diff options
author | Mark Michelson <mmichelson@digium.com> | 2013-07-30 18:14:50 +0000 |
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committer | Mark Michelson <mmichelson@digium.com> | 2013-07-30 18:14:50 +0000 |
commit | 735b30ad71110c2a51404cb8686bbe3cf14b630c (patch) | |
tree | 76b1f10135c1b7f210e576be1359539de7e3476c /res/res_pjsip_rfc3326.c | |
parent | 895c8e0d2c97cd04299f3f179e99d8a3873c06c6 (diff) |
The large GULP->PJSIP renaming effort.
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_rfc3326.c')
-rw-r--r-- | res/res_pjsip_rfc3326.c | 147 |
1 files changed, 147 insertions, 0 deletions
diff --git a/res/res_pjsip_rfc3326.c b/res/res_pjsip_rfc3326.c new file mode 100644 index 000000000..66594fef5 --- /dev/null +++ b/res/res_pjsip_rfc3326.c @@ -0,0 +1,147 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2013, Digium, Inc. + * + * Joshua Colp <jcolp@digium.com> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*** MODULEINFO + <depend>pjproject</depend> + <depend>res_pjsip</depend> + <depend>res_pjsip_session</depend> + <support_level>core</support_level> + ***/ + +#include "asterisk.h" + +#include <pjsip.h> +#include <pjsip_ua.h> + +#include "asterisk/res_pjsip.h" +#include "asterisk/res_pjsip_session.h" +#include "asterisk/module.h" +#include "asterisk/causes.h" + +static void rfc3326_use_reason_header(struct ast_sip_session *session, struct pjsip_rx_data *rdata) +{ + const pj_str_t str_reason = { "Reason", 6 }; + pjsip_generic_string_hdr *header = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_reason, NULL); + char buf[20], *cause, *text; + int code; + + if (!header) { + return; + } + + ast_copy_pj_str(buf, &header->hvalue, sizeof(buf)); + cause = ast_skip_blanks(buf); + + if (strncasecmp(cause, "Q.850", 5) || !(cause = strstr(cause, "cause="))) { + return; + } + + /* If text is present get rid of it */ + if ((text = strstr(cause, ";"))) { + *text = '\0'; + } + + if (sscanf(cause, "cause=%30d", &code) != 1) { + return; + } + + ast_channel_hangupcause_set(session->channel, code & 0x7f); +} + +static int rfc3326_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata) +{ + if ((pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_bye_method) && + pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_cancel_method)) || + !session->channel) { + return 0; + } + + rfc3326_use_reason_header(session, rdata); + + return 0; +} + +static void rfc3326_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata) +{ + struct pjsip_status_line status = rdata->msg_info.msg->line.status; + + if ((status.code < 300) || !session->channel) { + return; + } + + rfc3326_use_reason_header(session, rdata); +} + +static void rfc3326_add_reason_header(struct ast_sip_session *session, struct pjsip_tx_data *tdata) +{ + char buf[20]; + + snprintf(buf, sizeof(buf), "Q.850;cause=%i", ast_channel_hangupcause(session->channel) & 0x7f); + ast_sip_add_header(tdata, "Reason", buf); + + if (ast_channel_hangupcause(session->channel) == AST_CAUSE_ANSWERED_ELSEWHERE) { + ast_sip_add_header(tdata, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\""); + } +} + +static void rfc3326_outgoing_request(struct ast_sip_session *session, struct pjsip_tx_data *tdata) +{ + if ((pjsip_method_cmp(&tdata->msg->line.req.method, &pjsip_bye_method) && + pjsip_method_cmp(&tdata->msg->line.req.method, &pjsip_cancel_method)) || + !session->channel) { + return; + } + + rfc3326_add_reason_header(session, tdata); +} + +static void rfc3326_outgoing_response(struct ast_sip_session *session, struct pjsip_tx_data *tdata) +{ + struct pjsip_status_line status = tdata->msg->line.status; + + if ((status.code < 300) || !session->channel) { + return; + } + + rfc3326_add_reason_header(session, tdata); +} + +static struct ast_sip_session_supplement rfc3326_supplement = { + .incoming_request = rfc3326_incoming_request, + .incoming_response = rfc3326_incoming_response, + .outgoing_request = rfc3326_outgoing_request, + .outgoing_response = rfc3326_outgoing_response, +}; + +static int load_module(void) +{ + ast_sip_session_register_supplement(&rfc3326_supplement); + return AST_MODULE_LOAD_SUCCESS; +} + +static int unload_module(void) +{ + ast_sip_session_unregister_supplement(&rfc3326_supplement); + return 0; +} + +AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP RFC3326 Support", + .load = load_module, + .unload = unload_module, + .load_pri = AST_MODPRI_APP_DEPEND, + ); |