711 DTMF Signaling in your new ShoreTel sip trunk group. You will also need to assign off system extensions in this trunk group for each conference that you wish to initiate on the PIAF system, i.e., if you wish to use 7500-7505 for conferencing these would be assigned as off system extensions in the ShoreTel sip trunk group.

[Gymratz]

If I want to have 5 conference rooms set up (7500 - 7504) each supporting 8 people each, would that need 5 SIP trunks (5 conferences) or 40 SIP trunks (40 Extensions could be using those 5 conferences)

I should read closer - "conferee" - being each person, I am now going to assume.

[Gymratz]

Ok so I've enabled "Outbound" (had it disabled to begin with).
I've unchecked all trunk services (people won't be calling outside the company with these trunks)...
I added extension 7500-7504 as Off System Extensions.
Is there anything else that I'm missing? Now that this is complete if I dial 7500 I still get the default error AA, so I don't know if it's working... However, I still don't have any SIP Trunks yet due to the RTP forwarding which requires a system reboot... (Finding time for that will be fun, we have offices in the US, Shanghai, Tokyo). Guess I stay up late!

[OldPhoneGuy]

If I want to have 5 conference rooms set up (7500 - 7504) each supporting 8 people each, would that need 5 SIP trunks (5 conferences) or 40 SIP trunks (40 Extensions could be using those 5 conferences)

I should read closer - "conferee" - being each person, I am now going to assume.

Correct... Five conference rooms with eight people in each room would require 40 sip trunks.

[Gymratz]

I feel like I'm so close! Something is wrong with my configuration though...

ShoreTeL Setup:
(Trunk Groups)
Name: APC_SIP
Teleworkers (No)
Enable SIP Info for G.711 (Yes)
Digest Authentication (All)
User ID: siptrunk
Password (pass / pass)

Number of digits from CO (4)
DNIS (Disabled)
DID (Disabled)
Extension (Enabled - Translation Table None)
Tandem Trunking <None>
Destination (My main AA)

Outbound Access Code (9)
Local Area Code (541)
All trunk services disabled.
Off System Extensions 7500-7604


Switch this is set up on is IP 10.10.5.47
I set up 5 SIP_Trunks and told it to use 10.10.5.46 as the IP address for them (Is this correct? Do they each need their OWN IP address? Should it be 10.10.5.47 to be the same as the switch? I tried deleting all but 1 to see if that made a difference... no difference.... (Correction, this is now 10.10.0.156, set to the PiaF server.


FreePBX:
Created a single SIP Trunk called APC_SIP Do I need more? Do I need 25 on here if I have 25 on ShoreTel?
Outbound Caller ID is our main number.
No dial rules
Outgoing Settings - This section now works!
PEER Details:
host=10.10.5.47
username=siptrunk
secret=pass
type=peer


Incoming Settings - No luck
USER Context: = siptrunk
USER Details:
secret=pass
type=user
context=from-trunk

Register String: siptrunk[email protected]/siptrunk (Is this right? i use .46 since that was the SIP trunk IP, but not the switch IP... I tried both, neither worked) This is now the switch IP of 10.10.5.47


Outbound Route:
APC_SIP
Intra Company Route (Yes)
Dial Patterns:
1NXXNXXXXXX
NXXNXXXXXX
XXXX

Trunk Sequence SIP/APC_SIP

Inbound Route
Description: ALL
Set Destination to Extension 7500 (this is a test user I have logged in using a soft phone so I can test...)

I can call OUT from the PiaF into ShoreTel and it works, caller ID and all.
If I call from ST to PiaF it grabs a trunk for a split second and then releases it and gives me a reorder tone.

[Gymratz]

I finally figured out getting to the Asterisk logs, which helped me determine the problem. I was using an incorrect name in the Incoming settings.

I can now call both in and out of the PiaF system. I set up a DID to forward to the conference extension. I tested calling in from two ShoreTel phones and two Cell phones at the same time, all were in the conference room.

The only problem I seem to have left is Inbound Routes. If I don't set up any route, I can only get to an extension "user" and not a conference. I try to get to a conference and it says it doesn't exist.
If I create a route and set a destination, it will ONLY go to that destination.

So what this means for me at this point is that I can only figure out how to set it up with a single conference port. Any ideas?

[Gymratz]

One of the things I had to fix was "allow anonymous calls..."
Could this have anything to do with the problem?

I can call any extension I create just fine, but it doesn't allow me to call any conference number. If I dial anything but an extension (like a number that doesn't exist, or a conference number) I get placed in my default route...

I can set this up to go to an IVR that lets them select which conference... but I'd prefer to have 5 conference rooms with 5 corresponding extensions/DIDs on ShoreTel.

Also, has anyone found out if there is a way to not require a PIN at all? If I don't put one there I still have to press #.

Also just some more information for anyone thinking of doing this. I tested this out this morning with 11 callers (2 external 9 internal) and the hardware utilization was minimal. We are using an old 1.8ghz machine and based on utilization iwth 11 callers could easily handle 100+ without the slightest problem.

[OldPhoneGuy]

Your inbound route should be set up as follows:

Description = Name of Conference Room
DID Number = DID Number for the Conference Room

Under "Set Destination" click the "Conferences" radio button and choose the conference from the drop down menu.

That's all you need to do for the inbound route to work.