[asterisk-commits] res pjsip: Add contact user to endpoint (asterisk[certified/13.1])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 17 13:18:16 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: res_pjsip: Add contact_user to endpoint
......................................................................
res_pjsip: Add contact_user to endpoint
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests. This may not work
on scheduled qualify requests where we haven't looked up the endpoint.
Change-Id: I7ce6b6c6678f66807885da1d42fb5fd6909ae55a
---
M CHANGES
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
6 files changed, 88 insertions(+), 0 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
Joshua Colp: Looks good to me, but someone else must approve; Verified
diff --git a/CHANGES b/CHANGES
index 8258b25..396eba9 100644
--- a/CHANGES
+++ b/CHANGES
@@ -9,6 +9,16 @@
==============================================================================
------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.1.0-cert8 to Asterisk 13.1-cert9 --
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * A new endpoint configuration parameter 'contact_user' has been added which
+ when set will override the default user set on Contact headers in outgoing
+ requests.
+
+------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.1.0-cert4 to Asterisk 13.1-cert8 --
------------------------------------------------------------------------------
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index 8ab6543..e97a395 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -725,6 +725,8 @@
;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified
; number of seconds when the channel is on hold (default:
; "0" or not enabled)
+;contact_user= ; On outgoing requests, force the user portion of the Contact
+ ; header to this value (default: "")
;==========================AUTH SECTION OPTIONS=========================
;[auth]
diff --git a/contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py b/contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py
new file mode 100644
index 0000000..6ac6103
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py
@@ -0,0 +1,22 @@
+"""Add contact_user to endpoint
+
+Revision ID: 4e2493ef32e6
+Revises: 28ce1e718f05
+Create Date: 2016-08-16 14:19:58.918466
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '4e2493ef32e6'
+down_revision = '28ce1e718f05'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+ op.add_column('ps_endpoints', sa.Column('contact_user', sa.String(80)))
+
+
+def downgrade():
+ op.drop_column('ps_endpoints', 'contact_user')
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 0ce5091..cffb78a 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -619,6 +619,8 @@
struct ast_variable *channel_vars;
/*! Whether to place a 'user=phone' parameter into the request URI if user is a number */
unsigned int usereqphone;
+ /*! Override the user on the outgoing Contact header with this value. */
+ char *contact_user;
};
/*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index d5b68bc..7ad47e5 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -775,6 +775,12 @@
channel is hung up. By default this option is set to 0, which means do not check.
</para></description>
</configOption>
+ <configOption name="contact_user" default="">
+ <synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
+ <description><para>
+ On outbound requests, force the user portion of the Contact header to this value.
+ </para></description>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>
@@ -2381,7 +2387,15 @@
/* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
+
dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
+
+ if (!ast_strlen_zero(endpoint->contact_user)) {
+ pjsip_sip_uri *sip_uri;
+
+ sip_uri = pjsip_uri_get_uri(dlg->local.contact->uri);
+ pj_strdup2(dlg->pool, &sip_uri->user, endpoint->contact_user);
+ }
/* If a request user has been specified and we are permitted to change it, do so */
if (!ast_strlen_zero(request_user)) {
@@ -2638,6 +2652,18 @@
return -1;
}
+ if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
+ pjsip_contact_hdr *contact_hdr;
+ pjsip_sip_uri *contact_uri;
+ static const pj_str_t HCONTACT = { "Contact", 7 };
+
+ contact_hdr = pjsip_msg_find_hdr_by_name((*tdata)->msg, &HCONTACT, NULL);
+ if (contact_hdr) {
+ contact_uri = pjsip_uri_get_uri(contact_hdr->uri);
+ pj_strdup2(pool, &contact_uri->user, endpoint->contact_user);
+ }
+ }
+
/* Add the user=phone parameter if applicable */
ast_sip_add_usereqphone(endpoint, (*tdata)->pool, (*tdata)->msg->line.req.uri);
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 866bfaa..5275e5f 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -877,6 +877,30 @@
return 0;
}
+static int contact_user_handler(const struct aco_option *opt,
+ struct ast_variable *var, void *obj)
+{
+ struct ast_sip_endpoint *endpoint = obj;
+
+ endpoint->contact_user = ast_strdup(var->value);
+ if (!endpoint->contact_user) {
+ return -1;
+ }
+
+ return 0;
+}
+
+static int contact_user_to_str(const void *obj, const intptr_t *args, char **buf)
+{
+ const struct ast_sip_endpoint *endpoint = obj;
+
+ *buf = ast_strdup(endpoint->contact_user);
+ if (!(*buf)) {
+ return -1;
+ }
+
+ return 0;
+}
static void *sip_nat_hook_alloc(const char *name)
{
@@ -1747,6 +1771,7 @@
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, set_var_to_vl, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "message_context", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, message_context));
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "accountcode", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, accountcode));
+ ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
if (ast_sip_initialize_sorcery_transport()) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
@@ -1878,6 +1903,7 @@
endpoint->pickup.named_pickupgroups = ast_unref_namedgroups(endpoint->pickup.named_pickupgroups);
ao2_cleanup(endpoint->persistent);
ast_variables_destroy(endpoint->channel_vars);
+ ast_free(endpoint->contact_user);
}
static int init_subscription_configuration(struct ast_sip_endpoint_subscription_configuration *subscription)
--
To view, visit https://gerrit.asterisk.org/3573
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Gerrit-MessageType: merged
Gerrit-Change-Id: I7ce6b6c6678f66807885da1d42fb5fd6909ae55a
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: certified/13.1
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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