Need a 50+ incoming only DID

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I have a friend of mine that needs to do a once a month conference call with all 50+ stores for a while. I was thinking of something like DID for Sale. however they are only 20 channels. any idea of a place that I can get something like this for the cheap. just like did for sale. I am doing this for free for him as a favor type of a thing (long story, trust me).
 
DID for Sale offers additional channels for $1 each. So the total monthly hit would be $38.99 for fifty channels. I am not sure you can find a cheaper offering. Hope you don't have to pop for this for very long:eek:.
 
I must be missing it, it seems as though it is per minute. after the first 20 channels
 
From their FAQ page:


Is there a limit to the number of DIDs or channels I can order?

Theoretically, there is no correlation between the number of DIDs that you have and the number of channels that you can have. Our DIDs come standard with 20 channels. If you need more that 20 channels then we can give you buy additional channels for $1 each. For example, if you need 30 channels then you pay $10 extra per month for your DID.
 
wow-zers.......

i feel dumb. it is late and a very long day maybe next time I should get some sleep and RTFM!!!!:banghead:
 
maybe just me being hopeful. but some of the ITSP's out there, such as ipkall, should be able to provide something like this????


say have everyone call that and have it dumped into a conference...
 
Fifty inbound calls into one system will eat an awful lot of bandwidth. With G711, you'll need more than two T1's of internet, very fast and stable cable or FIOS to feed it. I assume you're looking at it from that angle as well...

There are a number of providers who will up your number of channels to as much as you can eat if you are pay-per-minute, so you need to shop around.
 
Okay,

I have to ask why take this approach to the issue. If it's 50 stores coming into a once a month conference I would looke at alternative connections.

You don't say what each store uses for network or phone, but a copy of a SIP or IAX2 softphone at each location with a single channel/extension to the server would work. Or having the stores system complete a SIP/IAX2 call to the server would also work, Dial(SIP://...)

I would find an answer beyond carrier connections given the senerio. Much easier to predict and control costs and bandwidth needs.. You could even maybe use a regional approach and link two or three sessions together so all 50 streams are not over one connection..

Good Luck..


I have a friend of mine that needs to do a once a month conference call with all 50+ stores for a while. I was thinking of something like DID for Sale. however they are only 20 channels. any idea of a place that I can get something like this for the cheap. just like did for sale. I am doing this for free for him as a favor type of a thing (long story, trust me).
 
Does anyone use PiaF (or any Asterisk/Freepbx) to run conferences this large? I think I'd be more worried about the system requirements to for this many conferences attendees.

Kenn10 also points out a valid concern about bandwidth required for this many concurrent calls. 85kbps x 50 is almost 3 x T1s bonded. It seems like this conference server should be hosted at a colo.

As far as VoIP service providers, it seems like any of the pay per minute guys would work. 50 callers x 60 min (assuming this call is an hour) x .015 = $45.
 
true. I am just wondering why not do it myself. of course with the one i found, it is free. but maybe more stable if I did this myself????...... I can see bandwidth being maybe an issue. but most likely not.
 
Besides the other relevant issues already brought up, 50 concurrent calls means 50 different users that may have issues. You run the risk of being blamed for something that may not be related, like a choppy voice form a cell phone user, bad line from a telco, etc.
IMHO - if you are doing conferencing with more than 10, you should look at a hosting provider. It's just not worth the headaches.
'Just because you can, doesn't mean you should'
Here is a provider to check out:
http://www.raindance.com/rndc/index.jsp
 
I co-wrote this a couple of years ago to facilitate a multiple Asterisk Box conference. We tested it with 3 machines and that worked swimmingly. The code needs some updating, but here it is. Feel free to update and re-write as needed. Just post your changes so all can share.

Code:
[macro-multimeet-2]
;this module joins the user to the conference, and initates outbound calls to
;other conferences

; To call this macro you enter 
; exten => _X.,n,macro(multimeet-2,<conf number>,<trunk1 code>,<trunk2 code>,... etc)
; I put the above line of code in from-internal-custom

; ARG1 is the conference number.  All other ARG's are the access codes for the
; remote boxes.  This software requires that a conference number of ARG1
; be set up in advance on each box.  The interconnecting trunks must be
; from-internal on the remote machines.

; This section counts the number of trunks. You need to know how many .call files 
; to make and this number is needed when you tear down the conferences at the end
exten => s,1,Set(meetmepin=${ARG1})   ;the conference number
exten => s,n,Set(ArgCount=2)        ;Set the loop counter to start at 2
exten => s,n(toploop),Noop(Parse ARG's to see how many there are)
exten => s,n,Gotoif(${ARG${ArgCount}}?:loopout)
exten => s,n,Set(ArgCount=$[${ArgCount} + 1])
exten => s,n,Goto(toploop)

; Set the Argument count to the correct number
exten => s,n(loopout),Set(ArgCount=$[${ArgCount} - 1])
exten => s,n,NoOp(Joining current caller to conference number ${meetmepin})

; See if the conference is already set up by counting the number of users 
; already there.  If there is more than one, we assume it is set up and 
; skip to the bottom and just add this user.
exten => s,n,MeetMeCount(${meetmepin}|count)
exten => s,n,NoOp()
exten => s,n,Gotoif,$[${count} >=1]?100

; If this is the first user, we set up the links between the servers

; Start reading data with $ARG2
exten => s,n,Set(newcount=2)

; Kick out of the loop if we have read all of the Arguments
exten => s,n(newtop),gotoif($[${newcount} > ${ArgCount}]?moveon)

; Create the .call file and send it to the outgoing directory
exten => s,n,system(echo "Channel: Local/${meetmepin}@from-internal/n" > /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(echo "MaxRetries: 1" >> /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(echo "WaitTime:5" >> /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(echo "Context: from-internal" >> /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(echo "Extension: ${ARG${newcount}}${meetmepin}" >> /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(echo "Priority: 1" >> /etc/asterisk/${ARG${newcount}}${meetmepin}.call)
exten => s,n,system(chown asterisk:asterisk ${ARG${newcount}}${meetmepin}.call)
exten => s,n,Wait(1)
exten => s,n,system(mv /etc/asterisk/${ARG${newcount}}${meetmepin}.call /var/spool/asterisk/outgoing)

; Increment the counter and go back to the top
exten => s,n,Set(newcount=$[${newcount} + 1])
exten => s,n,goto(newtop)

;this code above creates a ".call file" that will be placed into /var/spool/asterisk/outgoing.
;This call file automatically generates a call within the asterisk PBX.
;The entry "Channel" defines the 'local' meeting the computer should join
;The entry "Extension" is the destination meeting it should join.
;This creates a 'trunk' between the two.

; The call file created is unique for each trunk called.  I found that if you didn't 
; create a different call file for each trunk, the existing one would be overwritten
; before it completed its connection.  

;On my two asterisk PBX's, I had created a meeting '6000' using FreePBX on each.
;In the above example, my 'far end' PBX is contacted with an outbound prefix of 77.
;So, when the person enters in the meeting they want to contact as 6000,
;the system will call 776000
;an outbound route in FreePBX simply of 77|. will send 6000 to the far end system.
;the far end system matches that to the local meeting and a link is made between
;the two systems.
;make sense? :)

;If you have multiple sites that you want to call, just repeat the code,
;modify the s,XX accordingly
;as well as the "EXTENSION" entry to reflect the outbound dial prefix that you need.

exten => s,n(moveon),Goto(s,100)

; This adds this user to the local conference
exten => s,100,MeetMe(${meetmepin}|Mdsp)

; This line sets things up for the hangup
exten => s,n,goto(h,1)

exten => h,1,goto(h,101)
exten => h,101,Noop(Break down trunk when all callers have left)
exten => h,102,MeetMeCount(${meetmepin}|newcount2)
exten => h,103,Gotoif($[${newcount2} > ${ArgCount} - 1]?104:106)
exten => h,104,Wait(30)
exten => h,105,goto(s,102)
exten => h,106,System(/usr/sbin/asterisk -rx "meetme kick ${meetmepin} all")
exten => h,n,hangup()


; the first three contexts are separated in order to group them by the appropriate dundi priority
 
A conference service will most likely have more options in terms of entry exit notification, muting one or all parties, recording, options for future playback after the call concludes, the hardware is already tweaked etc.

Murphy's law indicates that on the day and time your planning there will be an unexplained outage or screwup of some kind. People will then start attaching a monetary value to there time vs. your "I did it myself factor". They will determine you were motivated by glory, didn't care about them and are an overall low life....

No good deed goes unpunished.... The ability to blame someone else shouldn't be underrated or overlooked....

(just a bit of humor)
 

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