diff options
author | root <root@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-03-12 16:08:23 +0000 |
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committer | root <root@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-03-12 16:08:23 +0000 |
commit | ffe54a30e43921c53f80978270fb038cb2098352 (patch) | |
tree | ef4f1e24a84dce7603a3594937ec0f59147e848c | |
parent | 044a75c7228d0d15f012f6d887cfbbb6551e6c66 (diff) |
automerge commit
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.2-netsec@12538 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | channels/chan_sip.c | 61 |
1 files changed, 39 insertions, 22 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 6a62ea5c6..844b7b2bb 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -2485,13 +2485,15 @@ static int sip_hangup(struct ast_channel *ast) if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) { if (needcancel) { /* Outgoing call, not up */ if (ast_test_flag(p, SIP_OUTGOING)) { + /* stop retransmitting an INVITE that has not received a response */ + __sip_pretend_ack(p); + + /* Send a new request: CANCEL */ transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0); /* Actually don't destroy us yet, wait for the 487 on our original INVITE, but do set an autodestruct just in case we never get it. */ ast_clear_flag(&locflags, SIP_NEEDDESTROY); - sip_scheddestroy(p, 15000); - /* stop retransmitting an INVITE that has not received a response */ - __sip_pretend_ack(p); + sip_scheddestroy(p, 32000); if ( p->initid != -1 ) { /* channel still up - reverse dec of inUse counter only if the channel is not auto-congested */ @@ -2521,12 +2523,34 @@ 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; + 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; - char *codec; + int res = 0; struct sip_pvt *p = ast->tech_pvt; ast_mutex_lock(&p->lock); @@ -2534,19 +2558,7 @@ 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) @@ -4580,6 +4592,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r respprep(&resp, p, msg, req); if (p->rtp) { ast_rtp_offered_from_local(p->rtp, 0); + 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); @@ -9329,6 +9342,8 @@ static char *function_sippeer(struct ast_channel *chan, char *cmd, char *data, c snprintf(buf, len, "%d", peer->call_limit); } else if (!strcasecmp(colname, "curcalls")) { snprintf(buf, len, "%d", peer->inUse); + } else if (!strcasecmp(colname, "accountcode")) { + ast_copy_string(buf, peer->accountcode, len); } else if (!strcasecmp(colname, "useragent")) { ast_copy_string(buf, peer->useragent, len); } else if (!strcasecmp(colname, "mailbox")) { @@ -9388,6 +9403,7 @@ struct ast_custom_function sippeer_function = { "- curcalls Current amount of calls \n" " Only available if call-limit is set\n" "- language Default language for peer\n" + "- accountcode Account code for this peer\n" "- useragent Current user agent id for peer\n" "- codec[x] Preferred codec index number 'x' (beginning with zero).\n" "\n" @@ -9549,12 +9565,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru break; case 183: /* Session progress */ sip_cancel_destroy(p); + /* Ignore 183 Session progress without SDP */ if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) { process_sdp(p, req); - } - if (!ignore && p->owner) { - /* Queue a progress frame */ - ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + if (!ignore && p->owner) { + /* Queue a progress frame */ + ast_queue_control(p->owner, AST_CONTROL_PROGRESS); + } } break; case 200: /* 200 OK on invite - someone's answering our call */ |