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Updated doc for OSP lookup application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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3
CHANGES
3
CHANGES
@@ -133,6 +133,9 @@ Applications
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during the Dial will be set to the argument passed to the 'f' option.
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* Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
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Gosub on the peer.
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* The OSP lookup application adds in/outbound network ID, optional security,
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number portability, QoS reporting, destination IP port, custom info and service
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type features.
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Dialplan Functions
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------------------
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@@ -67,6 +67,14 @@ From 1.6.2 to 1.8:
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working that will no longer work.
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Please see doc/externalivr.txt for details.
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* OSP lookup application changes following variable names:
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OSPPEERIP to OSPINPEERIP
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OSPTECH to OSPOUTTECH
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OSPDEST to OSPDESTINATION
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OSPCALLING to OSPOUTCALLING
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OSPCALLED to OSPOUTCALLED
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OSPRESULTS to OSPDESTREMAILS
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From 1.6.1 to 1.6.2:
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* SIP no longer sends the 183 progress message for early media by
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@@ -222,6 +222,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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<variable name="OSPOUTCALLIDTYPES">
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<para>The outbound Call-ID types.</para>
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</variable>
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<variable name="OSPOUTCALLID">
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<para>The outbound Call-ID. Only for H.323.</para>
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</variable>
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<variable name="OSPDIALSTR">
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<para>The outbound Dial command string.</para>
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</variable>
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@@ -908,7 +911,7 @@ static int osp_get_provider(
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* \param name OSP provider context name
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* \param trans OSP transaction handle, output
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* \param source Source of provider, output
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* \param sourcesize Size of source buffer, in
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* \param srcsize Size of source buffer, in
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* \return OSK_OK Success, OSK_FAILED Failed, OSP_ERROR Error
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*/
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static int osp_create_transaction(
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@@ -2910,7 +2913,6 @@ static int osp_load(int reload)
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osp_security = 1;
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}
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ast_debug(1, "OSP: osp_security '%d'\n", osp_security);
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if ((cvar = ast_variable_retrieve(cfg, OSP_GENERAL_CAT, "tokenformat"))) {
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if ((sscanf(cvar, "%30d", &ivar) == 1) &&
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