[asterisk-commits] oej: branch oej/calleridutf8 r89552 - in /team/oej/calleridutf8: ./ channels/...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Nov 25 05:13:33 CST 2007


Author: oej
Date: Sun Nov 25 05:13:32 2007
New Revision: 89552

URL: http://svn.digium.com/view/asterisk?view=rev&rev=89552
Log:
Update to trunk

Modified:
    team/oej/calleridutf8/   (props changed)
    team/oej/calleridutf8/channels/chan_sip.c
    team/oej/calleridutf8/include/asterisk/channel.h

Propchange: team/oej/calleridutf8/
------------------------------------------------------------------------------
--- automerge (original)
+++ automerge Sun Nov 25 05:13:32 2007
@@ -1,1 +1,1 @@
-Sponsor chan_sip3 - visit http://www.codename-pineapple.org
+http://www.codename-pineapple.org/

Propchange: team/oej/calleridutf8/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Nov 25 05:13:32 2007
@@ -1,1 +1,1 @@
-/trunk:1-89548
+/trunk:1-89551

Modified: team/oej/calleridutf8/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/calleridutf8/channels/chan_sip.c?view=diff&rev=89552&r1=89551&r2=89552
==============================================================================
--- team/oej/calleridutf8/channels/chan_sip.c (original)
+++ team/oej/calleridutf8/channels/chan_sip.c Sun Nov 25 05:13:32 2007
@@ -12217,7 +12217,7 @@
 		e->command = "sip show history";
 		e->usage =
 			"Usage: sip show history <call-id>\n"
-			"       Provides detailed dialog history on a given SIP call (specified by call-hid).\n";
+			"       Provides detailed dialog history on a given SIP call (specified by call-id).\n";
 		return NULL;
 	case CLI_GENERATE:
 		return complete_sip_show_history(a->line, a->word, a->pos, a->n);

Modified: team/oej/calleridutf8/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/team/oej/calleridutf8/include/asterisk/channel.h?view=diff&rev=89552&r1=89551&r2=89552
==============================================================================
--- team/oej/calleridutf8/include/asterisk/channel.h (original)
+++ team/oej/calleridutf8/include/asterisk/channel.h Sun Nov 25 05:13:32 2007
@@ -207,9 +207,6 @@
  * SIP and IAX2 has utf8 encoded Unicode caller ID names.
  * In some cases, we also have an alternative (RPID) E.164 number that can be used
  * as caller ID on numeric E.164 phone networks (zaptel or SIP/IAX2 to pstn gateway).
-
- * SIP and IAX2 will prefer the cid_utf8 if it exists, otherwise use the cid_name.
- * \todo Implement this in chan_sip.c and chan_iax2.c
  *
  * \todo Implement settings for transliteration between UTF8 caller ID names in
  *       to Ascii Caller ID's (Zaptel). Östen Åsklund might be transliterated into
@@ -223,9 +220,6 @@
 	char *cid_name;		/*!< Malloc'd Caller Name (ASCII) */
 	char *cid_ani;		/*!< Malloc'd ANI */
 	char *cid_rdnis;	/*!< Malloc'd RDNIS */
-	char *cid_utf8;		/*!< Malloc'd Caller ID name in utf8 */
-	char *cid_domain;	/*!< Malloc'd Caller ID domain (ascii. IDN supported) */
-	char *cid_e164;		/*!< Malloc'd Alternative Caller ID E.164 (alternative to SIP/IAX2 utf8 uri, RPID) */
 	int cid_pres;		/*!< Callerid presentation/screening */
 	int cid_ani2;		/*!< Callerid ANI 2 (Info digits) */
 	int cid_ton;		/*!< Callerid Type of Number */




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