From b0c016cf6e0bcbe743f4f8286fb9b5ded830ccf7 Mon Sep 17 00:00:00 2001 From: Kevin Harwell Date: Wed, 2 Aug 2017 09:43:56 -0500 Subject: alembic/res_pjsip: Add "webrtc" configuration option When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b --- res/res_pjsip/pjsip_configuration.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'res/res_pjsip/pjsip_configuration.c') diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index 4e12b04cc..d3ff1f32b 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1994,7 +1994,7 @@ int ast_res_pjsip_initialize_configuration(void) ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_audio_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_audio_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_video_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_video_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "bundle", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.bundle)); - ast_sorcery_object_field_register(sip_sorcery, "endpoint", "webrtc", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.webrtc)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "webrtc", "no", OPT_YESNO_T, 1, FLDSET(struct ast_sip_endpoint, media.webrtc)); if (ast_sip_initialize_sorcery_transport()) { ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n"); -- cgit v1.2.3