Adding the ability to specify the To: header in an outbound INVITE
by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej
parent
7c309cc622
commit
f93a8656aa
2
CHANGES
2
CHANGES
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@ -109,6 +109,8 @@ SIP changes
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* New settings for timer T1 and timer B on a global level or per device. This makes it
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possible to force timeout faster on non-responsive SIP servers. These settings are
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considered advanced, so don't use them unless you have a problem.
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* Added a dial string option to be able to set the To: header in an INVITE to any
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SIP uri.
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IAX2 changes
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------------
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@ -1053,6 +1053,7 @@ struct sip_pvt {
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AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
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AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
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AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
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AST_STRING_FIELD(todnid); /*!< DNID of this call (overrides host) */
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AST_STRING_FIELD(language); /*!< Default language for this call */
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AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
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AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
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@ -7839,17 +7840,30 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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if (p->options && p->options->uri_options)
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ast_str_append(&invite, 0, ";%s", p->options->uri_options);
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/* This is the request URI, which is the next hop of the call
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which may or may not be the destination of the call
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*/
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ast_string_field_set(p, uri, invite->str);
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if (!ast_strlen_zero(p->todnid)) {
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/*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
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if (!strchr(p->todnid, '@')) {
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/* We have no domain in the dnid */
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snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
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} else {
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snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
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}
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} else {
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if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
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/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
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snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "" : "sip:"), p->uri, p->theirtag);
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} else if (p->options && p->options->vxml_url) {
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/* If there is a VXML URL append it to the SIP URL */
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else
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snprintf(to, sizeof(to), "<%s>", p->uri);
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}
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if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
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/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
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snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "" : "sip:"), p->uri, p->theirtag);
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} else if (p->options && p->options->vxml_url) {
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/* If there is a VXML URL append it to the SIP URL */
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else
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snprintf(to, sizeof(to), "<%s>", p->uri);
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init_req(req, sipmethod, p->uri);
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/* now tmp_n is available so reuse it to build the CSeq */
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snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
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@ -7858,6 +7872,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
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/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
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* OTOH, then we won't have anything in p->route anyway */
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/* Build Remote Party-ID and From */
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if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) {
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build_rpid(p);
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@ -17304,7 +17319,13 @@ static int sip_devicestate(void *data)
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}
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/*! \brief PBX interface function -build SIP pvt structure
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SIP calls initiated by the PBX arrive here */
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SIP calls initiated by the PBX arrive here
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SIP Dial string syntax
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SIP/exten@host!dnid
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or SIP/host/exten!dnid
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or SIP/host!dnid
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*/
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static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
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{
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struct sip_pvt *p;
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@ -17312,6 +17333,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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char *ext, *host;
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char tmp[256];
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char *dest = data;
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char *dnid;
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int oldformat = format;
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/* mask request with some set of allowed formats.
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@ -17344,7 +17366,18 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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return NULL;
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}
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/* Save the destination, the SIP dial string */
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ast_copy_string(tmp, dest, sizeof(tmp));
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/* Find DNID and take it away */
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dnid = strchr(tmp, '!');
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if (dnid != NULL) {
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*dnid++ = '\0';
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ast_string_field_set(p, todnid, dnid);
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}
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/* Find at sign - @ */
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host = strchr(tmp, '@');
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if (host) {
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*host++ = '\0';
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@ -17356,6 +17389,11 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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host = tmp;
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}
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/* We now have
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host = peer name, DNS host name or DNS domain (for SRV)
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ext = extension (user part of URI)
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dnid = destination of the call (applies to the To: header)
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*/
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if (create_addr(p, host)) {
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*cause = AST_CAUSE_UNREGISTERED;
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ast_debug(3, "Cant create SIP call - target device not registred\n");
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@ -17372,7 +17410,8 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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/* We have an extension to call, don't use the full contact here */
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/* This to enable dialing registered peers with extension dialling,
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like SIP/peername/extension
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SIP/peername will still use the full contact */
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SIP/peername will still use the full contact
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*/
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if (ext) {
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ast_string_field_set(p, username, ext);
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ast_string_field_set(p, fullcontact, NULL);
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@ -1,17 +1,37 @@
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;
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; SIP Configuration example for Asterisk
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;
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; Syntax for specifying a SIP device in extensions.conf is
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; SIP/devicename where devicename is defined in a section below.
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; SIP dial strings
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;-----------------------------------------------------------
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; In the dialplan (extensions.conf) you can use several
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; syntaxes for dialing SIP devices.
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; SIP/devicename
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; SIP/username@domain (SIP uri)
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; SIP/username@host:port
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; SIP/devicename/extension
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;
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; You may also use
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; SIP/username@domain to call any SIP user on the Internet
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; (Don't forget to enable DNS SRV records if you want to use this)
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;
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; If you define a SIP proxy as a peer below, you may call
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; SIP/proxyhostname/user or SIP/user@proxyhostname
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; where the proxyhostname is defined in a section below
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;
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; Devicename
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; devicename is defined as a peer in a section below.
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;
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; username@domain
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; Call any SIP user on the Internet
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; (Don't forget to enable DNS SRV records if you want to use this)
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;
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; devicename/extension
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; If you define a SIP proxy as a peer below, you may call
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; SIP/proxyhostname/user or SIP/user@proxyhostname
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; where the proxyhostname is defined in a section below
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; This syntax also works with ATA's with FXO ports
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;
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; All of these dial strings specify the SIP request URI.
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; In addition, you can specify a specific To: header by adding an
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; exclamation mark after the dial string, like
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;
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; SIP/sales@mysipproxy!sales@edvina.net
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;
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; CLI Commands
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; -------------------------------------------------------------
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; Useful CLI commands to check peers/users:
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; sip show peers Show all SIP peers (including friends)
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; sip show users Show all SIP users (including friends)
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@ -23,7 +43,7 @@
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; Active SIP peers will not be reconfigured
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;
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; ** Deprecated options **
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; ** Deprecated configuration options **
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; The "call-limit" configuation option is deprecated. It still works in
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; this version of Asterisk, but will disappear in the next version.
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; You are encouraged to use the dialplan groupcount functionality
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