diff options
author | Kevin Harwell <kharwell@digium.com> | 2013-11-22 17:27:55 +0000 |
---|---|---|
committer | Kevin Harwell <kharwell@digium.com> | 2013-11-22 17:27:55 +0000 |
commit | 1c45a32ee861fa427e0243abe03c729966fa4436 (patch) | |
tree | 47e0020d224c7e7f6fd0e7537da00c4d9b358a5e /res/res_pjsip.c | |
parent | 2147e3930380b599a0cdab6a8533f0b3d39d0091 (diff) |
res_pjsip: convert configuration settings names to snake case
Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore). For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...
Review: https://reviewboard.asterisk.org/r/3002/
........
Merged revisions 403022 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip.c')
-rw-r--r-- | res/res_pjsip.c | 80 |
1 files changed, 40 insertions, 40 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 3f6fd8c69..cda22a3f5 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -189,7 +189,7 @@ <configOption name="disallow"> <synopsis>Media Codec(s) to disallow</synopsis> </configOption> - <configOption name="dtmfmode" default="rfc4733"> + <configOption name="dtmf_mode" default="rfc4733"> <synopsis>DTMF mode</synopsis> <description> <para>This setting allows to choose the DTMF mode for endpoint communication.</para> @@ -247,7 +247,7 @@ <configOption name="mailboxes"> <synopsis>Mailbox(es) to be associated with</synopsis> </configOption> - <configOption name="mohsuggest" default="default"> + <configOption name="moh_suggest" default="default"> <synopsis>Default Music On Hold class</synopsis> </configOption> <configOption name="outbound_auth"> @@ -388,49 +388,49 @@ to indicate ringing and will NOT send it as audio. </para></description> </configOption> - <configOption name="callgroup"> + <configOption name="call_group"> <synopsis>The numeric pickup groups for a channel.</synopsis> <description><para> Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). </para></description> </configOption> - <configOption name="pickupgroup"> + <configOption name="pickup_group"> <synopsis>The numeric pickup groups that a channel can pickup.</synopsis> <description><para> Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). </para></description> </configOption> - <configOption name="namedcallgroup"> + <configOption name="named_call_group"> <synopsis>The named pickup groups for a channel.</synopsis> <description><para> Can be set to a comma separated list of case sensitive strings limited by supported line length. </para></description> </configOption> - <configOption name="namedpickupgroup"> + <configOption name="named_pickup_group"> <synopsis>The named pickup groups that a channel can pickup.</synopsis> <description><para> Can be set to a comma separated list of case sensitive strings limited by supported line length. </para></description> </configOption> - <configOption name="devicestate_busy_at" default="0"> + <configOption name="device_state_busy_at" default="0"> <synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis> <description><para> When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the PJSIP channel driver will return busy as the device state instead of in use. </para></description> </configOption> - <configOption name="t38udptl" default="no"> + <configOption name="t38_udptl" default="no"> <synopsis>Whether T.38 UDPTL support is enabled or not</synopsis> <description><para> If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted and relayed. </para></description> </configOption> - <configOption name="t38udptl_ec" default="none"> + <configOption name="t38_udptl_ec" default="none"> <synopsis>T.38 UDPTL error correction method</synopsis> <description> <enumlist> @@ -446,34 +446,34 @@ </enumlist> </description> </configOption> - <configOption name="t38udptl_maxdatagram" default="0"> + <configOption name="t38_udptl_maxdatagram" default="0"> <synopsis>T.38 UDPTL maximum datagram size</synopsis> <description><para> This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. </para></description> </configOption> - <configOption name="faxdetect" default="no"> + <configOption name="fax_detect" default="no"> <synopsis>Whether CNG tone detection is enabled</synopsis> <description><para> This option can be set to send the session to the fax extension when a CNG tone is detected. </para></description> </configOption> - <configOption name="t38udptl_nat" default="no"> + <configOption name="t38_udptl_nat" default="no"> <synopsis>Whether NAT support is enabled on UDPTL sessions</synopsis> <description><para> When enabled the UDPTL stack will send UDPTL packets to the source address of received packets. </para></description> </configOption> - <configOption name="t38udptl_ipv6" default="no"> + <configOption name="t38_udptl_ipv6" default="no"> <synopsis>Whether IPv6 is used for UDPTL Sessions</synopsis> <description><para> When enabled the UDPTL stack will use IPv6. </para></description> </configOption> - <configOption name="tonezone"> + <configOption name="tone_zone"> <synopsis>Set which country's indications to use for channels created for this endpoint.</synopsis> </configOption> <configOption name="language"> @@ -486,7 +486,7 @@ <ref type="configOption">recordofffeature</ref> </see-also> </configOption> - <configOption name="recordonfeature" default="automixmon"> + <configOption name="record_on_feature" default="automixmon"> <synopsis>The feature to enact when one-touch recording is turned on.</synopsis> <description> <para>When an INFO request for one-touch recording arrives with a Record header set to "on", this @@ -499,7 +499,7 @@ <ref type="configOption">recordofffeature</ref> </see-also> </configOption> - <configOption name="recordofffeature" default="automixmon"> + <configOption name="record_off_feature" default="automixmon"> <synopsis>The feature to enact when one-touch recording is turned off.</synopsis> <description> <para>When an INFO request for one-touch recording arrives with a Record header set to "off", this @@ -512,16 +512,16 @@ <ref type="configOption">recordonfeature</ref> </see-also> </configOption> - <configOption name="rtpengine" default="asterisk"> + <configOption name="rtp_engine" default="asterisk"> <synopsis>Name of the RTP engine to use for channels created for this endpoint</synopsis> </configOption> - <configOption name="allowtransfer" default="yes"> + <configOption name="allow_transfer" default="yes"> <synopsis>Determines whether SIP REFER transfers are allowed for this endpoint</synopsis> </configOption> - <configOption name="sdpowner" default="-"> + <configOption name="sdp_owner" default="-"> <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis> </configOption> - <configOption name="sdpsession" default="Asterisk"> + <configOption name="sdp_session" default="Asterisk"> <synopsis>String used for the SDP session (s=) line.</synopsis> </configOption> <configOption name="tos_audio"> @@ -548,29 +548,29 @@ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings </para></description> </configOption> - <configOption name="allowsubscribe" default="yes"> + <configOption name="allow_subscribe" default="yes"> <synopsis>Determines if endpoint is allowed to initiate subscriptions with Asterisk.</synopsis> </configOption> - <configOption name="subminexpiry" default="60"> + <configOption name="sub_min_expiry" default="60"> <synopsis>The minimum allowed expiry time for subscriptions initiated by the endpoint.</synopsis> </configOption> - <configOption name="fromuser"> + <configOption name="from_user"> <synopsis>Username to use in From header for requests to this endpoint.</synopsis> </configOption> - <configOption name="mwifromuser"> + <configOption name="mwi_from_user"> <synopsis>Username to use in From header for unsolicited MWI NOTIFYs to this endpoint.</synopsis> </configOption> - <configOption name="fromdomain"> + <configOption name="from_domain"> <synopsis>Domain to user in From header for requests to this endpoint.</synopsis> </configOption> - <configOption name="dtlsverify"> + <configOption name="dtls_verify"> <synopsis>Verify that the provided peer certificate is valid</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is set to <literal>dtls</literal>. </para></description> </configOption> - <configOption name="dtlsrekey"> + <configOption name="dtls_rekey"> <synopsis>Interval at which to renegotiate the TLS session and rekey the SRTP session</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is @@ -579,21 +579,21 @@ If this is not set or the value provided is 0 rekeying will be disabled. </para></description> </configOption> - <configOption name="dtlscertfile"> + <configOption name="dtls_cert_file"> <synopsis>Path to certificate file to present to peer</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is set to <literal>dtls</literal>. </para></description> </configOption> - <configOption name="dtlsprivatekey"> + <configOption name="dtls_private_key"> <synopsis>Path to private key for certificate file</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is set to <literal>dtls</literal>. </para></description> </configOption> - <configOption name="dtlscipher"> + <configOption name="dtls_cipher"> <synopsis>Cipher to use for DTLS negotiation</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is @@ -603,21 +603,21 @@ http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS </para></description> </configOption> - <configOption name="dtlscafile"> + <configOption name="dtls_ca_file"> <synopsis>Path to certificate authority certificate</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is set to <literal>dtls</literal>. </para></description> </configOption> - <configOption name="dtlscapath"> + <configOption name="dtls_ca_path"> <synopsis>Path to a directory containing certificate authority certificates</synopsis> <description><para> This option only applies if <replaceable>media_encryption</replaceable> is set to <literal>dtls</literal>. </para></description> </configOption> - <configOption name="dtlssetup"> + <configOption name="dtls_setup"> <synopsis>Whether we are willing to accept connections, connect to the other party, or both.</synopsis> <description> <para> @@ -767,7 +767,7 @@ </enumlist> </description> </configOption> - <configOption name="localnet"> + <configOption name="local_net"> <synopsis>Network to consider local (used for NAT purposes).</synopsis> <description><para>This must be in CIDR or dotted decimal format with the IP and mask separated with a slash ('/').</para></description> @@ -775,7 +775,7 @@ <configOption name="password"> <synopsis>Password required for transport</synopsis> </configOption> - <configOption name="privkey_file"> + <configOption name="priv_key_file"> <synopsis>Private key file (TLS ONLY)</synopsis> </configOption> <configOption name="protocol" default="udp"> @@ -952,7 +952,7 @@ before the SIP stack is initialized. The only way to reset these values is to either restart Asterisk, or unload res_pjsip.so and then load it again. </para></description> - <configOption name="timert1" default="500"> + <configOption name="timer_t1" default="500"> <synopsis>Set transaction timer T1 value (milliseconds).</synopsis> <description><para> Timer T1 is the base for determining how long to wait before retransmitting @@ -960,7 +960,7 @@ For more information on this timer, see RFC 3261, Section 17.1.1.1. </para></description> </configOption> - <configOption name="timerb" default="32000"> + <configOption name="timer_b" default="32000"> <synopsis>Set transaction timer B value (milliseconds).</synopsis> <description><para> Timer B determines the maximum amount of time to wait after sending an INVITE @@ -969,7 +969,7 @@ this timer, see RFC 3261, Section 17.1.1.1. </para></description> </configOption> - <configOption name="compactheaders" default="no"> + <configOption name="compact_headers" default="no"> <synopsis>Use the short forms of common SIP header names.</synopsis> </configOption> <configOption name="threadpool_initial_size" default="0"> @@ -995,13 +995,13 @@ The settings in this section are global. Unlike options in the <literal>system</literal> section, these options can be refreshed by performing a reload. </para></description> - <configOption name="maxforwards" default="70"> + <configOption name="max_forwards" default="70"> <synopsis>Value used in Max-Forwards header for SIP requests.</synopsis> </configOption> <configOption name="type"> <synopsis>Must be of type 'global'.</synopsis> </configOption> - <configOption name="useragent" default="Asterisk <Asterisk Version>"> + <configOption name="user_agent" default="Asterisk <Asterisk Version>"> <synopsis>Value used in User-Agent header for SIP requests and Server header for SIP responses.</synopsis> </configOption> </configObject> |