chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL URIs

This patch is a continuation of https://reviewboard.asterisk.org/r/3349/,
committed in r412303.

It resolves a finding oej had that the phone-context be available in a
channel variable separate from SIPDOMAIN. This patch adds that variable as
SIPURIPHONECONTEXT. It also allows a local number (or global number specified
in the TEL URI) to be used to look up as a peer.

(issue ASTERISK-17179)

Review: https://reviewboard.asterisk.org/r/3349/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412467 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index b4e0d4e..dc26105 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -98,7 +98,7 @@ chan_sip
 -------------------------
  * TEL URI support for inbound INVITE requests has been added. chan_sip will
    now handle TEL schemes in the Request and From URIs. The phone-context in
-   the Request URI will be stored in the TELPHONECONTEXT channel variable on
+   the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
    the inbound channel.
 
 Debugging

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index eb0a5f2..e63657f 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8249,6 +8249,9 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
        if (!ast_strlen_zero(i->domain)) {
                pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
        }
+       if (!ast_strlen_zero(i->tel_phone_context)) {
+               pbx_builtin_setvar_helper(tmp, "SIPURIPHONECONTEXT", i->tel_phone_context);
+       }
        if (!ast_strlen_zero(i->callid)) {
                pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
        }
@@ -17694,6 +17697,12 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
 
        extract_host_from_hostport(&domain);
 
+       if (strncasecmp(get_in_brackets(tmp), "tel:", 4)) {
+               ast_string_field_set(p, domain, domain);
+       } else {
+               ast_string_field_set(p, tel_phone_context, domain);
+       }
+
        if (ast_strlen_zero(uri)) {
                /*
                 * Either there really was no extension found or the request
@@ -17703,8 +17712,6 @@ static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_re
                uri = "s";
        }
 
-       ast_string_field_set(p, domain, domain);
-
        /* Now find the From: caller ID and name */
        /* XXX Why is this done in get_destination? Isn't it already done?
           Needs to be checked
@@ -18358,7 +18365,7 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
                if (!peer) {
                        char *uri_tmp, *callback = NULL, *dummy;
                        uri_tmp = ast_strdupa(uri2);
-                       parse_uri(uri_tmp, "sip:,sips:", &callback, &dummy, &dummy, &dummy);
+                       parse_uri(uri_tmp, "sip:,sips:,tel:", &callback, &dummy, &dummy, &dummy);
                        if (!ast_strlen_zero(callback) && (peer = sip_find_peer_by_ip_and_exten(&p->recv, callback, p->socket.type))) {
                                ; /* found, fall through */
                        } else {

diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 4ea20ea..9737806 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -1038,6 +1038,7 @@ struct sip_pvt {
                AST_STRING_FIELD(last_presence_subtype);   /*!< The last presence subtype sent for a subscription. */
                AST_STRING_FIELD(last_presence_message);   /*!< The last presence message for a subscription */
                AST_STRING_FIELD(msg_body);     /*!< Text for a MESSAGE body */
+               AST_STRING_FIELD(tel_phone_context);       /*!< The phone-context portion of a TEL URI */
        );
        char via[128];                          /*!< Via: header */
        int maxforwards;                        /*!< SIP Loop prevention */