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Add the ability to play a courtesy tone to the transfer target in a native SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson 2008-12-08 16:02:42 +00:00
parent bc03323251
commit f6dda1e544
2 changed files with 6 additions and 0 deletions
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2
CHANGES
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@ -28,6 +28,8 @@ SIP Changes
* Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
option is enabled, a SIP channel will go to the fax extension (if it exists)
after T38 is negotiated. This option is disabled by default.
* If ATTENDED_TRANSFER_COMPLETE_SOUND is set, the sound will be played to the
target of an attended transfer
Skinny Changes
--------------

4
channels/chan_sip.c
View File

@ -18927,11 +18927,15 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
} else {
/* Transfer succeeded! */
const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");
/* Tell transferer that we're done. */
transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
append_history(transferer, "Xfer", "Refer succeeded");
transferer->refer->status = REFER_200OK;
if (target.chan2 && !ast_strlen_zero(xfersound) && ast_streamfile(target.chan2, xfersound, target.chan2->language) >= 0) {
ast_waitstream(target.chan2, "");
}
if (targetcall_pvt->owner) {
ast_debug(1, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
ast_channel_unlock(targetcall_pvt->owner);