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Add a bunch of options from sip.conf to res_sip.conf

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For a complete list of the options added, see the review linked
at the bottom of this commit message.

(closes issue ASTERISK-21506)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2671



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2013-07-18 19:25:51 +00:00
parent 3c86832f9f
commit c47787feab
17 changed files with 749 additions and 14 deletions
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50
include/asterisk/res_sip.h
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@@ -115,6 +115,10 @@ struct ast_sip_transport {
struct ast_sockaddr external_address;
/*! Transport state information */
struct ast_sip_transport_state *state;
/*! QOS DSCP TOS bits */
unsigned int tos;
/*! QOS COS value */
unsigned int cos;
};
/*!
@@ -323,6 +327,26 @@ struct ast_sip_endpoint {
AST_STRING_FIELD(external_media_address);
/*! Configured voicemail boxes for this endpoint. Used for MWI */
AST_STRING_FIELD(mailboxes);
/*! Configured RTP engine for this endpoint. */
AST_STRING_FIELD(rtp_engine);
/*! Configured tone zone for this endpoint. */
AST_STRING_FIELD(zone);
/*! Configured language for this endpoint. */
AST_STRING_FIELD(language);
/*! Feature to enact when one-touch recording INFO with Record: On is received */
AST_STRING_FIELD(recordonfeature);
/*! Feature to enact when one-touch recording INFO with Record: Off is received */
AST_STRING_FIELD(recordofffeature);
/*! SDP origin username */
AST_STRING_FIELD(sdpowner);
/*! SDP session name */
AST_STRING_FIELD(sdpsession);
/*! Default username to place in From header */
AST_STRING_FIELD(fromuser);
/*! Domain to place in From header */
AST_STRING_FIELD(fromdomain);
/*! Username to use when sending MWI NOTIFYs to this endpoint */
AST_STRING_FIELD(mwi_from);
);
/*! Identification information for this endpoint */
struct ast_party_id id;
@@ -408,6 +432,20 @@ struct ast_sip_endpoint {
struct ast_endpoint *persistent;
/*! The number of channels at which busy device state is returned */
unsigned int devicestate_busy_at;
/*! Determines if transfers (using REFER) are allowed by this endpoint */
unsigned int allowtransfer;
/*! DSCP TOS bits for audio streams */
unsigned int tos_audio;
/*! Priority for audio streams */
unsigned int cos_audio;
/*! DSCP TOS bits for video streams */
unsigned int tos_video;
/*! Priority for video streams */
unsigned int cos_video;
/*! Indicates if endpoint is allowed to initiate subscriptions */
unsigned int allowsubscribe;
/*! The minimum allowed expiration for subscriptions from endpoint */
unsigned int subminexpiry;
};
/*!
@@ -1057,13 +1095,15 @@ struct ast_sip_body {
* \param endpoint Optional. If specified, the request will be created out-of-dialog
* to the endpoint.
* \param uri Optional. If specified, the request will be sent to this URI rather
* this value.
* than one configured for the endpoint.
* \param[out] tdata The newly-created request
* \retval 0 Success
* \retval -1 Failure
*/
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
struct ast_sip_endpoint *endpoint, const char *uri, pjsip_tx_data **tdata);
struct ast_sip_endpoint *endpoint, const char *uri,
pjsip_tx_data **tdata);
/*!
* \brief General purpose method for sending a SIP request
@@ -1308,4 +1348,12 @@ void ast_sip_report_auth_success(struct ast_sip_endpoint *endpoint, pjsip_rx_dat
*/
void ast_sip_report_auth_challenge_sent(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pjsip_tx_data *tdata);
void ast_sip_initialize_global_headers(void);
void ast_sip_destroy_global_headers(void);
int ast_sip_add_global_request_header(const char *name, const char *value, int replace);
int ast_sip_add_global_response_header(const char *name, const char *value, int replace);
int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery);
#endif /* _RES_SIP_H */