Add the ability to play a courtesy tone to the transfer target in a native SIP attend...
authorTerry Wilson <twilson@digium.com>
Mon, 8 Dec 2008 16:02:42 +0000 (16:02 +0000)
committerTerry Wilson <twilson@digium.com>
Mon, 8 Dec 2008 16:02:42 +0000 (16:02 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES
channels/chan_sip.c

diff --git a/CHANGES b/CHANGES
index 82d6bd5..9d0d3e6 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -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
 --------------
index a5ec482..e230c51 100644 (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);