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: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index 91a6c3f..0ccb9a0 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -109,6 +109,8 @@ SIP changes
   * New settings for timer T1 and timer B on a global level or per device. This makes it 
     possible to force timeout faster on non-responsive SIP servers. These settings are
     considered advanced, so don't use them unless you have a problem.
+  * Added a dial string option to be able to set the To: header in an INVITE to any
+    SIP uri.
 
 IAX2 changes
 ------------

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 023922f..96076ac 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1053,6 +1053,7 @@ struct sip_pvt {
                AST_STRING_FIELD(fromuser);     /*!< User to show in the user field */
                AST_STRING_FIELD(fromname);     /*!< Name to show in the user field */
                AST_STRING_FIELD(tohost);       /*!< Host we should put in the "to" field */
+               AST_STRING_FIELD(todnid);       /*!< DNID of this call (overrides host) */
                AST_STRING_FIELD(language);     /*!< Default language for this call */
                AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
                AST_STRING_FIELD(mohsuggest);   /*!< MOH class to suggest when putting a peer on hold */
@@ -7839,17 +7840,30 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
        if (p->options && p->options->uri_options)
                ast_str_append(&invite, 0, ";%s", p->options->uri_options);
        
+       /* This is the request URI, which is the next hop of the call
+               which may or may not be the destination of the call
+       */
        ast_string_field_set(p, uri, invite->str);
+  
+       if (!ast_strlen_zero(p->todnid)) {
+               /*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
+               if (!strchr(p->todnid, '@')) {
+                       /* We have no domain in the dnid */
+                       snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
+               } else {
+                       snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
+               }
+       } else {
+               if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) { 
+                       /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
+                       snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "" : "sip:"), p->uri, p->theirtag);
+               } else if (p->options && p->options->vxml_url) {
+                       /* If there is a VXML URL append it to the SIP URL */
+                       snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
+               } else 
+                       snprintf(to, sizeof(to), "<%s>", p->uri);
+       }
 
-       if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) { 
-               /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
-               snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "" : "sip:"), p->uri, p->theirtag);
-       } else if (p->options && p->options->vxml_url) {
-               /* If there is a VXML URL append it to the SIP URL */
-               snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
-       } else 
-               snprintf(to, sizeof(to), "<%s>", p->uri);
-       
        init_req(req, sipmethod, p->uri);
        /* now tmp_n is available so reuse it to build the CSeq */
        snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
@@ -7858,6 +7872,7 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
        add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
        /* SLD: FIXME?: do Route: here too?  I think not cos this is the first request.
         * OTOH, then we won't have anything in p->route anyway */
+
        /* Build Remote Party-ID and From */
        if (ast_test_flag(&p->flags[0], SIP_SENDRPID) && (sipmethod == SIP_INVITE)) {
                build_rpid(p);
@@ -17304,7 +17319,13 @@ static int sip_devicestate(void *data)
 }
 
 /*! \brief PBX interface function -build SIP pvt structure 
-       SIP calls initiated by the PBX arrive here */
+       SIP calls initiated by the PBX arrive here 
+
+       SIP Dial string syntax
+               SIP/exten@host!dnid
+       or      SIP/host/exten!dnid
+       or      SIP/host!dnid
+*/
 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
 {
        struct sip_pvt *p;
@@ -17312,6 +17333,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
        char *ext, *host;
        char tmp[256];
        char *dest = data;
+       char *dnid;
        int oldformat = format;
 
        /* mask request with some set of allowed formats.
@@ -17344,7 +17366,18 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
                return NULL;
        }
 
+       /* Save the destination, the SIP dial string */
        ast_copy_string(tmp, dest, sizeof(tmp));
+
+
+       /* Find DNID and take it away */
+       dnid = strchr(tmp, '!');
+       if (dnid != NULL) {
+               *dnid++ = '\0';
+               ast_string_field_set(p, todnid, dnid);
+       }
+
+       /* Find at sign - @ */
        host = strchr(tmp, '@');
        if (host) {
                *host++ = '\0';
@@ -17356,6 +17389,11 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
                host = tmp;
        }
 
+       /* We now have 
+               host = peer name, DNS host name or DNS domain (for SRV) 
+               ext = extension (user part of URI)
+               dnid = destination of the call (applies to the To: header)
+       */
        if (create_addr(p, host)) {
                *cause = AST_CAUSE_UNREGISTERED;
                ast_debug(3, "Cant create SIP call - target device not registred\n");
@@ -17372,7 +17410,8 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
        /* We have an extension to call, don't use the full contact here */
        /* This to enable dialing registered peers with extension dialling,
           like SIP/peername/extension  
-          SIP/peername will still use the full contact */
+          SIP/peername will still use the full contact 
+        */
        if (ext) {
                ast_string_field_set(p, username, ext);
                ast_string_field_set(p, fullcontact, NULL);

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index d8e25e6..78ed480 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1,17 +1,37 @@
 ;
 ; SIP Configuration example for Asterisk
 ;
-; Syntax for specifying a SIP device in extensions.conf is
-; SIP/devicename where devicename is defined in a section below.
-;
-; You may also use 
-; SIP/username@domain to call any SIP user on the Internet
-; (Don't forget to enable DNS SRV records if you want to use this)
-; 
-; If you define a SIP proxy as a peer below, you may call
-; SIP/proxyhostname/user or SIP/user@proxyhostname 
-; where the proxyhostname is defined in a section below 
+; SIP dial strings
+;-----------------------------------------------------------
+; In the dialplan (extensions.conf) you can use several 
+; syntaxes for dialing SIP devices.
+;      SIP/devicename
+;      SIP/username@domain   (SIP uri)
+;      SIP/username@host:port
+;      SIP/devicename/extension
+;
+;
+; Devicename
+;      devicename is defined as a peer in a section below.
+;
+; username@domain
+;      Call any SIP user on the Internet
+;      (Don't forget to enable DNS SRV records if you want to use this)
 ; 
+; devicename/extension
+;      If you define a SIP proxy as a peer below, you may call
+;      SIP/proxyhostname/user or SIP/user@proxyhostname 
+;      where the proxyhostname is defined in a section below 
+;      This syntax also works with ATA's with FXO ports
+;
+; All of these dial strings specify the SIP request URI.
+; In addition, you can specify a specific To: header by adding an
+; exclamation mark after the dial string, like
+;
+;      SIP/sales@mysipproxy!sales@edvina.net
+;
+; CLI Commands
+; -------------------------------------------------------------
 ; Useful CLI commands to check peers/users:
 ;   sip show peers             Show all SIP peers (including friends)
 ;   sip show users             Show all SIP users (including friends)
@@ -23,7 +43,7 @@
 ;                              Active SIP peers will not be reconfigured
 ;
 
-; ** Deprecated options **
+; ** Deprecated configuration options **
 ; The "call-limit" configuation option is deprecated. It still works in
 ; this version of Asterisk, but will disappear in the next version.
 ; You are encouraged to use the dialplan groupcount functionality