[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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