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Removing whitespace that causes red dots in reviewboard

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Olle Johansson 16 years ago
parent 0f6b01f914
commit 8c56b871de
1 changed files with 561 additions and 561 deletions
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20
channels/chan_sip.c
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@ -103,8 +103,8 @@
* If someone registers a SIPS uri, this forces us to set up a TLS connection back. * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
* \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP. * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
* The tcpbindaddr config option should only be used to open ADDITIONAL ports * The tcpbindaddr config option should only be used to open ADDITIONAL ports
* So we should propably go back to * So we should propably go back to
* bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
* if tlsenable=yes, open TLS port (provided we also have cert) * if tlsenable=yes, open TLS port (provided we also have cert)
* tcpbindaddr = extra address for additional TCP connections * tcpbindaddr = extra address for additional TCP connections
@ -137,7 +137,7 @@
* \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
* devices directly from the dialplan. UDP is only a fallback if no other method works, * devices directly from the dialplan. UDP is only a fallback if no other method works,
* in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the * in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
* MTU (like INIVTE with video, audio and RTT) TCP should be preferred. * MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
* *
* When dialling unconfigured peers (with no port number) or devices in external domains * When dialling unconfigured peers (with no port number) or devices in external domains
* NAPTR records MUST be consulted to find configured transport. If they are not found, * NAPTR records MUST be consulted to find configured transport. If they are not found,
@ -619,7 +619,7 @@ static int mwi_expiry = DEFAULT_MWI_EXPIRY;
#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */ #define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */ #define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */ #define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
#define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */ #define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
#define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */ #define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
@ -641,7 +641,7 @@ static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration
static const char config[] = "sip.conf"; /*!< Main configuration file */ static const char config[] = "sip.conf"; /*!< Main configuration file */
static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */ static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
#define RTP 1 #define RTP 1
#define NO_RTP 0 #define NO_RTP 0
/*! \brief Authorization scheme for call transfers /*! \brief Authorization scheme for call transfers
@ -730,7 +730,7 @@ static const struct cfsubscription_types {
const char * const text; const char * const text;
} subscription_types[] = { } subscription_types[] = {
{ NONE, "-", "unknown", "unknown" }, { NONE, "-", "unknown", "unknown" },
/* RFC 4235: SIP Dialog event package */ /* RFC 4235: SIP Dialog event package */
{ DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" }, { DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
{ CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */ { CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
{ PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */ { PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
@ -786,13 +786,13 @@ enum sipregistrystate {
REG_STATE_REGISTERED, /*!< Registered and done */ REG_STATE_REGISTERED, /*!< Registered and done */
REG_STATE_REJECTED, /*!< Registration rejected * REG_STATE_REJECTED, /*!< Registration rejected
* \note only used when the remote party has an expire larger than * \note only used when the remote party has an expire larger than
* our max-expire. This is a final state from which we do not * our max-expire. This is a final state from which we do not
* recover (not sure how correctly). * recover (not sure how correctly).
*/ */
REG_STATE_TIMEOUT, /*!< Registration timed out * REG_STATE_TIMEOUT, /*!< Registration timed out
* \note XXX unused */ * \note XXX unused */
REG_STATE_NOAUTH, /*!< We have no accepted credentials REG_STATE_NOAUTH, /*!< We have no accepted credentials
@ -907,7 +907,7 @@ static const struct cfsip_methods {
{ SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG }, { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
{ SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG }, { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
{ SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG }, { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
{ SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG }, { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
{ SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG }, { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
{ SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG }, { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
{ SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG }, { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
@ -924,7 +924,7 @@ static const struct cfsip_methods {
}; };
/*! Define SIP option tags, used in Require: and Supported: headers /*! Define SIP option tags, used in Require: and Supported: headers
We need to be aware of these properties in the phones to use We need to be aware of these properties in the phones to use
the replace: header. We should not do that without knowing the replace: header. We should not do that without knowing
that the other end supports it... that the other end supports it...
This is nothing we can configure, we learn by the dialog This is nothing we can configure, we learn by the dialog

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