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alembic/res_pjsip: Add "webrtc" configuration option

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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 6edefc9480
commit b0c016cf6e
2 changed files with 32 additions and 1 deletions
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2
res/res_pjsip/pjsip_configuration.c
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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");