diff options
| -rw-r--r-- | CHANGES | 6 | ||||
| -rw-r--r-- | configs/samples/pjsip.conf.sample | 2 | ||||
| -rw-r--r-- | contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py | 22 | ||||
| -rw-r--r-- | include/asterisk/res_pjsip.h | 2 | ||||
| -rw-r--r-- | res/res_pjsip.c | 26 | ||||
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 27 |
6 files changed, 85 insertions, 0 deletions
@@ -37,6 +37,12 @@ chan_pjsip dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP channel to be re-negotiated and updated after session set up. +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 13.10.0 to Asterisk 13.11.0 ---------- ------------------------------------------------------------------------------ diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample index 2851945ed..994c8e01c 100644 --- a/configs/samples/pjsip.conf.sample +++ b/configs/samples/pjsip.conf.sample @@ -760,6 +760,8 @@ ;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 000000000..f91cff04e --- /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: 3772f8f828da +Create Date: 2016-08-16 14:19:58.918466 + +""" + +# revision identifiers, used by Alembic. +revision = '4e2493ef32e6' +down_revision = '3772f8f828da' + +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 72a2ff351..002182b9a 100644 --- a/include/asterisk/res_pjsip.h +++ b/include/asterisk/res_pjsip.h @@ -751,6 +751,8 @@ struct ast_sip_endpoint { struct ast_acl_list *contact_acl; /*! The number of seconds into call to disable fax detection. (0 = disabled) */ unsigned int faxdetect_timeout; + /*! 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 aafb3a211..96c07d501 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -913,6 +913,12 @@ then the <replaceable>context</replaceable> setting is used. </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> @@ -2869,8 +2875,16 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, /* 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)) { pjsip_sip_uri *sip_uri; @@ -3172,6 +3186,18 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s 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 9e757e230..c3012c4b2 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1211,6 +1211,31 @@ static int voicemail_extension_to_str(const void *obj, const intptr_t *args, cha 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) { return ast_sorcery_generic_alloc(sizeof(struct ast_sip_nat_hook), NULL); @@ -1907,6 +1932,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_permit", "", endpoint_acl_handler, NULL, NULL, 0, 0); ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_acl", "", endpoint_acl_handler, contact_acl_to_str, NULL, 0, 0); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subscribe_context", "", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct ast_sip_endpoint, subscription.context)); + 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"); @@ -2038,6 +2064,7 @@ static void endpoint_destructor(void* obj) ao2_cleanup(endpoint->persistent); ast_variables_destroy(endpoint->channel_vars); AST_VECTOR_FREE(&endpoint->ident_method_order); + ast_free(endpoint->contact_user); } static int init_subscription_configuration(struct ast_sip_endpoint_subscription_configuration *subscription) |