[kwlug-disc] OT: conference call using voip.ms & FXS

[Raul Suarez]

It seems that the Flash/dial/flash is a feature of your ATA, not your phone, So if your phone has that capability, maybe you do a three way on the ATA paired with a three way on the phone. 

See page 77 on the manual
http://www.manualowl.com/m/Linksys/SPA2002-ER/Manual/245813


Alternative2: Can you set up a Softphone for one of the parties so they can also do a three way on their side?
I will refer to you as "party 1" and the person on the soft phone as "party 2" and the other two people as Party 3 and 4
1. Set up a softphone as an extension for "party 2"

2. Party 2's extension
3. You flash, dial party 3, flash (now you have party 1, party 2 and party 3)
4 Party 2 initiates a 3 way call on the SIP phone (procedure will depend on the Softphone), calls party 4 and joins

If everything else fails and depending on the value of that conference call you could try renting a conference line by the minute


Note, I haven't used any service other than my Bell conference line at work, but googling will give you many results such as this: http://www.canadaconferencecalls.com/index.html


Raul Suarez 
  
Technology consultant 
Software, Hardware and Practices 
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http://rarsa.blogspot.com/ 
An eclectic collection of random thoughts



On Monday, February 10, 2014 4:18:44 PM, Daniel Allen <danielrobertallen at gmail.com> wrote:
 
This is sort of the opposite of a Linux question, but I figure someone may have a suggestion.
I have a voip.ms account and a Cisco SPA2002 box to give me two RJ11 ports. I have them 
working with a speakerphone. I've connected three parties together using the phone's built-in 3-way calling (flash/dial 2nd number/flash will 
connect me with two parties).
I want to connect a 4th party; for a call in the next few days.


None of the other participants have 3-way calling.

I have a DID with voip.ms and I imagine a 4th person could call the DID number, but I haven't 
found instructions on bridging together that incoming call with the 
outgoing calls. 
I imagine I could have two people call the DID, along with me calling it
 from the speakerphone, but I don't know how to bridge the calls on the 
DID either.

Options I've rejected: 
- bridging in the 3rd and 4th parties via my cellphone, which makes a crackly laggy mess
- setting up an asterisk or freeswitch box, due to not wanting to store/maintain extra hardware
- setting up a second speakerphone, to run on the SPA2002's second RJ11 
port, connect to the 4th person, and pick up the other speakerphone 
output over the air (which starts to look more like performance-art)

Any ideas- or better yet, cookbook instructions or URLs? 
Thanks.
-Daniel
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