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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jan 30 00:12:12 CST 2009


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:              2009-01-30 00:12 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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---------------------------------------------------------------------- 
 (0099109) maxgo (reporter) - 2009-01-30 00:12
 http://bugs.digium.com/view.php?id=12569#c99109 
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phsultan: My pleasure, the jabberreceive version works great, but I found a
couple of details:


1) When I submit a sip reload, I keep getting this message, which repeat
until Asterisk crashes:

- Failed to extend from 4 to 56

- Additionally, the OPTIONS messages keep getting sent to the realtime
sipusers...

- I'm using the sipusers realtime table provided in the contrib folder as
it comes.


2) When I use the VoiceMailMain application, there are some functions that
make the sound messages to get scrambled and overlapped (switch folders,
delete messages, undelete messages).


I'm running Asterisk jabberreceive within a LAN with the VoIP phone I'm
using to hear the voicemail messages, and by seeing "top" in Linux when the
messages get scrambled, I see that CPU as well as Memory have plenty left
of resources (CPU uses only about 3% of a 2.4 GHz P4, and there are still
about 256MB of RAM free).


I've tested all the other apps I used with 1.2 and 1.4 (MySQL CDR,
Realtime Dialplan, Realtime sipusers, Realtime iaxusers, AGI, Background)
and everything is working great; only those 2 issues appear.


The * box is working as gateway of the LAN also so our LAN phones get
proxied outbound and inbound, but I tested with phones in the WAN obtaining
the same results. Registrations are working well also, no problem with NAT
in this type of config in neither side of the network.


Well, I'll appreciate very much if you could test on your side this apps
to see if you can reproduce them. I get always the same results.


Best Regards 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-30 00:12 maxgo          Note Added: 0099109                          
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