[asterisk-commits] trunk - r8482 /trunk/channels/chan_sip.c

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Mon Jan 23 09:27:03 MST 2006


Author: oej
Date: Mon Jan 23 10:27:01 2006
New Revision: 8482

URL: http://svn.digium.com/view/asterisk?rev=8482&view=rev
Log:
Doxygen update

Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=8482&r1=8481&r2=8482&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jan 23 10:27:01 2006
@@ -12883,10 +12883,12 @@
 	return 0;
 }
 
-/*! \brief  sip_sipredirect: Transfer call before connect with a 302 redirect */
-/* Called by the transfer() dialplan application through the sip_transfer() */
-/* pbx interface function if the call is in ringing state */
-/* coded by Martin Pycko (m78pl at yahoo.com) */
+/*! \brief  sip_sipredirect: Transfer call before connect with a 302 redirect
+ * Called by the transfer() dialplan application through the sip_transfer()
+ * pbx interface function if the call is in ringing state 
+ * \todo Fix this function so that we wait for reply to the REFER and
+ * 	 react to errors, denials or other issues the other end might have.
+ */
 static int sip_sipredirect(struct sip_pvt *p, const char *dest)
 {
 	char *cdest;



More information about the asterisk-commits mailing list