[asterisk-dev] RES: live chat with asterisk

[Dean Collins]

Hi Carlos,

 

This is the first of 3 issues with any non telephony agent 'interaction' technology (be it chat or efax or other interaction channel).

 

1....how do I know the agent isn't on a call.

 

2....how do I route a chat to 'qualified' agents

 

3....how do I distribute inbound chats to multiple qualified agents on a rotating round robin basis.

 

 

 

There are simple ways to solve and technical ways to solve.

 

 

The easiest is you have 1 agent available at her desk but not logged into the telephony queue all day and she just answers inbound chats that day.

 

 

The next harder way is - when an agent receives a chat you build an api to let the queue know that they agent is on a web chat (hard depending on what chat application you are running - though asterisk has the api's for busy agent).

 

 

Problems 2 and 3 involve commercial chat applications and are beyond the scope of this discussion but that should give you a start.

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
dean at cognation.net 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 

________________________________

From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Carlos Carvalhar
Sent: Tuesday, 11 March 2008 3:03 PM
To: 'Asterisk Developers Mailing List'
Subject: [asterisk-dev] RES: live chat with asterisk

 

Hey,

 

Sorry for the ignorance but I'm a newbie about asterisk...

 

At my work, there is a call center using asterisk to control the queue of the clients already. This part is ok.

But now I need to make a chat room at the site and someone of the call center will need to answer that client.

So my doubt is how to implement a solution that identifies an operator who is free and put him to talk by chat and then make him busy to phone calls.

After the web chat is finished, the operator turns automatically free again.

 

I'm not sure, but for what I want do I need a ToIP?

 

Thanks

Carlos

 

 

 

________________________________

De: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] Em nome de Gunnar Hellström
Enviada em: terça-feira, 11 de março de 2008 04:03
Para: 'Asterisk Developers Mailing List'
Assunto: Re: [asterisk-dev] live chat with asterisk

 

Carlos,

There is support for T.140 real-time text ( RFC 4103 ) in Asterisk.

 

It can go with any other media in a SIP call.

But I am afraid that meet-me or app-conference do not support it yet.

 

It is activated in sip.conf  by:

textsupport=yes

allow=t140

allow=red

 

There was basic T.140 support in Astersik 1.4. But you would need 1.6 or vidcaps to get a more mature support.

You can use it with clients supporting RFC 4103. There is a free softphone in the project Tipcon1 in Sourceforge.

 

Gunnar

-------------------------------------------------------------------

Gunnar Hellström

Omnitor

gunnar.hellstrom at omnitor.se

Tel: +46708204288

www.omnitor.se <http://www.omnitor.se/> 

 

 

________________________________

From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Carlos Carvalhar
Sent: Monday, March 10, 2008 9:27 PM
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] live chat with asterisk

Hello,

 

How can I make a live chat (mainly text, but with voice chat if possible) interacting with asterisk?

Can asterisk control simultaneously the queue between people calling by phone and people by web chat?

Is there any free solution?

 

Does MeetMe and app_conference have something about it?

 

Thanks

Carlos



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