[asterisk-dev] live chat with asterisk

[Gunnar Hellström]

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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