From b27fa8bfc71c17dfbc4fe61d530fd67e20bc217e Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Thu, 9 Mar 2006 17:41:38 +0000
Subject: [PATCH] Support SIP_CODEC channel variable for early media. (Imported
from 1.2, with a small change for const char* channel variables)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 41 ++++++++++++++++++++++++++---------------
1 file changed, 26 insertions(+), 15 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 06ef739104..7e1f8287bf 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2605,12 +2605,34 @@ static int sip_hangup(struct ast_channel *ast)
return 0;
}
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+ int fmt;
+ const char *codec;
+
+ codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+ if (!codec)
+ return;
+
+ fmt = ast_getformatbyname(codec);
+ if (fmt) {
+ ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n", codec);
+ if (p->jointcapability & fmt) {
+ p->jointcapability &= fmt;
+ p->capability &= fmt;
+ } else
+ ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+ } else
+ ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
+ return;
+}
+
/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
* Part of PBX interface */
static int sip_answer(struct ast_channel *ast)
{
- int res = 0,fmt;
- const char *codec;
+ int res = 0;
struct sip_pvt *p = ast->tech_pvt;
ast_mutex_lock(&p->lock);
@@ -2618,19 +2640,7 @@ static int sip_answer(struct ast_channel *ast)
#ifdef OSP_SUPPORT
time(&p->ospstart);
#endif
-
- codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
- if (codec) {
- fmt=ast_getformatbyname(codec);
- if (fmt) {
- ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
- if (p->jointcapability & fmt) {
- p->jointcapability &= fmt;
- p->capability &= fmt;
- } else
- ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
- } else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
- }
+ try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
if (option_debug)
@@ -4671,6 +4681,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
}
respprep(&resp, p, msg, req);
if (p->rtp) {
+ try_suggested_sip_codec(p);
add_sdp(&resp, p);
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
--
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