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authorJoshua Colp <jcolp@digium.com>2008-03-25 15:18:41 +0000
committerJoshua Colp <jcolp@digium.com>2008-03-25 15:18:41 +0000
commit738e4ec94e7144147b21c158764dc7d8fbbc20bf (patch)
treeb4442a43ce0035f8b86c6c864e171fcf650fca3c
parent358ac2f76a0313f1008627ab91e374e5bcaf4873 (diff)
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 #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r--CHANGES5
-rw-r--r--channels/chan_sip.c11
-rw-r--r--configs/sip.conf.sample2
-rw-r--r--main/channel.c11
4 files changed, 29 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index ba0ede212..bc224b4a8 100644
--- a/CHANGES
+++ b/CHANGES
@@ -26,6 +26,11 @@ Application Changes
of how many names are in your company. For large companies, this should be
quite helpful.
+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.
+
------------------------------------------------------------------------------
--- 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 a6e6b4bb5..128175234 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -16869,6 +16869,17 @@ 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/sip.conf.sample b/configs/sip.conf.sample
index baabc9b6e..1080777aa 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -930,6 +930,8 @@ 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
;[pre14-asterisk]
;type=friend
diff --git a/main/channel.c b/main/channel.c
index 920ee37dd..005f76ff1 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -4355,6 +4355,7 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
for (/* ever */;;) {
struct timeval now = { 0, };
int to;
+ const char *bridge_play_sound = NULL;
to = -1;
@@ -4438,6 +4439,16 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
pbx_builtin_setvar_helper(c1, "BRIDGEPVTCALLID", c0->tech->get_pvt_uniqueid(c0));
if (c1->tech->get_pvt_uniqueid)
pbx_builtin_setvar_helper(c0, "BRIDGEPVTCALLID", c1->tech->get_pvt_uniqueid(c1));
+
+ /* See if we need to play an audio file to any side of the bridge */
+ if ((bridge_play_sound = pbx_builtin_getvar_helper(c0, "BRIDGE_PLAY_SOUND"))) {
+ bridge_playfile(c0, c1, bridge_play_sound, 0);
+ pbx_builtin_setvar_helper(c0, "BRIDGE_PLAY_SOUND", NULL);
+ }
+ if ((bridge_play_sound = pbx_builtin_getvar_helper(c1, "BRIDGE_PLAY_SOUND"))) {
+ bridge_playfile(c1, c0, bridge_play_sound, 0);
+ pbx_builtin_setvar_helper(c1, "BRIDGE_PLAY_SOUND", NULL);
+ }
if (c0->tech->bridge &&
(c0->tech->bridge == c1->tech->bridge) &&