mirror of
				https://github.com/asterisk/asterisk.git
				synced 2025-10-31 18:55:19 +00:00 
			
		
		
		
	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
This commit is contained in:
		| @@ -0,0 +1,31 @@ | |||||||
|  | """add webrtc option to ps_endpoints | ||||||
|  |  | ||||||
|  | Revision ID: 44ccced114ce | ||||||
|  | Revises: 164abbd708c | ||||||
|  | Create Date: 2017-07-10 17:07:25.926150 | ||||||
|  |  | ||||||
|  | """ | ||||||
|  |  | ||||||
|  | # revision identifiers, used by Alembic. | ||||||
|  | revision = '44ccced114ce' | ||||||
|  | down_revision = '164abbd708c' | ||||||
|  |  | ||||||
|  | from alembic import op | ||||||
|  | import sqlalchemy as sa | ||||||
|  | from sqlalchemy.dialects.postgresql import ENUM | ||||||
|  |  | ||||||
|  | YESNO_NAME = 'yesno_values' | ||||||
|  | YESNO_VALUES = ['yes', 'no'] | ||||||
|  |  | ||||||
|  | def upgrade(): | ||||||
|  |     ############################# Enums ############################## | ||||||
|  |  | ||||||
|  |     # yesno_values have already been created, so use postgres enum object | ||||||
|  |     # type to get around "already created" issue - works okay with mysql | ||||||
|  |     yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False) | ||||||
|  |  | ||||||
|  |     op.add_column('ps_endpoints', sa.Column('webrtc', yesno_values)) | ||||||
|  |  | ||||||
|  |  | ||||||
|  | def downgrade(): | ||||||
|  |     op.drop_column('ps_endpoints', 'webrtc') | ||||||
| @@ -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_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", "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", "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()) { | 	if (ast_sip_initialize_sorcery_transport()) { | ||||||
| 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n"); | 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n"); | ||||||
|   | |||||||
		Reference in New Issue
	
	Block a user