From 1c45a32ee861fa427e0243abe03c729966fa4436 Mon Sep 17 00:00:00 2001 From: Kevin Harwell Date: Fri, 22 Nov 2013 17:27:55 +0000 Subject: 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 --- res/res_pjsip.c | 80 ++++++++++++++++++++++++++++----------------------------- 1 file changed, 40 insertions(+), 40 deletions(-) (limited to 'res/res_pjsip.c') 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 @@ Media Codec(s) to disallow - + DTMF mode This setting allows to choose the DTMF mode for endpoint communication. @@ -247,7 +247,7 @@ Mailbox(es) to be associated with - + Default Music On Hold class @@ -388,49 +388,49 @@ to indicate ringing and will NOT send it as audio. - + The numeric pickup groups for a channel. Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). - + The numeric pickup groups that a channel can pickup. Can be set to a comma separated list of numbers or ranges between the values of 0-63 (maximum of 64 groups). - + The named pickup groups for a channel. Can be set to a comma separated list of case sensitive strings limited by supported line length. - + The named pickup groups that a channel can pickup. Can be set to a comma separated list of case sensitive strings limited by supported line length. - + The number of in-use channels which will cause busy to be returned as device state 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. - + Whether T.38 UDPTL support is enabled or not If set to yes T.38 UDPTL support will be enabled, and T.38 negotiation requests will be accepted and relayed. - + T.38 UDPTL error correction method @@ -446,34 +446,34 @@ - + T.38 UDPTL maximum datagram size This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. - + Whether CNG tone detection is enabled This option can be set to send the session to the fax extension when a CNG tone is detected. - + Whether NAT support is enabled on UDPTL sessions When enabled the UDPTL stack will send UDPTL packets to the source address of received packets. - + Whether IPv6 is used for UDPTL Sessions When enabled the UDPTL stack will use IPv6. - + Set which country's indications to use for channels created for this endpoint. @@ -486,7 +486,7 @@ recordofffeature - + The feature to enact when one-touch recording is turned on. When an INFO request for one-touch recording arrives with a Record header set to "on", this @@ -499,7 +499,7 @@ recordofffeature - + The feature to enact when one-touch recording is turned off. When an INFO request for one-touch recording arrives with a Record header set to "off", this @@ -512,16 +512,16 @@ recordonfeature - + Name of the RTP engine to use for channels created for this endpoint - + Determines whether SIP REFER transfers are allowed for this endpoint - + String placed as the username portion of an SDP origin (o=) line. - + String used for the SDP session (s=) line. @@ -548,29 +548,29 @@ See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings - + Determines if endpoint is allowed to initiate subscriptions with Asterisk. - + The minimum allowed expiry time for subscriptions initiated by the endpoint. - + Username to use in From header for requests to this endpoint. - + Username to use in From header for unsolicited MWI NOTIFYs to this endpoint. - + Domain to user in From header for requests to this endpoint. - + Verify that the provided peer certificate is valid This option only applies if media_encryption is set to dtls. - + Interval at which to renegotiate the TLS session and rekey the SRTP session This option only applies if media_encryption is @@ -579,21 +579,21 @@ If this is not set or the value provided is 0 rekeying will be disabled. - + Path to certificate file to present to peer This option only applies if media_encryption is set to dtls. - + Path to private key for certificate file This option only applies if media_encryption is set to dtls. - + Cipher to use for DTLS negotiation This option only applies if media_encryption is @@ -603,21 +603,21 @@ http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS - + Path to certificate authority certificate This option only applies if media_encryption is set to dtls. - + Path to a directory containing certificate authority certificates This option only applies if media_encryption is set to dtls. - + Whether we are willing to accept connections, connect to the other party, or both. @@ -767,7 +767,7 @@ - + Network to consider local (used for NAT purposes). This must be in CIDR or dotted decimal format with the IP and mask separated with a slash ('/'). @@ -775,7 +775,7 @@ Password required for transport - + Private key file (TLS ONLY) @@ -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. - + Set transaction timer T1 value (milliseconds). 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. - + Set transaction timer B value (milliseconds). 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. - + Use the short forms of common SIP header names. @@ -995,13 +995,13 @@ The settings in this section are global. Unlike options in the system section, these options can be refreshed by performing a reload. - + Value used in Max-Forwards header for SIP requests. Must be of type 'global'. - + Value used in User-Agent header for SIP requests and Server header for SIP responses. -- cgit v1.2.3