diff options
author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-16 14:31:18 +0000 |
---|---|---|
committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-16 14:31:18 +0000 |
commit | 70636d3013bd77c6808a9c478caca3f9f3b83780 (patch) | |
tree | c2159929aaf8eb42742e20ab545fd2eb9607c512 /channels | |
parent | eae9ae1b04e8097aa5b67f0bf11ee8a2f82b22f4 (diff) |
Formatting, whitespace fixes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54862 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 152 |
1 files changed, 76 insertions, 76 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index f61ea3a4f..e852a778f 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -168,7 +168,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") /* guard min must be < 1000, and should be >= 250 */ #define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */ #define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of - EXPIRY_GUARD_SECS */ + EXPIRY_GUARD_SECS */ #define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If GUARD_PCT turns out to be lower than this, it will use this time instead. @@ -210,7 +210,7 @@ static int expiry = DEFAULT_EXPIRY; /*! \brief Global jitterbuffer configuration - by default, jb is disabled */ static struct ast_jb_conf default_jbconf = { - .flags = 0, + .flags = 0, .max_size = -1, .resync_threshold = -1, .impl = "" @@ -843,12 +843,12 @@ static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_ /*! \brief T38 States for a call */ enum t38state { - T38_DISABLED = 0, /*!< Not enabled */ - T38_LOCAL_DIRECT, /*!< Offered from local */ - T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */ - T38_PEER_DIRECT, /*!< Offered from peer */ - T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */ - T38_ENABLED /*!< Negotiated (enabled) */ + T38_DISABLED = 0, /*!< Not enabled */ + T38_LOCAL_DIRECT, /*!< Offered from local */ + T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */ + T38_PEER_DIRECT, /*!< Offered from peer */ + T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */ + T38_ENABLED /*!< Negotiated (enabled) */ }; /*! \brief T.38 channel settings (at some point we need to make this alloc'ed */ @@ -862,15 +862,15 @@ struct t38properties { /*! \brief Parameters to know status of transfer */ enum referstatus { - REFER_IDLE, /*!< No REFER is in progress */ - REFER_SENT, /*!< Sent REFER to transferee */ - REFER_RECEIVED, /*!< Received REFER from transferrer */ - REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */ - REFER_ACCEPTED, /*!< Accepted by transferee */ - REFER_RINGING, /*!< Target Ringing */ - REFER_200OK, /*!< Answered by transfer target */ - REFER_FAILED, /*!< REFER declined - go on */ - REFER_NOAUTH /*!< We had no auth for REFER */ + REFER_IDLE, /*!< No REFER is in progress */ + REFER_SENT, /*!< Sent REFER to transferee */ + REFER_RECEIVED, /*!< Received REFER from transferrer */ + REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING */ + REFER_ACCEPTED, /*!< Accepted by transferee */ + REFER_RINGING, /*!< Target Ringing */ + REFER_200OK, /*!< Answered by transfer target */ + REFER_FAILED, /*!< REFER declined - go on */ + REFER_NOAUTH /*!< We had no auth for REFER */ }; static const struct c_referstatusstring { @@ -1736,7 +1736,7 @@ static int proxy_update(struct sip_proxy *proxy) { /* if it's actually an IP address and not a name, there's no need for a managed lookup */ - if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) { + if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) { /* Ok, not an IP address, then let's check if it's a domain or host */ /* XXX Todo - if we have proxy port, don't do SRV */ if (ast_get_ip_or_srv(&proxy->ip, proxy->name, global_srvlookup ? "_sip._udp" : NULL) < 0) { @@ -2361,13 +2361,13 @@ static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittyp */ static const char *find_closing_quote(const char *start, const char *lim) { - char last_char = '\0'; - const char *s; - for (s = start; *s && s != lim; last_char = *s++) { - if (*s == '"' && last_char != '\\') - break; - } - return s; + char last_char = '\0'; + const char *s; + for (s = start; *s && s != lim; last_char = *s++) { + if (*s == '"' && last_char != '\\') + break; + } + return s; } /*! \brief Pick out text in brackets from character string @@ -2388,10 +2388,10 @@ static char *get_in_brackets(char *tmp) /* * Skip any quoted text until we find the part in brackets. - * On any error give up and return the full string. - */ - while ( (first_bracket = strchr(parse, '<')) ) { - char *first_quote = strchr(parse, '"'); + * On any error give up and return the full string. + */ + while ( (first_bracket = strchr(parse, '<')) ) { + char *first_quote = strchr(parse, '"'); if (!first_quote || first_quote > first_bracket) break; /* no need to look at quoted part */ @@ -6823,27 +6823,27 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p) if (!p->owner || !ast_internal_timing_enabled(p->owner)) ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n"); - if (min_audio_packet_size) - ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size); - + if (min_audio_packet_size) + ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size); + /* XXX don't think you can have ptime for video */ - if (min_video_packet_size) - ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size); - + if (min_video_packet_size) + ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size); + /* XXX don't think you can have ptime for text */ if (min_text_packet_size) ast_build_string(&a_text_next, &a_text_left, "a=ptime:%d\r\n", min_text_packet_size); if ((m_audio_left < 2) || (m_video_left < 2) || (m_text_left < 2) || (a_audio_left == 0) || (a_video_left == 0) || (a_text_left == 0)) - ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); - + ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n"); + ast_build_string(&m_audio_next, &m_audio_left, "\r\n"); if (needvideo) ast_build_string(&m_video_next, &m_video_left, "\r\n"); if (needtext) ast_build_string(&m_text_next, &m_text_left, "\r\n"); - + len = strlen(version) + strlen(subject) + strlen(owner) + strlen(connection) + strlen(stime) + strlen(m_audio) + strlen(a_audio) + strlen(hold); if (needvideo) /* only if video response is appropriate */ len += strlen(m_video) + strlen(a_video) + strlen(bandwidth) + strlen(hold); @@ -8564,7 +8564,7 @@ static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request * /* --- We have auth, so check it */ /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting - an example in the spec of just what it is you're doing a hash on. */ + an example in the spec of just what it is you're doing a hash on. */ /* Make a copy of the response and parse it */ @@ -11047,13 +11047,13 @@ static const struct cfsubscription_types *find_subscription_type(enum subscripti /*! \brief Show active SIP channels */ static int sip_show_channels(int fd, int argc, char *argv[]) { - return __sip_show_channels(fd, argc, argv, 0); + return __sip_show_channels(fd, argc, argv, 0); } /*! \brief Show active SIP subscriptions */ static int sip_show_subscriptions(int fd, int argc, char *argv[]) { - return __sip_show_channels(fd, argc, argv, 1); + return __sip_show_channels(fd, argc, argv, 1); } /*! \brief SIP show channels CLI (main function) */ @@ -13363,7 +13363,7 @@ static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target /* We will try to connect the transferee with the target and hangup - all channels to the transferer */ + all channels to the transferer */ if (option_debug > 3) { ast_log(LOG_DEBUG, "Sip transfer:--------------------\n"); if (transferer->chan1) @@ -13520,19 +13520,19 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str /* From the RFC... A minimal, but complete, implementation can respond with a single - NOTIFY containing either the body: - SIP/2.0 100 Trying + NOTIFY containing either the body: + SIP/2.0 100 Trying - if the subscription is pending, the body: - SIP/2.0 200 OK - if the reference was successful, the body: - SIP/2.0 503 Service Unavailable - if the reference failed, or the body: - SIP/2.0 603 Declined - - if the REFER request was accepted before approval to follow the - reference could be obtained and that approval was subsequently denied - (see Section 2.4.7). + if the subscription is pending, the body: + SIP/2.0 200 OK + if the reference was successful, the body: + SIP/2.0 503 Service Unavailable + if the reference failed, or the body: + SIP/2.0 603 Declined + + if the REFER request was accepted before approval to follow the + reference could be obtained and that approval was subsequently denied + (see Section 2.4.7). If there are several REFERs in the same dialog, we need to match the ID of the event header... @@ -13994,9 +13994,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int ast_log(LOG_NOTICE, "Sending fake auth rejection for user %s\n", get_header(req, "From")); transmit_fake_auth_response(p, req, 1); } else { - ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From")); + ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From")); transmit_response_reliable(p, "403 Forbidden", req); - } + } p->invitestate = INV_COMPLETED; sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); ast_string_field_free(p, theirtag); @@ -14555,27 +14555,27 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int return res; /* If this is a blind transfer, we have the following - channels to work with: - - chan1, chan2: The current call between transferer and transferee (2 channels) - - target_channel: A new call from the transferee to the target (1 channel) - We need to stay tuned to what happens in order to be able - to bring back the call to the transferer */ + channels to work with: + - chan1, chan2: The current call between transferer and transferee (2 channels) + - target_channel: A new call from the transferee to the target (1 channel) + We need to stay tuned to what happens in order to be able + to bring back the call to the transferer */ /* If this is a attended transfer, we should have all call legs within reach: - - chan1, chan2: The call between the transferer and transferee (2 channels) - - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels) + - chan1, chan2: The call between the transferer and transferee (2 channels) + - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels) We want to bridge chan2 with targetcall_pvt! - The replaces call id in the refer message points - to the call leg between Asterisk and the transferer. - So we need to connect the target and the transferee channel - and hangup the two other channels silently + The replaces call id in the refer message points + to the call leg between Asterisk and the transferer. + So we need to connect the target and the transferee channel + and hangup the two other channels silently - If the target is non-local, the call ID could be on a remote - machine and we need to send an INVITE with replaces to the - target. We basically handle this as a blind transfer - and let the sip_call function catch that we need replaces - header in the INVITE. + If the target is non-local, the call ID could be on a remote + machine and we need to send an INVITE with replaces to the + target. We basically handle this as a blind transfer + and let the sip_call function catch that we need replaces + header in the INVITE. */ @@ -14665,7 +14665,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader); } /* Must release lock now, because it will not longer - be accessible after the transfer! */ + be accessible after the transfer! */ *nounlock = 1; ast_channel_unlock(current.chan1); ast_channel_unlock(current.chan2); @@ -14678,8 +14678,8 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int /* For blind transfer, this will lead to a new call */ /* For attended transfer to remote host, this will lead to - a new SIP call with a replaces header, if the dial plan allows it - */ + a new SIP call with a replaces header, if the dial plan allows it + */ if (!current.chan2) { /* We have no bridge, so we're talking with Asterisk somehow */ /* We need to masquerade this call */ @@ -16903,7 +16903,7 @@ static int reload_config(enum channelreloadreason reason) /* Create the dialogs list */ if (!strcasecmp(v->name, "context")) { ast_copy_string(default_context, v->value, sizeof(default_context)); - } else if (!strcasecmp(v->name, "allowguest")) { + } else if (!strcasecmp(v->name, "allowguest")) { global_allowguest = ast_true(v->value) ? 1 : 0; } else if (!strcasecmp(v->name, "realm")) { ast_copy_string(global_realm, v->value, sizeof(global_realm)); |