[asterisk-bugs] [Asterisk 0014504]: [patch] Allow app_dial to provide 'tone while ringing' like 'option m' which will provide 'music on hold while dialling'

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14504 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14504
Category:                   Applications/NewFeature
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
JIRA:                       SWP-545 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-19 05:23 CST
Last Modified:              2009-12-15 11:03 CST
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Summary:                    [patch] Allow app_dial to provide 'tone while
ringing' like 'option m' which will provide 'music on hold while dialling'
Description: 
Application is to allow asterisk to be an ISDN providered to attempt
delivery of calls over Cellular Trunks connected to asterisk, if it cannot
deliver the call then signal CONGESTION to the calling party switch, who
will then respond appropriately by delivering the call out over its PSTN
connection.

Currently this can be achived, with option 'm([classs])' in the dial
command, which will indicate PROGRESS to the calling party and provide moh.
But it's just not right to play music as a ring tone.

option 'r' from app_dial indicates 'RINGING' to the calling party, which
is fine if you don't then what to indicate CONGESTION.  Because the calling
party's switch already believes asterisk can deliver the call, it won't try
an alternative.

The patch I have provided doesn't change the default functionality of
option 'r' but adds to it.
Like option 'm', which will play a 'class' of music as defined in
musiconhold.conf, option 'r' will play a 'tone' as defined in
indications.conf, it will also indicate PROGRESS to the calling party if a
tone is defined, otherwise it will still indicate RINGING. 
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Relationships       ID      Summary
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related to          0015405 Early media causes "Asked to trans...
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 (0115272) jsmith (administrator) - 2009-12-15 11:03
 https://issues.asterisk.org/view.php?id=14504#c115272 
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I tried out the patch, but didn't get any ringback tone at all.  (I'm
trying with chan_local, but can't see how that might affect the problem.)

[lab]
exten => 123,1,Dial(Local/124 at lab,,r(starwars_theme))
exten => 124,1,Progress()
exten => 124,2,Wait(30)
exten => 124,3,Goto(2)

When I dial extension 123 in the lab context, I get the following:

    -- Executing [123 at lab:1] Dial("SIP/polycom-00000003",
"Local/124 at lab,,r(starwars_theme)") in new stack
    -- Called 124 at lab
    -- Executing [124 at lab:1] NoOp("Local/124 at lab-c77b;2", "") in new
stack
    -- Executing [124 at lab:2] Wait("Local/124 at lab-c77b;2", "30") in new
stack
[Dec 15 11:01:58] WARNING[20848]: chan_sip.c:6550 sip_write: Asked to
transmit frame type slin, while native formats is 0x4 (ulaw) read/write =
0x4 (ulaw)/0x4 (ulaw)
[Dec 15 11:01:58] WARNING[20848]: chan_sip.c:6550 sip_write: Asked to
transmit frame type slin, while native formats is 0x4 (ulaw) read/write =
0x4 (ulaw)/0x4 (ulaw)
[Dec 15 11:01:58] WARNING[20848]: chan_sip.c:6550 sip_write: Asked to
transmit frame type slin, while native formats is 0x4 (ulaw) read/write =
0x4 (ulaw)/0x4 (ulaw)
[Dec 15 11:01:58] WARNING[20848]: chan_sip.c:6550 sip_write: Asked to
transmit frame type slin, while native formats is 0x4 (ulaw) read/write =
0x4 (ulaw)/0x4 (ulaw)

and I get *hundreds* of those WARNINGs scrolling across my console. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-15 11:03 jsmith         Note Added: 0115272                          
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