[asterisk-dev] T.140 real time text

Gunnar Hellström gunnar.hellstrom at omnitor.se
Sat Mar 15 17:04:29 CDT 2008


Basic T.140 real-time-text is included in the 1.6 betas and the trunk.
 
Look at this message for guidance on how to get it into 1.4:
http://lists.digium.com/pipermail/asterisk-commits/2007-February/011340.html
 
There are some lines needed in sip.conf:
 
[general]

allow=t140

textsupport=yes

videosupport=yes ; Needed to make astersik capable of handling multiple
media in sdp.

 

--------

Redundancy support is apparently not out yet. It should be soon.

You should get it working without redundancy first.

Here is a link to an open source client for your trials:

http://sourceforge.net/projects/tipcon1

 

Gunnar

-------------------------------------------------------------------
Gunnar Hellström
Omnitor
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 anupam bairagi
Sent: Saturday, March 15, 2008 9:39 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] T.140 real time text


for T.140 should i need to modify chan_sip then recompile on the 1.4.18 or
just add on sip.conf
 
thanks

 
On 3/14/08, Gunnar Hellström <gunnar.hellstrom at omnitor.se> wrote: 

We are working with T.140 real time text in Asterisk 1.4 and 1.6, but so far
only in Linux on PC an IP04.
 
The support for redundancy is not included in plain 1.4, but basic support
should be there. 
so the allow=red would not be supported there.
 
Gunnar 
 
-------------------------------------------------------------------
Gunnar Hellström
Omnitor
 


  _____  

From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of anupam bairagi
Sent: Thursday, March 13, 2008 12:27 PM
To: Asterisk Developers Mailing List
Subject: [asterisk-dev] T.140 real time text

 

Anybody worked with T.140 real time text with asterisk 1.4  on centOS 5 ???
 
what are the steps ?
 
does is need modify code chan_sip on asterisk 1.4 ?
 
 

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