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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: http://svn.digium.com/svn/asterisk/trunk@110631 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
file 2008-03-25 15:18:41 +00:00
parent 3eb4aac765
commit 663b7622ce
4 changed files with 29 additions and 0 deletions
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5
CHANGES
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@ -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 -------------
------------------------------------------------------------------------------

11
channels/chan_sip.c
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@ -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,

2
configs/sip.conf.sample
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@ -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

11
main/channel.c
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@ -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) &&