[Asterisk-code-review] chan pjsip/res pjsip: Add CHANNEL(pjsip,request uri) (asterisk[certified/13.18])
Jenkins2
asteriskteam at digium.com
Wed Dec 13 10:40:20 CST 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/7513 )
Change subject: chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)
......................................................................
chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
---
M channels/chan_pjsip.c
M channels/pjsip/dialplan_functions.c
M include/asterisk/res_pjsip_session.h
M res/res_pjsip_session.c
4 files changed, 52 insertions(+), 11 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
Kevin Harwell: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 75dcd76..e49dd1f 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -511,6 +511,11 @@
ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
+ if (!ast_strlen_zero(exten)) {
+ /* Set provided DNID on the new channel. */
+ ast_channel_dialed(chan)->number.str = ast_strdup(exten);
+ }
+
ast_channel_priority_set(chan, 1);
ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
index ae1c265..4a75123 100644
--- a/channels/pjsip/dialplan_functions.c
+++ b/channels/pjsip/dialplan_functions.c
@@ -388,7 +388,7 @@
</enumlist>
</enum>
<enum name="target_uri">
- <para>The request URI of the <literal>INVITE</literal> request associated with the creation of this channel.</para>
+ <para>The contact URI where requests are sent.</para>
</enum>
<enum name="local_uri">
<para>The local URI.</para>
@@ -401,6 +401,10 @@
</enum>
<enum name="remote_tag">
<para>Tag in To header</para>
+ </enum>
+ <enum name="request_uri">
+ <para>The request URI of the incoming <literal>INVITE</literal>
+ associated with the creation of this channel.</para>
</enum>
<enum name="t38state">
<para>The current state of any T.38 fax on this channel.</para>
@@ -657,6 +661,27 @@
return 0;
}
+static int print_escaped_uri(struct ast_channel *chan, const char *type,
+ pjsip_uri_context_e context, const void *uri, char *buf, size_t size)
+{
+ int res;
+ char *buf_copy;
+
+ res = pjsip_uri_print(context, uri, buf, size);
+ if (res < 0) {
+ ast_log(LOG_ERROR, "Channel %s: Unescaped %s too long for %d byte buffer\n",
+ ast_channel_name(chan), type, (int) size);
+
+ /* Empty buffer that likely is not terminated. */
+ buf[0] = '\0';
+ return -1;
+ }
+
+ buf_copy = ast_strdupa(buf);
+ ast_escape_quoted(buf_copy, buf, size);
+ return 0;
+}
+
/*!
* \internal \brief Handle reading signalling information
*/
@@ -665,6 +690,7 @@
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
char *buf_copy;
pjsip_dialog *dlg;
+ int res = 0;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
@@ -690,25 +716,27 @@
return -1;
#endif
} else if (!strcmp(type, "target_uri")) {
- pjsip_uri_print(PJSIP_URI_IN_REQ_URI, dlg->target, buf, buflen);
- buf_copy = ast_strdupa(buf);
- ast_escape_quoted(buf_copy, buf, buflen);
+ res = print_escaped_uri(chan, type, PJSIP_URI_IN_REQ_URI, dlg->target, buf,
+ buflen);
} else if (!strcmp(type, "local_uri")) {
- pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri, buf, buflen);
- buf_copy = ast_strdupa(buf);
- ast_escape_quoted(buf_copy, buf, buflen);
+ res = print_escaped_uri(chan, type, PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri,
+ buf, buflen);
} else if (!strcmp(type, "local_tag")) {
ast_copy_pj_str(buf, &dlg->local.info->tag, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "remote_uri")) {
- pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->remote.info->uri, buf, buflen);
- buf_copy = ast_strdupa(buf);
- ast_escape_quoted(buf_copy, buf, buflen);
+ res = print_escaped_uri(chan, type, PJSIP_URI_IN_FROMTO_HDR,
+ dlg->remote.info->uri, buf, buflen);
} else if (!strcmp(type, "remote_tag")) {
ast_copy_pj_str(buf, &dlg->remote.info->tag, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
+ } else if (!strcmp(type, "request_uri")) {
+ if (channel->session->request_uri) {
+ res = print_escaped_uri(chan, type, PJSIP_URI_IN_REQ_URI,
+ channel->session->request_uri, buf, buflen);
+ }
} else if (!strcmp(type, "t38state")) {
ast_copy_string(buf, t38state_to_string[channel->session->t38state], buflen);
} else if (!strcmp(type, "local_addr")) {
@@ -744,7 +772,7 @@
return -1;
}
- return 0;
+ return res;
}
/*! \brief Struct used to push function arguments to task processor */
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 073cd2e..57d5631 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -159,6 +159,8 @@
unsigned int ended_while_deferred:1;
/*! DTMF mode to use with this session, from endpoint but can change */
enum ast_sip_dtmf_mode dtmf;
+ /*! Initial incoming INVITE Request-URI. NULL otherwise. */
+ pjsip_uri *request_uri;
};
typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 2b24bb7..5a962d5 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -2019,6 +2019,12 @@
ast_copy_pj_str(domain, &sip_ruri->host, size);
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
+ /*
+ * Save off the INVITE Request-URI in case it is
+ * needed: CHANNEL(pjsip,request_uri)
+ */
+ session->request_uri = pjsip_uri_clone(session->inv_session->pool, ruri);
+
return SIP_GET_DEST_EXTEN_FOUND;
}
--
To view, visit https://gerrit.asterisk.org/7513
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.18
Gerrit-MessageType: merged
Gerrit-Change-Id: I512e60d1f162395c946451becb37af3333337b33
Gerrit-Change-Number: 7513
Gerrit-PatchSet: 2
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
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