diff options
Diffstat (limited to 'channels/sip')
| -rw-r--r-- | channels/sip/include/globals.h | 2 | ||||
| -rw-r--r-- | channels/sip/include/sip.h | 15 |
2 files changed, 8 insertions, 9 deletions
diff --git a/channels/sip/include/globals.h b/channels/sip/include/globals.h index 0bd2f4d2d..d7c9f13d0 100644 --- a/channels/sip/include/globals.h +++ b/channels/sip/include/globals.h @@ -28,7 +28,7 @@ extern struct ast_sockaddr bindaddr; /*!< UDP: The address we bind to */ extern struct ast_sched_context *sched; /*!< The scheduling context */ /*! \brief Definition of this channel for PBX channel registration */ -extern const struct ast_channel_tech sip_tech; +extern struct ast_channel_tech sip_tech; /*! \brief This version of the sip channel tech has no send_digit_begin * callback so that the core knows that the channel does not want diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index 57c155e14..e5d8205e5 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -216,7 +216,6 @@ #define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */ #define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */ #define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */ -#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263); #endif /*@}*/ @@ -695,7 +694,7 @@ struct sip_settings { char default_context[AST_MAX_CONTEXT]; char default_subscribecontext[AST_MAX_CONTEXT]; struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */ - format_t capability; /*!< Supported codecs */ + struct ast_format_cap *caps; /*!< Supported codecs */ int tcp_enabled; int default_max_forwards; /*!< Default max forwards (SIP Anti-loop) */ }; @@ -995,13 +994,13 @@ struct sip_pvt { unsigned int sipoptions; /*!< Supported SIP options on the other end */ unsigned int reqsipoptions; /*!< Required SIP options on the other end */ struct ast_codec_pref prefs; /*!< codec prefs */ - format_t capability; /*!< Special capability (codec) */ - format_t jointcapability; /*!< Supported capability at both ends (codecs) */ - format_t peercapability; /*!< Supported peer capability */ - format_t prefcodec; /*!< Preferred codec (outbound only) */ + struct ast_format_cap *caps; /*!< Special capability (codec) */ + struct ast_format_cap *jointcaps; /*!< Supported capability at both ends (codecs) */ + struct ast_format_cap *peercaps; /*!< Supported peer capability */ + struct ast_format_cap *redircaps; /*!< Redirect codecs */ + struct ast_format_cap *prefcaps; /*!< Preferred codec (outbound only) */ int noncodeccapability; /*!< DTMF RFC2833 telephony-event */ int jointnoncodeccapability; /*!< Joint Non codec capability */ - format_t redircodecs; /*!< Redirect codecs */ int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */ int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */ int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */ @@ -1217,7 +1216,7 @@ struct sip_peer { int maxcallbitrate; /*!< Maximum Bitrate for a video call */ int expire; /*!< When to expire this peer registration */ - format_t capability; /*!< Codec capability */ + struct ast_format_cap *caps; /*!< Codec capability */ int rtptimeout; /*!< RTP timeout */ int rtpholdtimeout; /*!< RTP Hold Timeout */ int rtpkeepalive; /*!< Send RTP packets for keepalive */ |