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chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL URIs

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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
This commit is contained in:
Matthew Jordan
2014-04-17 19:50:05 +00:00
parent 3043cd363d
commit 7d26eefce4
3 changed files with 12 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
CHANGES
View File

@@ -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

13
channels/chan_sip.c
View File

@@ -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 {

1
channels/sip/include/sip.h
View File

@@ -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 */