From 1d6b192ce0262db43b91bf736f15cd2d7f39082a Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Wed, 19 Dec 2007 08:57:45 +0000
Subject: [PATCH] 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
---
CHANGES | 2 ++
channels/chan_sip.c | 61 +++++++++++++++++++++++++++++++++--------
configs/sip.conf.sample | 42 ++++++++++++++++++++--------
3 files changed, 83 insertions(+), 22 deletions(-)
diff --git a/CHANGES b/CHANGES
index 91a6c3f3b6..0ccb9a0cbe 100644
--- 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 023922fea1..96076ac20d 100644
--- 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 d8e25e6421..78ed4806f6 100644
--- 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
--
GitLab