From: Russell Bryant Date: Mon, 26 Mar 2007 17:57:50 +0000 (+0000) Subject: Merged revisions 59209 via svnmerge from X-Git-Tag: 1.6.0-beta1~3^2~2963 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=46b15992c7a515c8e5984ada9e694b0b964f9cba;p=thirdparty%2Fasterisk.git 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_sip.c b/channels/chan_sip.c index b054465a20..286bb21aab 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -14800,7 +14800,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req) } } -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 @@ static int acf_audiortpqos_read(struct ast_channel *chan, char *funcname, char * 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 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) 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 @@ static struct ast_cli_entry cli_sip[] = { }; 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 @@ struct ast_custom_function acf_audiortpqos = { " 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()", + .syntax = "RTPAUDIOQOS()", .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 @@ struct ast_custom_function acf_videortpqos = { " 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()", + .syntax = "RTPVIDEOQOS()", .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 @@ struct ast_custom_function acf_textrtpqos = { " 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()", + .syntax = "RTPTEXTQOS()", .read = acf_audiortpqos_read, };