[svn-commits] russell: trunk r59211 - /trunk/channels/chan_sip.c
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Mon Mar 26 10:57:50 MST 2007
Author: russell
Date: Mon Mar 26 12:57:50 2007
New Revision: 59211
URL: http://svn.digium.com/view/asterisk?view=rev&rev=59211
Log:
Merged revisions 59209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line
Rename the new dialplan functions to match the variable name
........
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=59211&r1=59210&r2=59211
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Mar 26 12:57:50 2007
@@ -14800,7 +14800,7 @@
}
}
-static int acf_audiortpqos_read(struct ast_channel *chan, char *funcname, char *args, char *buf, size_t buflen)
+static int acf_audiortpqos_read(struct ast_channel *chan, const char *funcname, char *args, char *buf, size_t buflen)
{
struct ast_rtp_quality qos;
struct sip_pvt *p = chan->tech_pvt;
@@ -14814,11 +14814,11 @@
memset(buf, 0, buflen);
memset(&qos, 0, sizeof(qos));
- if (strcmp(funcname, "AUDIORTPQOS") == 0) {
+ if (strcmp(funcname, "RTPAUDIOQOS") == 0) {
all = ast_rtp_get_quality(p->rtp, &qos);
- } else if (strcmp(funcname, "VIDEORTPQOS") == 0) {
+ } else if (strcmp(funcname, "RTPVIDEOQOS") == 0) {
all = ast_rtp_get_quality(p->vrtp, &qos);
- } else if (strcmp(funcname, "TEXTRTPQOS") == 0) {
+ } else if (strcmp(funcname, "RTPTEXTQOS") == 0) {
all = ast_rtp_get_quality(p->trtp, &qos);
}
@@ -14886,7 +14886,7 @@
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
}
if (p->trtp) {
- textqos = ast_rtp_get_quality(p->trtp);
+ textqos = ast_rtp_get_quality(p->trtp, NULL);
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RTCPtext", "Quality:%s", textqos);
if (p->owner)
@@ -18131,7 +18131,7 @@
};
struct ast_custom_function acf_audiortpqos = {
- .name = "AUDIORTPQOS",
+ .name = "RTPAUDIOQOS",
.synopsis = "Retrieve statistics about an RTP audio stream",
.desc =
"The following statistics may be retrieved:\n"
@@ -18145,13 +18145,13 @@
" remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)",
- .syntax = "AUDIORTPQOS(<field>)",
+ .syntax = "RTPAUDIOQOS(<field>)",
.read = acf_audiortpqos_read,
};
struct ast_custom_function acf_videortpqos = {
- .name = "VIDEORTPQOS",
- .synopsis = "Retrieve statistics about an RTP audio stream",
+ .name = "RTPVIDEOQOS",
+ .synopsis = "Retrieve statistics about an RTP video stream",
.desc =
"The following statistics may be retrieved:\n"
" local_ssrc - Local SSRC (stream ID)\n"
@@ -18164,12 +18164,12 @@
" remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)",
- .syntax = "AUDIORTPQOS(<field>)",
+ .syntax = "RTPVIDEOQOS(<field>)",
.read = acf_audiortpqos_read,
};
struct ast_custom_function acf_textrtpqos = {
- .name = "TEXTRTPQOS",
+ .name = "RTPTEXTQOS",
.synopsis = "Retrieve statistics about an RTP text stream",
.desc =
"The following statistics may be retrieved:\n"
@@ -18183,7 +18183,7 @@
" remote_count - Number of transmitted packets\n"
" rtt - Round trip time\n"
" all - All statistics (in a form suited to logging, but not for parsing)",
- .syntax = "TEXTRTPQOS(<field>)",
+ .syntax = "RTPTEXTQOS(<field>)",
.read = acf_audiortpqos_read,
};
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