[asterisk-dev] RES: RES: live chat with asterisk

Carlos Carvalhar ccarvalhar at globalnova.com.br
Tue Mar 11 15:51:55 CDT 2008


Hi Dean,

 

Can you tell me the asterisk apis involved with busy agents? 

Eg.: how do I set one agent as busy? I can set it by php, don’t I?

 

I’m planning to use php to set an asterisk variable telling the agent is
free or busy. 

 

Is there any software like this one, Centriphone Millennium, for free?

http://www.vocalcom.com/asterisk.html

 

 

Thanks,

Carlos

 

 

  _____  

De: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] Em nome de Dean Collins
Enviada em: terça-feira, 11 de março de 2008 16:43
Para: Asterisk Developers Mailing List
Assunto: Re: [asterisk-dev] RES: live chat with asterisk

 

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



__________ NOD32 2933 (20080310) Information __________

Detta meddelande dr genomsvkt av NOD32 Antivirus.
http://www.nod32.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080311/e3452015/attachment.htm 


More information about the asterisk-dev mailing list