[asterisk-bugs] [Asterisk 0012569]: [branch] Receiving Text from res_jabber

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12569 
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Reported By:                eech55
Assigned To:                phsultan
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Project:                    Asterisk
Issue ID:                   12569
Category:                   Resources/res_jabber
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 115340 
Request Review:              
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Date Submitted:             2008-05-02 00:38 CDT
Last Modified:              2009-02-19 11:15 CST
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Summary:                    [branch] Receiving Text from res_jabber
Description: 
Hello,

So far, chan_gtalk sends text to GTalk clients only. There is no way to
hear input from them which limit us a lot.

It will be awesome if we are able to receive text from GTalk users, to
provide extra services such as:

1) Voice menus. The user can select menus by sending corresponding keys
2) Or, "text" menus by which the chan_gtalk replies by text messages
instead of voice

I am glad that phsultan thought about it, however very disappointed that
it was canceled.
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=8659


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Relationships       ID      Summary
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related to          0008659 [patch] Add a jabber text receiver appl...
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---------------------------------------------------------------------- 
 (0100385) jtodd (administrator) - 2009-02-19 11:15
 http://bugs.digium.com/view.php?id=12569#c100385 
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FWIW: I suspect this is a problem with core and not with this module, but
IAX2 calls lock up after a few minutes and then will not accept any further
calls via IAX2.  Just figured it would be worth mentioning.

# asterisk -rvvvc
Asterisk SVN-phsultan-jabberreceive-r176523-/trunk, Copyright (C) 1999 -
2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk SVN-phsultan-jabberreceive-r176523-/trunk currently
running on core1 (pid = 27168)
Verbosity was 0 and is now 3
    -- Executing [6060 at from-handset:2] Dial("SIP/2001-8b2980c0",
"IAX2/blork6083/6060 at from-users") in new stack
[Feb 19 17:12:12] WARNING[27168]: app_dial.c:1712 dial_exec_full: Unable
to create channel of type 'IAX2' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [6060 at from-handset:3] Congestion("SIP/2001-8b2980c0", "")
in new stack
  == Spawn extension (from-handset, 6060, 3) exited non-zero on
'SIP/2001-8b2980c0'
    -- Executing [h at from-handset:1] Hangup("SIP/2001-8b2980c0", "") in new
stack
  == Spawn extension (from-handset, h, 1) exited non-zero on
'SIP/2001-8b2980c0'
core1*CLI> 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-19 11:15 jtodd          Note Added: 0100385                          
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