[asterisk-bugs] [Asterisk 0009239]: [patch] sip attended transfer - xfersound

[noreply at bugs.digium.com]

The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=9239 
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Reported By:                sunder
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   9239
Category:                   Channels/chan_sip/Transfers
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     feedback
Asterisk Version:            1.2.15 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             03-08-2007 12:42 CST
Last Modified:              03-11-2008 15:20 CDT
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Summary:                    [patch] sip attended transfer - xfersound
Description: 
I have had multiple requests for asterisk to play an xfersound sound when
the call is bridged during a attend transfer, from our customers. So I sat
down it wrote it. I works great so far, but I had to make some changes to
channel.c, channel.h, and chan_sip.so. The code that I wrote is mainly a
proof of concept. 
I would like everyone to review the code real quick to see if this I will
break anything else,  if I am going about it the wrong way, or if i need to
change something. The way that I wrote this, it will easily be able to move
the code over to other channels, especially IAX. I have developed this on
asterisk-1.2.15, since I am the most familiar with 1.2, and have not
touched 1.4 yet. After I get thing nailed down and working properly I will
get it working in the trunk version and then submit it to bugs.digium.com.


Plays a ?beep? only on the phone that is receiving the transferred call.

Related bug tracker item http://bugs.digium.com/view.php?id=3819 , I found
this after I wrote the code.


How it works.
File: channels.h
	Added xfersound variable to the ast_channel structure

File: Chan_sip.c
	Added xfersound variable to the ast_channel structure


	If there is a attend transfer request set sip_pvt->xfersound to 1
	Then the function attempt_transfer() is call to try and bridge the
channels.
		If the sip_pvt->xfersound flag is set to 1 
Then set ast_channel->xfersound to 1

File: Channels.c
	During ast_generic_bridge()
	During the infinite ?for(;;)? loop
		If the xfersound flag is set in the ast_channel structure of either
peer. 
                       Bridge_playfile(?beep?)
		




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Relationships       ID      Summary
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related to          0003819 [request] xfersound = beep   for SIP tr...
related to          0011900 Call-bridged Macro feature request
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---------------------------------------------------------------------- 
 file - 03-11-08 15:20  
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What about a dialplan variable? 

Issue History 
Date Modified   Username       Field                    Change               
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03-11-08 15:20  file           Note Added: 0083760                          
03-11-08 15:20  file           Status                   acknowledged => feedback
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