[asterisk-commits] rizzo: branch rizzo/astobj2 r51282 -
/team/rizzo/astobj2/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Jan 19 02:06:48 MST 2007
Author: rizzo
Date: Fri Jan 19 03:06:47 2007
New Revision: 51282
URL: http://svn.digium.com/view/asterisk?view=rev&rev=51282
Log:
merge from trunk 50469,
Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.
Modified:
team/rizzo/astobj2/channels/chan_sip.c
Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=51282&r1=51281&r2=51282
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Fri Jan 19 03:06:47 2007
@@ -14306,7 +14306,6 @@
/* Chan 2: Call from Asterisk to target */
int res = 0;
struct sip_pvt *targetcall_pvt;
- int error = 0;
/* Check if the call ID of the replaces header does exist locally */
if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag,
@@ -14334,38 +14333,32 @@
if (!targetcall_pvt->owner) { /* No active channel */
if (option_debug > 3)
ast_log(LOG_DEBUG, "SIP attended transfer: Error: No owner of target call\n");
- error = 1;
- }
+ /* Cancel transfer */
+ transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
+ append_history(transferer, "Xfer", "Refer failed");
+ ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
+ transferer->refer->status = REFER_FAILED;
+ sip_pvt_unlock(targetcall_pvt);
+ ast_channel_unlock(current->chan1);
+ ast_channel_unlock(targetcall_pvt->owner);
+ return -1;
+ }
+
/* We have a channel, find the bridge */
target.chan1 = targetcall_pvt->owner; /* Transferer to Asterisk */
- if (!error) {
- target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */
-
- if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) {
- /* Wrong state of new channel */
- if (option_debug > 3) {
- if (target.chan2)
- ast_log(LOG_DEBUG, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
- else if (target.chan1->_state != AST_STATE_RING)
- ast_log(LOG_DEBUG, "SIP attended transfer: Error: No target channel\n");
- else
- ast_log(LOG_DEBUG, "SIP attended transfer: Attempting transfer in ringing state\n");
- }
- if (target.chan1->_state != AST_STATE_RING)
- error = 1;
- }
- }
- if (error) { /* Cancel transfer */
- transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
- append_history(transferer, "Xfer", "Refer failed");
- ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
- transferer->refer->status = REFER_FAILED;
- sip_pvt_unlock(targetcall_pvt);
- pvt_unref(targetcall_pvt);
- ast_channel_unlock(current->chan1);
- ast_channel_unlock(target.chan1);
- return -1;
+ target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */
+
+ if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) {
+ /* Wrong state of new channel */
+ if (option_debug > 3) {
+ if (target.chan2)
+ ast_log(LOG_DEBUG, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
+ else if (target.chan1->_state != AST_STATE_RING)
+ ast_log(LOG_DEBUG, "SIP attended transfer: Error: No target channel\n");
+ else
+ ast_log(LOG_DEBUG, "SIP attended transfer: Attempting transfer in ringing state\n");
+ }
}
/* Transfer */
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