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:
Kevin Harwell 2017-08-02 09:43:56 -05:00
parent 2be8d91c0f
commit 521b6fed12
2 changed files with 32 additions and 1 deletions

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@ -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')

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@ -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");