Callcenter. hard to do

Asterisk Asterisk Support
1

Hello.

Is the way to realize callcenter for incoming calls like that?

  1. We have several agents with computers and sufficiently cheap hardware IP-phones (which one?) with autoanswer function and with headsets.
  2. Agent logged in (AgentCallbackLogin by manager API or by phone call, AgentLogin by phone call - it doesnt matter)
  3. The call is coming to queue
  4. Agent press the key on their computer(!!!) and answer to call.
  5. Agent press the key on their computer(!!!) and hang up the call.

I understand that special software is needed for 4 and 5… I can make that. I dont understand what software should do with asterisk. I couldnt find any API function that can help me.

Have you got any ideas?

2

  1. The autoanswer capability is possible with numerous sip phones. I know it is possible with the Grandstream line

  2. I’m not following what feature you need here

3 and 4) are possible with the trixbox hud lite program

3
  1. grandstream bt200 or gxp2000. NOT bt1xx as they do not have auto answer that I know of. bt200 and gxp2000 are both using the same firmware which supports intercom. I know the gxp supports a headset (buy a cheap 3.5mm to 2.5mm adapter and you have a standard port) and i think the bt200 does too.
4

if you don’t mind too much about voiding warranty, the BT1xx range does have auto-answer and there’s plenty of room inside for a 3.5mm stereo socket.

i “converted” 6 of these last week … 5 so that the ringing could be amplied by external speakers for use in a noisy factory, and 1 to use as a input to a paging/tannoy amplifier and so was set to auto-answer. works a treat.

of course, the quality isn’t brilliant, but you get what you pay for.

5

er yes btw to clarify- bt1xx and all gs phones (afaik at least) have an ‘auto answer’ function, ie they immediately answer any incoming call on speaker.

BT200 and gxp2000 (NOT bt1xx) have an ‘intercom’ function whereby the phone will operate normally however can be set to auto-answer on a per-call basis by adding a SIP header.