diff --git a/CHANGES b/CHANGES
index 28b224ad3780a66ed5fe95f6f17f8973090a71da..7418e195b243d336bb61fe23333798a2d450d000 100644
--- a/CHANGES
+++ b/CHANGES
@@ -109,8 +109,6 @@ Application Changes
SIP Changes
-----------
- * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
- audio file to be played upon completion of an attended transfer.
* Added DNS manager support to registrations for peers referencing peer entries.
DNS manager runs in the background which allows DNS lookups to be run asynchronously
as well as periodically updating the IP address. These properties allow for
@@ -180,6 +178,13 @@ Queue changes
"QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
as well as an explanation about timeout options in general
+Miscellaneous
+-------------
+ * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
+ the 'setvar' option to cause a given audio file to be played upon completion
+ of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
+ Skinny channels only.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
------------------------------------------------------------------------------
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 746edfd2339672da171df4e0f49203b96fb17b1b..af353db1a0dc686ee488585679b319bcd9fa4b6f 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17748,17 +17748,6 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
- /* If we are performing an attended transfer and we have two channels involved then copy sound file information to play upon attended transfer completion */
- if (target.chan2) {
- const char *chan1_attended_sound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND"), *chan2_attended_sound = pbx_builtin_getvar_helper(target.chan2, "ATTENDED_TRANSFER_COMPLETE_SOUND");
- if (!ast_strlen_zero(chan1_attended_sound)) {
- pbx_builtin_setvar_helper(target.chan1, "BRIDGE_PLAY_SOUND", chan1_attended_sound);
- }
- if (!ast_strlen_zero(chan2_attended_sound)) {
- pbx_builtin_setvar_helper(target.chan2, "BRIDGE_PLAY_SOUND", chan2_attended_sound);
- }
- }
-
/* Perform the transfer */
manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Attended\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\n",
transferer->owner->name,
diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample
index 9c486b13d01c4f4c1d8f0ede91aca24c38a12214..6c86a0842808cc8e0451922befe688855b944879 100644
--- a/configs/chan_dahdi.conf.sample
+++ b/configs/chan_dahdi.conf.sample
@@ -538,7 +538,11 @@ callgroup=1
pickupgroup=1
; Channel variable to be set for all calls from this channel
-;setvar=CHANNEL=42
+;setvar=CHANNEL=42
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
+ ; cause the given audio file to
+ ; be played upon completion of
+ ; an attended transfer.
;
; Specify whether the channel should be answered immediately or if the simple
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 97faa2b98652501204ccb2980bfe863f75b325e8..bbdfdca89f48ab00b534b123339fe7a615e1bf67 100644
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -368,7 +368,10 @@ inkeys=freeworlddialup
;context=local
;auth=md5,plaintext,rsa
;secret=markpasswd
-;setvar=foo=bar
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
+ ; cause the given audio file to
+ ; be played upon completion of
+ ; an attended transfer.
;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
;transfer=no ; Disable IAX native transfer
;transfer=mediaonly ; When doing IAX native transfers, transfer
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 9bdced5c3cdf96f2b4a7d56ec48dbd3638d0478b..0dcbced76bfe793b62279aebeb6ce8133e870c24 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -961,8 +961,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;defaultuser=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
-;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will cause the given audio file to be played
- ; upon completion of an attended transfer
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
+ ; cause the given audio file to
+ ; be played upon completion of
+ ; an attended transfer.
;[pre14-asterisk]
;type=friend
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index 0cfb4fb6f466ff0e2dafcd1b71b4d98911a11d7e..389e986fd4ae881310227d9aafaacf07fd20f89f 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -87,6 +87,10 @@ keepalive=120
;nat=yes
;callerid="George W. Bush" <202-456-1414>
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
+;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
+ ; cause the given audio file to
+ ; be played upon completion of
+ ; an attended transfer.
;mailbox=500
;callwaiting=yes
;transfer=yes
diff --git a/main/features.c b/main/features.c
index 0d8ea9363ea3a2f541c3a48911eb023b3371f3f5..a7eeedb2d725bfb37d14759c617b31369adf6773 100644
--- a/main/features.c
+++ b/main/features.c
@@ -1235,6 +1235,21 @@ static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, st
l = strlen(xferto);
snprintf(xferto + l, sizeof(xferto) - l, "@%s/n", transferer_real_context); /* append context */
+
+ /* If we are performing an attended transfer and we have two channels involved then
+ copy sound file information to play upon attended transfer completion */
+ if (transferee) {
+ const char *chan1_attended_sound = pbx_builtin_getvar_helper(transferer, "ATTENDED_TRANSFER_COMPLETE_SOUND");
+ const char *chan2_attended_sound = pbx_builtin_getvar_helper(transferee, "ATTENDED_TRANSFER_COMPLETE_SOUND");
+
+ if (!ast_strlen_zero(chan1_attended_sound)) {
+ pbx_builtin_setvar_helper(transferer, "BRIDGE_PLAY_SOUND", chan1_attended_sound);
+ }
+ if (!ast_strlen_zero(chan2_attended_sound)) {
+ pbx_builtin_setvar_helper(transferee, "BRIDGE_PLAY_SOUND", chan2_attended_sound);
+ }
+ }
+
newchan = ast_feature_request_and_dial(transferer, transferee, "Local", ast_best_codec(transferer->nativeformats),
xferto, atxfernoanswertimeout, &outstate, transferer->cid.cid_num, transferer->cid.cid_name, 1, transferer->language);