Skip to main content

Recommended setting for those running 1.6.x.x

Posted by eeman on Thu, 02/25/2010

for those of you running 1.6.x.x builds I suggest putting the following into your globals

ATTENDED_TRANSFER_COMPLETE_SOUND=beep


Submitted by remiq on Thu, 02/25/2010 Permalink

I don't think it officially made it to 1.6.0 branch, I see it blocked:

Repository: asterisk
Revision: 110632

_U branches/1.6.0/

------------------------------------------------------------------------
r110632 | file | 2008-03-25 10:16:05 -0500 (Tue, 25 Mar 2008) | 11 lines

Blocked revisions 110631 via svnmerge

........
r110631 | file | 2008-03-25 12:18:41 -0300 (Tue, 25 Mar 2008) | 4 lines

Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue 0009239)
Reported by: sunder

........

Looks like this is available starting in 1.6.2.

Submitted by eeman on Thu, 02/25/2010 Permalink

theres comments about it in the 1.6.1 sip.conf.sample as well as the code

[root@eeman asterisk-1.6.1.13]# grep -r ATTENDED_TRANSFER_COMPLETE_SOUND *
CHANGES: * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
configs/sip.conf.sample:;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
configs/skinny.conf.sample:;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
configs/iax.conf.sample:;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
configs/chan_dahdi.conf.sample:;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
main/features.c: const char *chan1_attended_sound = pbx_builtin_getvar_helper(transferer, "ATTENDED_TRANSFER_COMPLETE_SOUND");
main/features.c: const char *chan2_attended_sound = pbx_builtin_getvar_helper(transferee, "ATTENDED_TRANSFER_COMPLETE_SOUND");

Submitted by k3leland on Mon, 03/01/2010 Permalink

eeman,

I've heard you mention about the dtmf problems with asterisk 1.6 a couple of times now. Could you please elaborate on these or point me to the open asterisk issue?

Thanks in an advance