diff options
author | Olle Johansson <oej@edvina.net> | 2006-03-09 17:41:38 +0000 |
---|---|---|
committer | Olle Johansson <oej@edvina.net> | 2006-03-09 17:41:38 +0000 |
commit | b27fa8bfc71c17dfbc4fe61d530fd67e20bc217e (patch) | |
tree | 4d39741379cffd6bfb3561337c761031dcdc3658 /channels | |
parent | 0752f8e41ed264b7e7abf09c6eebd45db7084aeb (diff) |
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
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 41 |
1 files changed, 26 insertions, 15 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 06ef73910..7e1f8287b 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); |