[asterisk-bugs] [LibSS7 0013495]: [patch] isup timers + q.764 compatibility + new cli commands

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13495 
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Reported By:                adomjan
Assigned To:                mattf
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Project:                    LibSS7
Issue ID:                   13495
Category:                   New/Feature
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 140434 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-16 12:43 CDT
Last Modified:              2008-12-03 10:52 CST
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Summary:                    [patch] isup timers + q.764 compatibility + new cli
commands
Description: 
This path introduce isup timers, + handle the abnormal situation as the
q.764 require.

not complete, isup_gra and the others have to move to call oriantated
model.

enable isup timers, put those lines after the sigchan lines(values just
example for testing):

isup_timer.t1 = 3000
isup_timer.t5 = 7000
isup_timer.t7 = 20000

isup_timer.t12 = 5000 
isup_timer.t13 = 17000
isup_timer.t14 = 5000 
isup_timer.t15 = 17000

isup_timer.t16 = 5000
isup_timer.t17 = 17000

isup_timer.t18 = 5000
isup_timer.t19 = 17000

isup_timer.t20 = 5000
isup_timer.t21 = 17000

isup_timer.t22 = 5000
isup_timer.t23 = 17000

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Relationships       ID      Summary
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related to          0012036 [patch] RFC 3372 SIP-T receive implemen...
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 (0095728) adomjan (reporter) - 2008-12-03 10:52
 http://bugs.digium.com/view.php?id=13495#c95728 
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welcome, please apply this trivial patch too!

The echo control stuff seems ok, but I think should turn off the
echocancel on fax call too.
The octasic dsp-s able to detect the fax tone?

I think we should do some cleanups too. Like removing point code from mtp2
link structure, because it is available in adjecent_sp structure.

link->dpc -> link->adj_sp->adjpc

the linkset terminology is not correct in the chan_dahdi.conf I think,
because in our linkset we handle more linksets :) 
The correct terminology would be ss7_network.

In chan_dahdi: start_pbx after the setting up the channel variables bug
present in the pri part too.
I run in this bug, when I tested the new variables and the 1st application
in the dialplan was DumpChan(), was not set up the variables...
when in the dialplan I wrote:
NoOP
DumpChan 
it was ok

another remark: using tons of SS7_ channel variables are ugly thing.
In feature asterisk versions (1.8.0, 2.0.0) would be nice if asterisk
handle many ss7 attributes internally like CALLERID() function/object, and
map of them as many as possible the other channel types (SIP, PRI), and
path through all of them via IAX2. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-03 10:52 adomjan        Note Added: 0095728                          
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