From 2e00b5edbdc8ef310670d693b0f6456c2fd9692e Mon Sep 17 00:00:00 2001 From: George Joseph Date: Mon, 14 Feb 2022 06:31:25 -0700 Subject: [PATCH] Makefile: Allow XML documentation to exist outside source files Moved the xmldoc build logic from the top-level Makefile into its own script "make_xml_documentation" in the build_tools directory. Created a new utility script "get_sourceable_makeopts", also in the build_tools directory, that dumps the top-level "makeopts" file in a format that can be "sourced" from shell sscripts. This allows scripts to easily get the values of common make build variables such as the location of the GREP, SED, AWK, etc. utilities as well as the AST* and library *_LIB and *_INCLUDE variables. Besides moving logic out of the Makefile, some optimizations were done like removing "third-party" from the list of subdirectories to be searched for documentation and changing some assignments from "=" to ":=" so they're only evaluated once. The speed increase is noticeable. The makeopts.in file was updated to include the paths to REALPATH and DIRNAME. The ./conifgure script was setting them but makeopts.in wasn't including them. So... With this change, you can now place documentation in any"c" source file AND you can now place it in a separate XML file altogether. The following are examples of valid locations: res/res_pjsip.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_configuration.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_doc.xml A fully-formed XML file. The "configInfo", "manager", "managerEvent", etc. elements that would be in the "c" file DOCUMENTATION fragment should be wrapped in proper XML. Example for "somemodule.xml": ... It's the "appdocsxml.dtd" that tells make_xml_documentation that this is a documentation XML file and not some other XML file. It also allows many XML-capable editors to do formatting and validation. Other than the ".xml" suffix, the name of the file is not significant. As a start... This change also moves the documentation that was in res_pjsip.c to 2 new XML files in res/res_pjsip: pjsip_config.xml and pjsip_manager.xml. This cut the number of lines in res_pjsip.c in half. :) Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade --- Makefile | 58 +- build_tools/get_sourceable_makeopts | 54 + build_tools/make_xml_documentation | 247 ++ makeopts.in | 2 + res/res_pjsip.c | 3242 --------------------------- res/res_pjsip/pjsip_config.xml | 2346 +++++++++++++++++++ res/res_pjsip/pjsip_manager.xml | 900 ++++++++ 7 files changed, 3560 insertions(+), 3289 deletions(-) create mode 100755 build_tools/get_sourceable_makeopts create mode 100755 build_tools/make_xml_documentation create mode 100644 res/res_pjsip/pjsip_config.xml create mode 100644 res/res_pjsip/pjsip_manager.xml diff --git a/Makefile b/Makefile index 234c2f6d5c..bb8cdd041a 100644 --- a/Makefile +++ b/Makefile @@ -475,60 +475,24 @@ endif $(INSTALL) -m 644 $$x "$(DESTDIR)$(ASTDATADIR)/rest-api" ; \ done -ifeq ($(GREP),) -else ifeq ($(GREP),:) -else - XML_core_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null)) -endif - +DOC_MOD_SUBDIRS := $(filter-out third-party,$(MOD_SUBDIRS)) +XML_core_en_US := $(shell build_tools/make_xml_documentation --command=print_dependencies --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)") +# core-en_US.xml is the normal documentation created with asterisk builds. doc/core-en_US.xml: makeopts .lastclean $(XML_core_en_US) - @printf "Building Documentation For: " - @echo "" > $@ - @echo "" >> $@ - @echo "" >> $@ - @echo "" >> $@ - @for x in $(MOD_SUBDIRS); do \ - printf "$$x " ; \ - for i in `find $$x -name '*.c'`; do \ - MODULEINFO=$$($(AWK) -f build_tools/get_moduleinfo $$i) ; \ - if [ -n "$$MODULEINFO" ] ; \ - then \ - echo "" >> $@ ; \ - echo "$$MODULEINFO" >> $@ ; \ - echo "" >> $@ ; \ - fi ; \ - $(AWK) -f build_tools/get_documentation $$i >> $@ ; \ - done ; \ - done - @echo - @echo "" >> $@ + @build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \ + --with-moduleinfo --validate --output-file=$@ -ifeq ($(GREP),) -else ifeq ($(GREP),:) -else - XML_full_en_US = $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null)) -endif - -doc/full-en_US.xml: makeopts .lastclean $(XML_full_en_US) +# The full-en_US.xml target is only called by the wiki documentation generation process +# and does special post-processing in preparation for uploading to the wiki. +# It creates full-en_US.xml but then re-creates core-en_US.xml as well. +doc/full-en_US.xml: makeopts .lastclean $(XML_core_en_US) ifeq ($(PYTHON),:) @echo "--------------------------------------------------------------------------" @echo "--- Please install python to build full documentation ---" @echo "--------------------------------------------------------------------------" else - @printf "Building Documentation For: " - @echo "" > $@ - @echo "" >> $@ - @echo "" >> $@ - @echo "" >> $@ - @for x in $(filter-out third-party,$(MOD_SUBDIRS)); do \ - printf "$$x " ; \ - for i in `find $$x -name '*.c'`; do \ - $(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \ - done ; \ - done - @echo - @echo "" >> $@ - @$(PYTHON) build_tools/post_process_documentation.py -i $@ -o "doc/core-en_US.xml" + @build_tools/make_xml_documentation --command=create_xml --source-tree=. --mod-subdirs="$(DOC_MOD_SUBDIRS)" \ + --for-wiki --validate --output-file=$@ --core-output-file=./doc/core-en_US.xml endif validate-docs: doc/core-en_US.xml diff --git a/build_tools/get_sourceable_makeopts b/build_tools/get_sourceable_makeopts new file mode 100755 index 0000000000..fbf4c38fc6 --- /dev/null +++ b/build_tools/get_sourceable_makeopts @@ -0,0 +1,54 @@ +#!/bin/sh +PROGNAME="${0##*/}" + +if [ "$1" = "-h" ] || [ "$1" = "--help" ] ; then + cat <<-EOF + Usage: ${PROGNAME}: [ ] [ ] + + This script takes an Asterisk makeopts file, or any file containing + "make" style variable assignments, and converts it into a format + that can be directly 'sourced' by shell scripts. + + * Any spaces around the equals sign are removed. + * The variable value is quoted. + * The "make" "or" command is evaluated. + + Both input and output files are optional and will default to + stdin and stdout respectively. + + NOTE: This script relies on NO external commands and only POSIX + constructs. It should be runnable by any shell. + EOF + exit 1 +fi + +input_file="/dev/stdin" +if [ "$1" != "" ] ; then + input_file="$1" +fi + +output_file="/dev/stdout" +if [ "$2" != "" ] ; then + output_file="$2" +fi + +# orfunc is a code fragment to be added to the outp[ut file. +# We don't WANT the variables evaluated. +# shellcheck disable=SC2016 +orfunc='or (){ before="${1%,*}" ; after="${1#*,}" ; if [ "$before" = "" ] ; then echo "${after}" ; else echo "${before}" ; fi ; }' +echo "${orfunc}" >"${output_file}" + +while read -r LINE ; do + var="${LINE%%=*}" + if [ "${var}" != "" ] ; then + val="${LINE#*=}" + if [ "${val}" != "${var}" ] ; then + if [ "${val%% *}" = "" ] ; then + echo "${var% *}=\"${val#* }\"" + else + echo "${var% *}=\"${val}\"" + fi + fi + fi +done <"${input_file}" >>"${output_file}" + diff --git a/build_tools/make_xml_documentation b/build_tools/make_xml_documentation new file mode 100755 index 0000000000..fafb81c376 --- /dev/null +++ b/build_tools/make_xml_documentation @@ -0,0 +1,247 @@ +#!/bin/sh +# The GREP, SED, FIND, etc variables are all set at run time from +# makeopts. +# shellcheck disable=SC2154 + +PROGNAME="${0##*/}" +PROGDIR="${0%/*}" + +# Fail on errors +set -e + +usage() { +cat <<-EOF + Usage: ${PROGNAME} --command=(create_xml | print_dependencies) + --source-tree= [ --mod-subdirs= ] + [ --with-moduleinfo ] [--for-wiki ] [ --validate ] + [ --output-file= ] + [ --core-output-file= ] + + command: + print_dependencies: Print the source files that have documentation + for use by "make" as dependencies. + create_xml: Create the actual XML output file. + + source-tree: The path to the Asterisk source tree. + + mod-subdirs: A quoted, space-separated list of module sub-directories + to search for documentation. Defaults to + "channels pbx apps codecs formats cdr cel bridges funcs tests main res addons" + + with-moduleinfo: Include the "MODULEINFO" block from source files. + Default is to not include MODULEINFO + + for-wiki: Perform special post processing for wiki documentation. + This creates two output files and therefore needs both + and . + Default is to not perform wiki post-processing. + + validate: Run xmllint or xmlstarlet to validate output-file. + + output-file: The XML file to write to if the command was + "create_xml". + + core-output-file: The additional XML file to write to if the command was + "create_xml" with "for-wiki". + +EOF +} + +with_moduleinfo=0 +for_wiki=0 +validate=0 +command="" +mod_subdirs="channels pbx apps codecs formats cdr cel bridges funcs tests main res addons" +source_tree="" +output_file="" +core_output_file="" + +for arg in "$@" ; do + case ${arg} in + --for-wiki) + for_wiki=1 + ;; + --with-moduleinfo) + with_moduleinfo=1 + ;; + --validate) + validate=1 + ;; + --command=*) + command=${arg#*=} + ;; + --source-tree=*) + source_tree=${arg#*=} + ;; + --mod-subdirs=*) + mod_subdirs="${arg#*=}" + ;; + --output-file=*) + output_file=${arg#*=} + ;; + --core-output-file=*) + core_output_file=${arg#*=} + ;; + -h|--help) + usage + exit 0 + ;; + *) + echo "unknown option '${arg}'." + usage + exit 1 + ;; + esac +done + +if [ "${command}" = "" ] ; then + echo "No command specified" + usage + exit 1 +fi + +if [ "${source_tree}" = "" ] ; then + echo "No source-tree specified" + usage + exit 1; +fi + +if [ ! -d "${source_tree}" ] ; then + echo "Asterisk source tree '${source_tree}' doesn't exist." + exit 1 +fi + +if [ ! -f "${source_tree}/Makefile" ] ; then + echo "There's no 'Makefile' in '${source_tree}'." + exit 1 +fi + +if [ ! -f "${source_tree}/makeopts" ] ; then + echo "There's no 'makeopts' in '${source_tree}'. Maybe you need to run ./configure?" + exit 1 +fi + +# This will get the paths to the utilities we need, all +# of which will be in makeopts. We need to convert the +# format so it's sourceable. +tmpname="/tmp/ast_makeopts.$$.env" +trap 'rm "$tmpname" >/dev/null 2>&1' INT QUIT TERM EXIT +"${PROGDIR}/get_sourceable_makeopts" "${source_tree}/makeopts" >"${tmpname}" +# The file to be sourced is generated at run time and can't be checked. +# shellcheck disable=SC1090 +. "${tmpname}" +rm "${tmpname}" > /dev/null 2>&1 || : +trap - INT QUIT TERM EXIT + +# Make sure we have everything we need. +for c in GREP FIND AWK DIRNAME BASENAME SED CAT ; do + bin=$(eval "echo \${${c}}") + if [ "${bin}" = "" ] ; then + echo "The '${c}' utility was not found." + exit 1 + fi +done + +if [ "${for_wiki}" -eq "1" ] || [ "${validate}" -eq "1" ]; then + if [ "${XMLLINT}${XMLSTARLET}" = "::" ] ; then + echo "Either xmllint or xmlstarlet is required for wiki post-processing or validation." + exit 1 + fi +fi + +if [ "${command}" = "print_dependencies" ] ; then + for subdir in ${mod_subdirs} ; do + subpath="${source_tree}/${subdir}" + # We WANT word splitting in the following line. + # shellcheck disable=SC2046 + ${GREP} -l -E '(language="en_US"|appdocsxml.dtd)' $(${FIND} "${subpath}" -name '*.c' -or -name '*.cc' -or -name '*.xml') || : + done + exit +fi + +if [ "${command}" != "create_xml" ] ; then + echo "Command '${command}' is invalid." + usage + exit 1 +fi + +if [ "${output_file}" = "" ] ; then + echo "output-file is required for command '${command}'." + usage + exit 1; +fi + +output_dir=$(${DIRNAME} "${output_file}") +if [ ! -d "${output_dir}" ] ; then + echo "output destination directory '${output_dir}' doesn't exist." + exit 1 +fi + +if [ "${for_wiki}" -eq "1" ] && [ "${core_output_file}" = "" ] ; then + echo "core-output-file is required for command '${command}' and 'for-wiki'." + usage + exit 1; +fi + +core_output_dir=$(${DIRNAME} "${core_output_file}") +if [ ! -d "${core_output_dir}" ] ; then + echo "core destination directory '${core_output_dir}' doesn't exist." + exit 1 +fi + +${CAT} > "${output_file}" <<-EOF + + + + +EOF + +printf "Building Documentation For: " + +for subdir in ${mod_subdirs} ; do + printf "%s " "${subdir}" + subdir_path="${source_tree}/${subdir}" + for i in $(${FIND} "${subdir_path}" -name '*.c' -or -name '*.cc'); do + if [ "${with_moduleinfo}" -eq "1" ] ; then + MODULEINFO=$(${AWK} -f "${source_tree}/build_tools/get_moduleinfo" "${i}") + if [ "${MODULEINFO}" != "" ] ; then + ${CAT} >> "${output_file}" <<-EOF + + ${MODULEINFO} + + EOF + fi + fi + if [ "${for_wiki}" -eq "1" ] ; then + ${PYTHON} build_tools/get_documentation.py < "${i}" >> "${output_file}" + else + ${AWK} -f "${source_tree}/build_tools/get_documentation" "${i}" >> "${output_file}" + fi + done + for i in $(${FIND} "${subdir_path}" -name '*.xml') ; do + ${GREP} -q "appdocsxml.dtd" "${i}" || continue + if [ "${validate}" -eq "1" ] ;then + if [ "${XMLLINT}" != ":" ] ; then + ${XMLLINT} --dtdvalid "${source_tree}/doc/appdocsxml.dtd" --path "${source_tree}/doc" --noout "${i}" || { echo "" ; exit 1 ; } + else + ${XMLSTARLET} val -d "${source_tree}/doc/appdocsxml.dtd" "${i}" || { echo "" ; exit 1 ; } + fi + fi + ${SED} -r "/^\s*(<[?]xml|<.DOCTYPE|<.?docs)/d" "${i}" >> "${output_file}" + done +done +echo "" >> "${output_file}" +echo "" + +if [ "${for_wiki}" -eq "1" ] ; then + ${PYTHON} build_tools/post_process_documentation.py -i "${output_file}" -o "${core_output_file}" +fi + +if [ "${validate}" -eq "1" ] ;then + if [ "${XMLLINT}" != ":" ] ; then + ${XMLLINT} --dtdvalid "${source_tree}/doc/appdocsxml.dtd" --path "${source_tree}/doc" --noout "${output_file}" || exit 1 + else + ${XMLSTARLET} val -d "${source_tree}/doc/appdocsxml.dtd" "${output_file}" || exit 1 + fi +fi + diff --git a/makeopts.in b/makeopts.in index 77bc974d84..54eaa04a63 100644 --- a/makeopts.in +++ b/makeopts.in @@ -46,6 +46,8 @@ SED=@SED@ NM=@NM@ CAT=@CAT@ CUT=@CUT@ +REALPATH=@REALPATH@ +DIRNAME=@DIRNAME@ BUILD_PLATFORM=@BUILD_PLATFORM@ BUILD_CPU=@BUILD_CPU@ diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 3d1f68c029..78226450e4 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -56,3248 +56,6 @@ core ***/ -/*** DOCUMENTATION - - SIP Resource using PJProject - - - Endpoint - - The Endpoint is the primary configuration object. - It contains the core SIP related options only, endpoints are NOT - dialable entries of their own. Communication with another SIP device is - accomplished via Addresses of Record (AoRs) which have one or more - contacts associated with them. Endpoints NOT configured to - use a transport will default to first transport found - in pjsip.conf that matches its type. - - Example: An Endpoint has been configured with no transport. - When it comes time to call an AoR, PJSIP will find the - first transport that matches the type. A SIP URI of sip:5000@[11::33] - will use the first IPv6 transport and try to send the request. - - If the anonymous endpoint identifier is in use an endpoint with the name - "anonymous@domain" will be searched for as a last resort. If this is not found - it will fall back to searching for "anonymous". If neither endpoints are found - the anonymous endpoint identifier will not return an endpoint and anonymous - calling will not be possible. - - - - Allow support for RFC3262 provisional ACK tags - - - - - - - - - - Condense MWI notifications into a single NOTIFY. - When enabled, aggregate_mwi condenses message - waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled, - individual NOTIFYs are sent for each mailbox. - - - Media Codec(s) to allow - - - Codec negotiation prefs for incoming offers. - - - This is a string that describes how the codecs - specified on an incoming SDP offer (pending) are reconciled with the codecs specified - on an endpoint (configured) before being sent to the Asterisk core. - The string actually specifies 4 name:value pair parameters - separated by commas. Whitespace is ignored and they may be specified in any order. - Note that this option is reserved for future functionality. - - - - Parameters: - - - - - - - The codec list from the caller. (default) - The codec list from the endpoint. - - - - - - - Only common codecs with the preferred codecs first. (default) - Use only the preferred codecs. - Use only the non-preferred codecs. - - - - - - - After the operation, keep all codecs. (default) - After the operation, keep only the first codec. - - - - - - - Allow transcoding. (default) - Prevent transcoding. - - - - - - - codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: all, transcode: allow - - - Prefer the codecs coming from the caller. Use only the ones that are common. - keeping the order of the preferred list. Keep all codecs in the result. Allow transcoding. - - - - - Codec negotiation prefs for outgoing offers. - - - This is a string that describes how the codecs specified in the topology that - comes from the Asterisk core (pending) are reconciled with the codecs specified on an - endpoint (configured) when sending an SDP offer. - The string actually specifies 4 name:value pair parameters - separated by commas. Whitespace is ignored and they may be specified in any order. - Note that this option is reserved for future functionality. - - - - Parameters: - - - - - - - The codec list from the core. (default) - The codec list from the endpoint. - - - - - - - Merge the lists with the preferred codecs first. (default) - Only common codecs with the preferred codecs first. (default) - Use only the preferred codecs. - Use only the non-preferred codecs. - - - - - - - After the operation, keep all codecs. (default) - After the operation, keep only the first codec. - - - - - - - Allow transcoding. (default) - Prevent transcoding. - - - - - - - codec_prefs_outgoing_offer = prefer: configured, operation: union, keep: first, transcode: prevent - - - Prefer the codecs coming from the endpoint. Merge them with the codecs from the core - keeping the order of the preferred list. Keep only the first one. No transcoding allowed. - - - - - Codec negotiation prefs for incoming answers. - - - This is a string that describes how the codecs specified in an incoming SDP answer - (pending) are reconciled with the codecs specified on an endpoint (configured) - when receiving an SDP answer. - The string actually specifies 4 name:value pair parameters - separated by commas. Whitespace is ignored and they may be specified in any order. - Note that this option is reserved for future functionality. - - - - Parameters: - - - - - - - The codec list in the received SDP answer. (default) - The codec list from the endpoint. - - - - - - - Merge the lists with the preferred codecs first. - Only common codecs with the preferred codecs first. (default) - Use only the preferred codecs. - Use only the non-preferred codecs. - - - - - - - After the operation, keep all codecs. (default) - After the operation, keep only the first codec. - - - - - The transcode parameter is ignored when processing answers. - - - - - - - codec_prefs_incoming_answer = keep: first - - - Use the defaults but keep oinly the first codec. - - - - - Codec negotiation prefs for outgoing answers. - - - This is a string that describes how the codecs that come from the core (pending) - are reconciled with the codecs specified on an endpoint (configured) - when sending an SDP answer. - The string actually specifies 4 name:value pair parameters - separated by commas. Whitespace is ignored and they may be specified in any order. - Note that this option is reserved for future functionality. - - - - Parameters: - - - - - - - The codec list that came from the core. (default) - The codec list from the endpoint. - - - - - - - Merge the lists with the preferred codecs first. - Only common codecs with the preferred codecs first. (default) - Use only the preferred codecs. - Use only the non-preferred codecs. - - - - - - - After the operation, keep all codecs. (default) - After the operation, keep only the first codec. - - - - - The transcode parameter is ignored when processing answers. - - - - - - - codec_prefs_incoming_answer = keep: first - - - Use the defaults but keep oinly the first codec. - - - - - Enable RFC3578 overlap dialing support. - - - AoR(s) to be used with the endpoint - - List of comma separated AoRs that the endpoint should be associated with. - - - - Authentication Object(s) associated with the endpoint - - This is a comma-delimited list of auth sections defined - in pjsip.conf to be used to verify inbound connection attempts. - - Endpoints without an authentication object - configured will allow connections without verification. - - Using the same auth section for inbound and outbound - authentication is not recommended. There is a difference in - meaning for an empty realm setting between inbound and outbound - authentication uses. See the auth realm description for details. - - - - - CallerID information for the endpoint - - Must be in the format Name <Number>, - or only <Number>. - - - - Default privacy level - - - - - - - - - - - - - - - - Internal id_tag for the endpoint - - - Dialplan context for inbound sessions - - - Mitigation of direct media (re)INVITE glare - - - This setting attempts to avoid creating INVITE glare scenarios - by disabling direct media reINVITEs in one direction thereby allowing - designated servers (according to this option) to initiate direct - media reINVITEs without contention and significantly reducing call - setup time. - - - A more detailed description of how this option functions can be found on - the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance - - - - - - - - - - Direct Media method type - - Method for setting up Direct Media between endpoints. - - - - Alias for the invite value. - - - - - - - Accept Connected Line updates from this endpoint - - - Send Connected Line updates to this endpoint - - - Connected line method type - - Method used when updating connected line information. - - - When set to invite, check the remote's Allow header and - if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP - renegotiation. If UPDATE is not Allowed, send INVITE. - - - Alias for the invite value. - - - If set to update, send UPDATE regardless of what the remote - Allows. - - - - - - Determines whether media may flow directly between endpoints. - - - Disable direct media session refreshes when NAT obstructs the media session - - - Media Codec(s) to disallow - - - DTMF mode - - This setting allows to choose the DTMF mode for endpoint communication. - - - DTMF is sent out of band of the main audio stream. This - supercedes the older RFC-2833 used within - the older chan_sip. - - - DTMF is sent as part of audio stream. - - - DTMF is sent as SIP INFO packets. - - - DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not. - - - DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not. - - - - - - IP address used in SDP for media handling - - At the time of SDP creation, the IP address defined here will be used as - the media address for individual streams in the SDP. - - - Be aware that the external_media_address option, set in Transport - configuration, can also affect the final media address used in the SDP. - - - - - Bind the RTP instance to the media_address - - If media_address is specified, this option causes the RTP instance to be bound to the - specified ip address which causes the packets to be sent from that address. - - - - - Force use of return port - - - Enable the ICE mechanism to help traverse NAT - - - Way(s) for the endpoint to be identified - - Endpoints and AORs can be identified in multiple ways. This - option is a comma separated list of methods the endpoint can be - identified. - - - This option controls both how an endpoint is matched for incoming - traffic and also how an AOR is determined if a registration - occurs. You must list at least one method that also matches for - AORs or the registration will fail. - - - - Matches the endpoint or AOR ID based on the username - and domain in the From header (or To header for AORs). If - an exact match on both username and domain/realm fails, the - match is retried with just the username. - - - - Matches the endpoint or AOR ID based on the username - and realm in the Authentication header. If an exact match - on both username and domain/realm fails, the match is - retried with just the username. - - This method of identification has some security - considerations because an Authentication header is not - present on the first message of a dialog when digest - authentication is used. The client can't generate it until - the server sends the challenge in a 401 response. Since - Asterisk normally sends a security event when an incoming - request can't be matched to an endpoint, using this method - requires that the security event be deferred until a request - is received with the Authentication header and only - generated if the username doesn't result in a match. This - may result in a delay before an attack is recognized. You - can control how many unmatched requests are received from - a single ip address before a security event is generated - using the unidentified_request - parameters in the "global" configuration object. - - - - Matches the endpoint based on the source IP address. - - This method of identification is not configured here - but simply allowed by this configuration option. See the - documentation for the identify - configuration section for more details on this method of - endpoint identification. - - - - Matches the endpoint based on a configured SIP header - value. - - This method of identification is not configured here - but simply allowed by this configuration option. See the - documentation for the identify - configuration section for more details on this method of - endpoint identification. - - - - - - - How redirects received from an endpoint are handled - - When a redirect is received from an endpoint there are multiple ways it can be handled. - If this option is set to user the user portion of the redirect target - is treated as an extension within the dialplan and dialed using a Local channel. If this option - is set to uri_core the target URI is returned to the dialing application - which dials it using the PJSIP channel driver and endpoint originally used. If this option is - set to uri_pjsip the redirect occurs within chan_pjsip itself and is not exposed - to the core at all. The uri_pjsip option has the benefit of being more efficient - and also supporting multiple potential redirect targets. The con is that since redirection occurs - within chan_pjsip redirecting information is not forwarded and redirection can not be - prevented. - - - - - - - - - - NOTIFY the endpoint when state changes for any of the specified mailboxes - - Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state - changes happen for any of the specified mailboxes. More than one mailbox can be - specified with a comma-delimited string. app_voicemail mailboxes must be specified - as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by - external sources, such as through the res_mwi_external module, you must specify - strings supported by the external system. - - For endpoints that SUBSCRIBE for MWI, use the mailboxes option in your AOR - configuration. - - - - An MWI subscribe will replace sending unsolicited NOTIFYs - - - The voicemail extension to send in the NOTIFY Message-Account header - - - Default Music On Hold class - - - Authentication object(s) used for outbound requests - - This is a comma-delimited list of auth - sections defined in pjsip.conf used to respond - to outbound connection authentication challenges. - - Using the same auth section for inbound and outbound - authentication is not recommended. There is a difference in - meaning for an empty realm setting between inbound and outbound - authentication uses. See the auth realm description for details. - - - - - Full SIP URI of the outbound proxy used to send requests - - - Allow Contact header to be rewritten with the source IP address-port - - On inbound SIP messages from this endpoint, the Contact header or an - appropriate Record-Route header will be changed to have the source IP - address and port. This option does not affect outbound messages sent to - this endpoint. This option helps servers communicate with endpoints - that are behind NATs. This option also helps reuse reliable transport - connections such as TCP and TLS. - - - - Allow use of IPv6 for RTP traffic - - - Enforce that RTP must be symmetric - - - Send the Diversion header, conveying the diversion - information to the called user agent - - - Send the History-Info header, conveying the diversion - information to the called and calling user agents - - - Send the P-Asserted-Identity header - - - Send the Remote-Party-ID header - - - Immediately send connected line updates on unanswered incoming calls. - - When enabled, immediately send 180 Ringing - or 183 Progress response messages to the - caller if the connected line information is updated before - the call is answered. This can send a 180 Ringing - response before the call has even reached the far end. The - caller can start hearing ringback before the far end even gets - the call. Many phones tend to grab the first connected line - information and refuse to update the display if it changes. The - first information is not likely to be correct if the call - goes to an endpoint not under the control of this Asterisk - box. - When disabled, a connected line update must wait for - another reason to send a message with the connected line - information to the caller before the call is answered. You can - trigger the sending of the information by using an appropriate - dialplan application such as Ringing. - - - - Minimum session timers expiration period - - Minimum session timer expiration period. Time in seconds. - - - - Session timers for SIP packets - - - - - - - Alias of always - - - - - Maximum session timer expiration period - - Maximum session timer expiration period. Time in seconds. - - - - Explicit transport configuration to use - - This will force the endpoint to use the - specified transport configuration to send SIP messages. You need - to already know what kind of transport (UDP/TCP/IPv4/etc) the - endpoint device will use. - - Not specifying a transport will select the first - configured transport in pjsip.conf which is - compatible with the URI we are trying to contact. - - Transport configuration is not affected by reloads. In order to - change transports, a full Asterisk restart is required - - - - Accept identification information received from this endpoint - This option determines whether Asterisk will accept - identification from the endpoint from headers such as P-Asserted-Identity - or Remote-Party-ID header. This option applies both to calls originating from the - endpoint and calls originating from Asterisk. If no, the - configured Caller-ID from pjsip.conf will always be used as the identity for - the endpoint. - - - Send private identification details to the endpoint. - This option determines whether res_pjsip will send private - identification information to the endpoint. If no, - private Caller-ID information will not be forwarded to the endpoint. - "Private" in this case refers to any method of restricting identification. - Example: setting callerid_privacy to any - prohib variation. - Example: If trust_id_inbound is set to - yes, the presence of a Privacy: id - header in a SIP request or response would indicate the identification - provided in the request is private. - - - Must be of type 'endpoint'. - - - Use Endpoint's requested packetization interval - - - Determines whether res_pjsip will use and enforce usage of AVPF for this - endpoint. - - If set to yes, res_pjsip will use the AVPF or SAVPF RTP - profile for all media offers on outbound calls and media updates and will - decline media offers not using the AVPF or SAVPF profile. - - If set to no, res_pjsip will use the AVP or SAVP RTP - profile for all media offers on outbound calls and media updates, and will - decline media offers not using the AVP or SAVP profile. - - - - Determines whether res_pjsip will use and enforce usage of AVP, - regardless of the RTP profile in use for this endpoint. - - If set to yes, res_pjsip will use the AVP, AVPF, SAVP, or - SAVPF RTP profile for all media offers on outbound calls and media updates including - those for DTLS-SRTP streams. - - If set to no, res_pjsip will use the respective RTP profile - depending on configuration. - - - - Determines whether res_pjsip will use the media transport received in the - offer SDP in the corresponding answer SDP. - - If set to yes, res_pjsip will use the received media transport. - - If set to no, res_pjsip will use the respective RTP profile - depending on configuration. - - - - Determines whether res_pjsip will use and enforce usage of media encryption - for this endpoint. - - - - res_pjsip will offer no encryption and allow no encryption to be setup. - - - res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP - transport should be used in conjunction with this option to prevent - exposure of media encryption keys. - - - res_pjsip will offer DTLS-SRTP setup. - - - - - - Determines whether encryption should be used if possible but does not terminate the - session if not achieved. - - This option only applies if media_encryption is - set to sdes or dtls. - - - - Force g.726 to use AAL2 packing order when negotiating g.726 audio - - When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 - packing order instead of what is recommended by RFC3551. Since this essentially - replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be - specified in the endpoint's allowed codec list. - - - - Determines whether chan_pjsip will indicate ringing using inband - progress. - - If set to yes, chan_pjsip will send a 183 Session Progress - when told to indicate ringing and will immediately start sending ringing - as audio. - - If set to no, chan_pjsip will send a 180 Ringing when told - 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 - - - - No error correction should be used. - - - Forward error correction should be used. - - - Redundancy error correction should be used. - - - - - - 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. - - - - How long into a call before fax_detect is disabled for the call - - The option determines how many seconds into a call before the - fax_detect option is disabled for the call. Setting the value - to zero disables the timeout. - - - - 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. - - - - Bind the UDPTL instance to the media_adress - - If media_address is specified, this option causes the UDPTL instance to be bound to - the specified ip address which causes the packets to be sent from that address. - - - - Set which country's indications to use for channels created for this endpoint. - - - Set the default language to use for channels created for this endpoint. - - - Determines whether one-touch recording is allowed for this endpoint. - - record_on_feature - record_off_feature - - - - 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 - feature will be enabled for the channel. The feature designated here can be any built-in - or dynamic feature defined in features.conf. - This setting has no effect if the endpoint's one_touch_recording option is disabled - - - one_touch_recording - record_off_feature - - - - 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 - feature will be enabled for the channel. The feature designated here can be any built-in - or dynamic feature defined in features.conf. - This setting has no effect if the endpoint's one_touch_recording option is disabled - - - one_touch_recording - record_on_feature - - - - Name of the RTP engine to use for channels created for this endpoint - - - Determines whether SIP REFER transfers are allowed for this endpoint - - - Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number - - - Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side - - - String placed as the username portion of an SDP origin (o=) line. - - - String used for the SDP session (s=) line. - - - DSCP TOS bits for audio streams - - See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings - - - - DSCP TOS bits for video streams - - See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings - - - - Priority for audio streams - - See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings - - - - Priority for video streams - - 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. - - It can be one of the following values: - - - meaning no verification is done. - - - meaning to verify the remote fingerprint. - - - meaning to verify the remote certificate. - - - meaning to verify both the remote fingerprint and certificate. - - - - - - Interval at which to renegotiate the TLS session and rekey the SRTP session - - This option only applies if media_encryption is - set to dtls. - - If this is not set or the value provided is 0 rekeying will be disabled. - - - - Whether or not to automatically generate an ephemeral X.509 certificate - - - If enabled, Asterisk will generate an X.509 certificate for each DTLS session. - This option only applies if media_encryption is set - to dtls. This option will be automatically enabled if - webrtc is enabled and dtls_cert_file is - not specified. - - - - - 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 - set to dtls. - - Many options for acceptable ciphers. See link for more: - 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. - - - This option only applies if media_encryption is - set to dtls. - - - - res_pjsip will make a connection to the peer. - - - res_pjsip will accept connections from the peer. - - - res_pjsip will offer and accept connections from the peer. - - - - - - Type of hash to use for the DTLS fingerprint in the SDP. - - - This option only applies if media_encryption is - set to dtls. - - - - - - - - - Determines whether 32 byte tags should be used instead of 80 byte tags. - - This option only applies if media_encryption is - set to sdes or dtls. - - - - Variable set on a channel involving the endpoint. - - When a new channel is created using the endpoint set the specified - variable(s) on that channel. For multiple channel variables specify - multiple 'set_var'(s). - - - - Context to route incoming MESSAGE requests to. - - If specified, incoming MESSAGE requests will be routed to the indicated - dialplan context. If no message_context is - specified, then the context setting is used. - - - - An accountcode to set automatically on any channels created for this endpoint. - - If specified, any channel created for this endpoint will automatically - have this accountcode set on it. - - - - Respond to a SIP invite with the single most preferred codec (DEPRECATED) - Respond to a SIP invite with the single most preferred codec - rather than advertising all joint codec capabilities. This limits the other side's codec - choice to exactly what we prefer. - This option has been deprecated in favor of - incoming_call_offer_pref. Setting both options is unsupported. - - - - incoming_call_offer_pref - - - - Preferences for selecting codecs for an incoming call. - - Based on this setting, a joint list of preferred codecs between those - received in an incoming SDP offer (remote), and those specified in the - endpoint's "allow" parameter (local) es created and is passed to the Asterisk - core. - This list will consist of only those codecs found in both lists. - - - Include all codecs in the local list that are also in the remote list - preserving the local order. (default). - - - Include only the first codec in the local list that is also in the remote list. - - - Include all codecs in the remote list that are also in the local list - preserving the remote order. - - - Include only the first codec in the remote list that is also in the local list. - - - - - - Preferences for selecting codecs for an outgoing call. - - Based on this setting, a joint list of preferred codecs between - those received from the Asterisk core (remote), and those specified in - the endpoint's "allow" parameter (local) is created and is used to create - the outgoing SDP offer. - - - Include all codecs in the local list that are also in the remote list - preserving the local order. - - - Include all codecs in the local list preserving the local order. - - - Include only the first codec in the local list. - - - Include all codecs in the remote list that are also in the local list - preserving the remote order. - - - Include all codecs in the local list preserving the remote order. (default) - - - Include only the first codec in the remote list that is also in the local list. - - - - - - Number of seconds between RTP comfort noise keepalive packets. - - At the specified interval, Asterisk will send an RTP comfort noise frame. This may - be useful for situations where Asterisk is behind a NAT or firewall and must keep - a hole open in order to allow for media to arrive at Asterisk. - - - - Maximum number of seconds without receiving RTP (while off hold) before terminating call. - - This option configures the number of seconds without RTP (while off hold) before - considering a channel as dead. When the number of seconds is reached the underlying - channel is hung up. By default this option is set to 0, which means do not check. - - - - Maximum number of seconds without receiving RTP (while on hold) before terminating call. - - This option configures the number of seconds without RTP (while on hold) before - considering a channel as dead. When the number of seconds is reached the underlying - channel is hung up. By default this option is set to 0, which means do not check. - - - - List of IP ACL section names in acl.conf - - This matches sections configured in acl.conf. The value is - defined as a list of comma-delimited section names. - - - - List of IP addresses to deny access from - - The value is a comma-delimited list of IP addresses. IP addresses may - have a subnet mask appended. The subnet mask may be written in either - CIDR or dotted-decimal notation. Separate the IP address and subnet - mask with a slash ('/') - - - - List of IP addresses to permit access from - - The value is a comma-delimited list of IP addresses. IP addresses may - have a subnet mask appended. The subnet mask may be written in either - CIDR or dotted-decimal notation. Separate the IP address and subnet - mask with a slash ('/') - - - - List of Contact ACL section names in acl.conf - - This matches sections configured in acl.conf. The value is - defined as a list of comma-delimited section names. - - - - List of Contact header addresses to deny - - The value is a comma-delimited list of IP addresses. IP addresses may - have a subnet mask appended. The subnet mask may be written in either - CIDR or dotted-decimal notation. Separate the IP address and subnet - mask with a slash ('/') - - - - List of Contact header addresses to permit - - The value is a comma-delimited list of IP addresses. IP addresses may - have a subnet mask appended. The subnet mask may be written in either - CIDR or dotted-decimal notation. Separate the IP address and subnet - mask with a slash ('/') - - - - Context for incoming MESSAGE requests. - - If specified, incoming SUBSCRIBE requests will be searched for the matching - extension in the indicated context. - If no subscribe_context is specified, - then the context setting is used. - - - - Force the user on the outgoing Contact header to this value. - - On outbound requests, force the user portion of the Contact header to this value. - - - - Allow the sending and receiving RTP codec to differ - - When set to "yes" the codec in use for sending will be allowed to differ from - that of the received one. PJSIP will not automatically switch the sending one - to the receiving one. - - - - Enable RFC 5761 RTCP multiplexing on the RTP port - - With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" - attribute on all media streams. This will result in RTP and RTCP being sent and received - on the same port. This shifts the demultiplexing logic to the application rather than - the transport layer. This option is useful when interoperating with WebRTC endpoints - since they mandate this option's use. - - - - Whether to notifies all the progress details on blind transfer - - Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" - after REFER has been accepted. If set to no then asterisk - will not send the progress details, but immediately will send "200 OK". - - - - Whether to notifies dialog-info 'early' on InUse&Ringing state - - Control whether dialog-info subscriptions get 'early' state - on Ringing when already INUSE. - - - - The maximum number of allowed audio streams for the endpoint - - This option enforces a limit on the maximum simultaneous negotiated audio - streams allowed for the endpoint. - - - - The maximum number of allowed video streams for the endpoint - - This option enforces a limit on the maximum simultaneous negotiated video - streams allowed for the endpoint. - - - - Enable RTP bundling - - With this option enabled, Asterisk will attempt to negotiate the use of bundle. - If negotiated this will result in multiple RTP streams being carried over the same - underlying transport. Note that enabling bundle will also enable the rtcp_mux option. - - - - Defaults and enables some options that are relevant to WebRTC - - When set to "yes" this also enables the following values that are needed in - order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and - use_received_transport. The following configuration settings also get defaulted - as follows: - media_encryption=dtls - dtls_auto_generate_cert=yes (if dtls_cert_file is not set) - dtls_verify=fingerprint - dtls_setup=actpass - - - - Mailbox name to use when incoming MWI NOTIFYs are received - - If an MWI NOTIFY is received from this endpoint, - this mailbox will be used when notifying other modules of MWI status - changes. If not set, incoming MWI NOTIFYs are ignored. - - - - Follow SDP forked media when To tag is different - - On outgoing calls, if the UAS responds with different SDP attributes - on subsequent 18X or 2XX responses (such as a port update) AND the - To tag on the subsequent response is different than that on the previous - one, follow it. This usually happens when the INVITE is forked to multiple - UASs and more than one sends an SDP answer. - - - This option must also be enabled in the system - section for it to take effect here. - - - - - Accept multiple SDP answers on non-100rel responses - - On outgoing calls, if the UAS responds with different SDP attributes - on non-100rel 18X or 2XX responses (such as a port update) AND the - To tag on the subsequent response is the same as that on the previous one, - process the updated SDP. This can happen when the UAS needs to change ports - for some reason such as using a separate port for custom ringback. - - - This option must also be enabled in the system - section for it to take effect here. - - - - - Suppress Q.850 Reason headers for this endpoint - - Some devices can't accept multiple Reason headers and get confused - when both 'SIP' and 'Q.850' Reason headers are received. This - option allows the 'Q.850' Reason header to be suppressed. - - - - Do not forward 183 when it doesn't contain SDP - - Certain SS7 internetworking scenarios can result in a 183 - to be generated for reasons other than early media. Forwarding - this 183 can cause loss of ringback tone. This flag emulates - the behavior of chan_sip and prevents these 183 responses from - being forwarded. - - - - Enable STIR/SHAKEN support on this endpoint - - Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs, - the Identity header will be checked for validity. On outgoing - INVITEs, an Identity header will be added. - - - - Skip authentication when receiving OPTIONS requests - - RFC 3261 says that the response to an OPTIONS request MUST be the - same had the request been an INVITE. Some UAs use OPTIONS requests - like a 'ping' and the expectation is that they will return a - 200 OK. - Enabling allow_unauthenticated_options - will skip authentication of OPTIONS requests for the given - endpoint. - There are security implications to enabling this setting as - it can allow information disclosure to occur - specifically, if - enabled, an external party could enumerate and find the endpoint - name by sending OPTIONS requests and examining the - responses. - - - - - Authentication type - - Authentication objects hold the authentication information for use - by other objects such as endpoints or registrations. - This also allows for multiple objects to use a single auth object. See - the auth_type config option for password style choices. - - - Authentication type - - This option specifies which of the password style config options should be read - when trying to authenticate an endpoint inbound request. If set to userpass - then we'll read from the 'password' option. For md5 we'll read - from 'md5_cred'. If set to google_oauth then we'll read from the - refresh_token/oauth_clientid/oauth_secret fields. The following values are valid: - - - - - - - - - - - This setting only describes whether the password is in - plain text or has been pre-hashed with MD5. It doesn't describe - the acceptable digest algorithms we'll accept in a received - challenge. - - - - - - Lifetime of a nonce associated with this authentication config. - - - MD5 Hash used for authentication. - - Only used when auth_type is md5. - As an alternative to specifying a plain text password, - you can hash the username, realm and password - together one time and place the hash value here. - The input to the hash function must be in the - following format: - - - - - <username>:<realm>:<password> - - - - - For incoming authentication (asterisk is the server), - the realm must match either the realm set in this object - or the default_realm set in in the - global object. - - - - - For outgoing authentication (asterisk is the UAC), - the realm must match what the server will be sending - in their WWW-Authenticate header. It can't be blank - unless you expect the server to be sending a blank - realm in the header. You can't use pre-hashed - passwords with a wildcard auth object. - You can generate the hash with the following shell - command: - - - - - $ echo -n "myname:myrealm:mypassword" | md5sum - - - - - Note the '-n'. You don't want a newline to be part - of the hash. - - - - Plain text password used for authentication. - Only used when auth_type is userpass. - - - OAuth 2.0 refresh token - - - OAuth 2.0 application's client id - - - OAuth 2.0 application's secret - - - SIP realm for endpoint - - For incoming authentication (asterisk is the UAS), - this is the realm to be sent on WWW-Authenticate - headers. If not specified, the global - object's default_realm will be used. - - - - - For outgoing authentication (asterisk is the UAC), this - must either be the realm the server is expected to send, - or left blank or contain a single '*' to automatically - use the realm sent by the server. If you have multiple - auth objects for an endpoint, the realm is also used to - match the auth object to the realm the server sent. - - - - - - Using the same auth section for inbound and outbound - authentication is not recommended. There is a difference in - meaning for an empty realm setting between inbound and outbound - authentication uses. - - - - - - - If more than one auth object with the same realm or - more than one wildcard auth object associated to - an endpoint, we can only use the first one of - each defined on the endpoint. - - - - - - Must be 'auth' - - - Username to use for account - - - - Domain Alias - - Signifies that a domain is an alias. If the domain on a session is - not found to match an AoR then this object is used to see if we have - an alias for the AoR to which the endpoint is binding. This objects - name as defined in configuration should be the domain alias and a - config option is provided to specify the domain to be aliased. - - - Must be of type 'domain_alias'. - - - Domain to be aliased - - - - SIP Transport - - Transports - - There are different transports and protocol derivatives - supported by res_pjsip. They are in order of - preference: UDP, TCP, and WebSocket (WS). - Changes to transport configuration in pjsip.conf will only be - effected on a complete restart of Asterisk. A module reload - will not suffice. - - - Number of simultaneous Asynchronous Operations - - - IP Address and optional port to bind to for this transport - - - File containing a list of certificates to read (TLS ONLY, not WSS) - - - Path to directory containing a list of certificates to read (TLS ONLY, not WSS) - - - Certificate file for endpoint (TLS ONLY, not WSS) - - A path to a .crt or .pem file can be provided. However, only - the certificate is read from the file, not the private key. - The priv_key_file option must supply a - matching key file. - - - - Preferred cryptography cipher names (TLS ONLY, not WSS) - - Comma separated list of cipher names or numeric equivalents. - Numeric equivalents can be either decimal or hexadecimal (0xX). - - There are many cipher names. Use the CLI command - pjsip list ciphers to see a list of cipher - names available for your installation. See link for more: - http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES - - - - - Domain the transport comes from - - - External IP address to use in RTP handling - - When a request or response is sent out, if the destination of the - message is outside the IP network defined in the option localnet, - and the media address in the SDP is within the localnet network, then the - media address in the SDP will be rewritten to the value defined for - external_media_address. - - - - External address for SIP signalling - - - External port for SIP signalling - - - Method of SSL transport (TLS ONLY, not WSS) - - - - The default as defined by PJSIP. This is currently TLSv1, but may change with future releases. - - - This option is equivalent to setting 'default' - - - - - - - - - - - - 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 ('/'). - - - Password required for transport - - - Private key file (TLS ONLY, not WSS) - - - Protocol to use for SIP traffic - - - - - - - - - - - - - Require client certificate (TLS ONLY, not WSS) - - - Must be of type 'transport'. - - - Require verification of client certificate (TLS ONLY, not WSS) - - - Require verification of server certificate (TLS ONLY, not WSS) - - - Enable TOS for the signalling sent over this transport - - See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service - for more information on this parameter. - This option does not apply to the ws - or the wss protocols. - - - - Enable COS for the signalling sent over this transport - - See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service - for more information on this parameter. - This option does not apply to the ws - or the wss protocols. - - - - The timeout (in milliseconds) to set on WebSocket connections. - - If a websocket connection accepts input slowly, the timeout - for writes to it can be increased to keep it from being disconnected. - Value is in milliseconds. - - - - Allow this transport to be reloaded. - - Allow this transport to be reloaded when res_pjsip is reloaded. - This option defaults to "no" because reloading a transport may disrupt - in-progress calls. - - - - Use the same transport for outgoing requests as incoming ones. - - When a request from a dynamic contact - comes in on a transport with this option set to 'yes', - the transport name will be saved and used for subsequent - outgoing requests like OPTIONS, NOTIFY and INVITE. It's - saved as a contact uri parameter named 'x-ast-txp' and will - display with the contact uri in CLI, AMI, and ARI output. - On the outgoing request, if a transport wasn't explicitly - set on the endpoint AND the request URI is not a hostname, - the saved transport will be used and the 'x-ast-txp' - parameter stripped from the outgoing packet. - - - - - - A way of creating an aliased name to a SIP URI - - Contacts are a way to hide SIP URIs from the dialplan directly. - They are also used to make a group of contactable parties when - in use with AoR lists. - - - Must be of type 'contact'. - - - SIP URI to contact peer - - - Time to keep alive a contact - - Time to keep alive a contact. String style specification. - - - - Interval at which to qualify a contact - - Interval between attempts to qualify the contact for reachability. - If 0 never qualify. Time in seconds. - - - - Timeout for qualify - - If the contact doesn't respond to the OPTIONS request before the timeout, - the contact is marked unavailable. - If 0 no timeout. Time in fractional seconds. - - - - Authenticates a qualify challenge response if needed - - If true and a qualify request receives a challenge response then - authentication is attempted before declaring the contact available. - - This option does nothing as we will always complete - the challenge response authentication if the qualify request is - challenged. - - - - - Outbound proxy used when sending OPTIONS request - - If set the provided URI will be used as the outbound proxy when an - OPTIONS request is sent to a contact for qualify purposes. - - - - Stored Path vector for use in Route headers on outgoing requests. - - - User-Agent header from registration. - - The User-Agent is automatically stored based on data present in incoming SIP - REGISTER requests and is not intended to be configured manually. - - - - Endpoint name - - The name of the endpoint this contact belongs to - - - - Asterisk Server name - - Asterisk Server name on which SIP endpoint registered. - - - - IP-address of the last Via header from registration. - - The last Via header should contain the address of UA which sent the request. - The IP-address of the last Via header is automatically stored based on data present - in incoming SIP REGISTER requests and is not intended to be configured manually. - - - - IP-port of the last Via header from registration. - - The IP-port of the last Via header is automatically stored based on data present - in incoming SIP REGISTER requests and is not intended to be configured manually. - - - - Call-ID header from registration. - - The Call-ID header is automatically stored based on data present - in incoming SIP REGISTER requests and is not intended to be configured manually. - - - - A contact that cannot survive a restart/boot. - - The option is set if the incoming SIP REGISTER contact is rewritten - on a reliable transport and is not intended to be configured manually. - - - - - The configuration for a location of an endpoint - - An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no - AoRs are specified, an endpoint will not be reachable by Asterisk. - Beyond that, an AoR has other uses within Asterisk, such as inbound - registration. - - An AoR is a way to allow dialing a group - of Contacts that all use the same - endpoint for calls. - - This can be used as another way of grouping a list of contacts to dial - rather than specifying them each directly when dialing via the dialplan. - This must be used in conjunction with the PJSIP_DIAL_CONTACTS. - - Registrations: For Asterisk to match an inbound registration to an endpoint, - the AoR object name must match the user portion of the SIP URI in the "To:" - header of the inbound SIP registration. That will usually be equivalent - to the "user name" set in your hard or soft phones configuration. - - - Permanent contacts assigned to AoR - - Contacts specified will be called whenever referenced - by chan_pjsip. - - Use a separate "contact=" entry for each contact required. Contacts - are specified using a SIP URI. - - - - Default expiration time in seconds for contacts that are dynamically bound to an AoR. - - - Allow subscriptions for the specified mailbox(es) - This option applies when an external entity subscribes to an AoR - for Message Waiting Indications. The mailboxes specified will be subscribed to. - More than one mailbox can be specified with a comma-delimited string. - app_voicemail mailboxes must be specified as mailbox@context; - for example: mailboxes=6001@default. For mailboxes provided by external sources, - such as through the res_mwi_external module, you must specify strings supported by - the external system. - - For endpoints that cannot SUBSCRIBE for MWI, you can set the mailboxes option in your - endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint. - - - - The voicemail extension to send in the NOTIFY Message-Account header - - - Maximum time to keep an AoR - - Maximum time to keep a peer with explicit expiration. Time in seconds. - - - - Maximum number of contacts that can bind to an AoR - - Maximum number of contacts that can associate with this AoR. This value does - not affect the number of contacts that can be added with the "contact" option. - It only limits contacts added through external interaction, such as - registration. - - The rewrite_contact option - registers the source address as the contact address to help with - NAT and reusing connection oriented transports such as TCP and - TLS. Unfortunately, refreshing a registration may register a - different contact address and exceed - max_contacts. The - remove_existing and - remove_unavailable options can help by - removing either the soonest to expire or unavailable contact(s) over - max_contacts which is likely the - old rewrite_contact contact source - address being refreshed. - - This should be set to 1 and - remove_existing set to yes if you - wish to stick with the older chan_sip behaviour. - - - - - Minimum keep alive time for an AoR - - Minimum time to keep a peer with an explicit expiration. Time in seconds. - - - - Determines whether new contacts replace existing ones. - - On receiving a new registration to the AoR should it remove enough - existing contacts not added or updated by the registration to - satisfy max_contacts? Any removed - contacts will expire the soonest. - - The rewrite_contact option - registers the source address as the contact address to help with - NAT and reusing connection oriented transports such as TCP and - TLS. Unfortunately, refreshing a registration may register a - different contact address and exceed - max_contacts. The - remove_existing option can help by - removing the soonest to expire contact(s) over - max_contacts which is likely the - old rewrite_contact contact source - address being refreshed. - - This should be set to yes and - max_contacts set to 1 if you - wish to stick with the older chan_sip behaviour. - - - - - Determines whether new contacts should replace unavailable ones. - - The effect of this setting depends on the setting of - remove_existing. - If remove_existing is set to - no (default), setting remove_unavailable to - yes will remove only unavailable contacts that exceed - max_contacts to allow an incoming - REGISTER to complete sucessfully. - If remove_existing is set to - yes, setting remove_unavailable to - yes will prioritize unavailable contacts for removal - instead of just removing the contact that expires the soonest. - See remove_existing and - max_contacts for further information about how - these 3 settings interact. - - - - - Must be of type 'aor'. - - - Interval at which to qualify an AoR - - Interval between attempts to qualify the AoR for reachability. - If 0 never qualify. Time in seconds. - - - - Timeout for qualify - - If the contact doesn't respond to the OPTIONS request before the timeout, - the contact is marked unavailable. - If 0 no timeout. Time in fractional seconds. - - - - Authenticates a qualify challenge response if needed - - If true and a qualify request receives a challenge response then - authentication is attempted before declaring the contact available. - - This option does nothing as we will always complete - the challenge response authentication if the qualify request is - challenged. - - - - - Outbound proxy used when sending OPTIONS request - - If set the provided URI will be used as the outbound proxy when an - OPTIONS request is sent to a contact for qualify purposes. - - - - Enables Path support for REGISTER requests and Route support for other requests. - - When this option is enabled, the Path headers in register requests will be saved - and its contents will be used in Route headers for outbound out-of-dialog requests - and in Path headers for outbound 200 responses. Path support will also be indicated - in the Supported header. - - - - - Options that apply to the SIP stack as well as other system-wide settings - - The settings in this section are global. In addition to being global, the values will - not be re-evaluated when a reload is performed. This is because the values must be set - 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 - requests that receive no response when using an unreliable transport (e.g. UDP). - 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 - request before terminating the transaction. It is recommended that this be set - to 64 * Timer T1, but it may be set higher if desired. For more information on - this timer, see RFC 3261, Section 17.1.1.1. - - - - Use the short forms of common SIP header names. - - - Initial number of threads in the res_pjsip threadpool. - - - The amount by which the number of threads is incremented when necessary. - - - Number of seconds before an idle thread should be disposed of. - - - Maximum number of threads in the res_pjsip threadpool. - A value of 0 indicates no maximum. - - - Disable automatic switching from UDP to TCP transports. - - Disable automatic switching from UDP to TCP transports if outgoing - request is too large. See RFC 3261 section 18.1.1. - - - - Follow SDP forked media when To tag is different - - On outgoing calls, if the UAS responds with different SDP attributes - on subsequent 18X or 2XX responses (such as a port update) AND the - To tag on the subsequent response is different than that on the previous - one, follow it. - - - This option must also be enabled on endpoints that require - this functionality. - - - - - Follow SDP forked media when To tag is the same - - On outgoing calls, if the UAS responds with different SDP attributes - on non-100rel 18X or 2XX responses (such as a port update) AND the - To tag on the subsequent response is the same as that on the previous one, - process the updated SDP. - - - This option must also be enabled on endpoints that require - this functionality. - - - - - Disable the use of rport in outgoing requests. - - Remove "rport" parameter from the outgoing requests. - - - - Must be of type 'system' UNLESS the object name is 'system'. - - - - Options that apply globally to all SIP communications - - 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. - - - The interval (in seconds) to send keepalives to active connection-oriented transports. - - - The interval (in seconds) to check for expired contacts. - - - Disable Multi Domain support - - If disabled it can improve realtime performance by reducing the number of database requests. - - - - The maximum amount of time from startup that qualifies should be attempted on all contacts. - If greater than the qualify_frequency for an aor, qualify_frequency will be used instead. - - - The number of seconds over which to accumulate unidentified requests. - - If unidentified_request_count unidentified requests are received - during unidentified_request_period, a security event will be generated. - - - - The number of unidentified requests from a single IP to allow. - - If unidentified_request_count unidentified requests are received - during unidentified_request_period, a security event will be generated. - - - - The interval at which unidentified requests are older than - twice the unidentified_request_period are pruned. - - - Must be of type 'global' UNLESS the object name is 'global'. - - - Value used in User-Agent header for SIP requests and Server header for SIP responses. - - - When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given - peer who registers or unregisters with us. - - - Endpoint to use when sending an outbound request to a URI without a specified endpoint. - - - The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor - - - Enable/Disable SIP debug logging. Valid options include yes, no, or - a host address - - - The order by which endpoint identifiers are processed and checked. - Identifier names are usually derived from and can be found in the endpoint - identifier module itself (res_pjsip_endpoint_identifier_*). - You can use the CLI command "pjsip show identifiers" to see the - identifiers currently available. - - - One of the identifiers is "auth_username" which matches on the username in - an Authentication header. This method has some security considerations because an - Authentication header is not present on the first message of a dialog when - digest authentication is used. The client can't generate it until the server - sends the challenge in a 401 response. Since Asterisk normally sends a security - event when an incoming request can't be matched to an endpoint, using auth_username - requires that the security event be deferred until a request is received with - the Authentication header and only generated if the username doesn't result in a - match. This may result in a delay before an attack is recognized. You can control - how many unmatched requests are received from a single ip address before a security - event is generated using the unidentified_request parameters. - - - - - When Asterisk generates an outgoing SIP request, the From header username will be - set to this value if there is no better option (such as CallerID) to be - used. - - - When Asterisk generates a challenge, the digest realm will be - set to this value if there is no better option (such as auth/realm) to be - used. - - - MWI taskprocessor high water alert trigger level. - - On a heavily loaded system you may need to adjust the - taskprocessor queue limits. If any taskprocessor queue size - reaches its high water level then pjsip will stop processing - new requests until the alert is cleared. The alert clears - when all alerting taskprocessor queues have dropped to their - low water clear level. - - - - - MWI taskprocessor low water clear alert level. - - On a heavily loaded system you may need to adjust the - taskprocessor queue limits. If any taskprocessor queue size - reaches its high water level then pjsip will stop processing - new requests until the alert is cleared. The alert clears - when all alerting taskprocessor queues have dropped to their - low water clear level. - - Set to -1 for the low water level to be 90% of - the high water level. - - - - Enable/Disable sending unsolicited MWI to all endpoints on startup. - - When the initial unsolicited MWI notification are - enabled on startup then the initial notifications - get sent at startup. If you have a lot of endpoints - (thousands) that use unsolicited MWI then you may - want to consider disabling the initial startup - notifications. - - When the initial unsolicited MWI notifications are - disabled on startup then the notifications will start - on the endpoint's next contact update. - - - - - Enable/Disable ignoring SIP URI user field options. - - If you have this option enabled and there are semicolons - in the user field of a SIP URI then the field is truncated - at the first semicolon. This effectively makes the semicolon - a non-usable character for PJSIP endpoint names, extensions, - and AORs. This can be useful for improving compatibility with - an ITSP that likes to use user options for whatever reason. - - - sip:1235557890;phone-context=national@x.x.x.x;user=phone - - - 1235557890;phone-context=national - - - 1235557890 - - The caller-id and redirecting number strings - obtained from incoming SIP URI user fields are always truncated - at the first semicolon. - - - - Place caller-id information into Contact header - - This option will cause Asterisk to place caller-id information into - generated Contact headers. - - - - Enable sending AMI ContactStatus event when a device refreshes its registration. - - - Trigger scope for taskprocessor overloads - - This option specifies the trigger the distributor will use for - detecting taskprocessor overloads. When it detects an overload condition, - the distrubutor will stop accepting new requests until the overload is - cleared. - - - (default) Any taskprocessor overload will trigger. - Only pjsip taskprocessor overloads will trigger. - No overload detection will be performed. - - - The "none" and "pjsip_only" options should be used - with extreme caution and only to mitigate specific issues. - Under certain conditions they could make things worse. - - - - - Advertise support for RFC4488 REFER subscription suppression - - - - - - - Qualify a chan_pjsip endpoint. - - - - - The endpoint you want to qualify. - - - - Qualify a chan_pjsip endpoint. - - - - - Provide details about an identify section. - - - The object's type. This will always be 'identify'. - - - The name of this object. - - - - - - - - - - - - - - - The name of the endpoint associated with this information. - - - - - - - Provide details about an Address of Record (AoR) section. - - - The object's type. This will always be 'aor'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - The total number of contacts associated with this AoR. - - - The number of non-permanent contacts associated with this AoR. - - - The name of the endpoint associated with this information. - - - - - - - Provide details about an authentication section. - - - The object's type. This will always be 'auth'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - The name of the endpoint associated with this information. - - - - - - - Provide details about an authentication section. - - - The object's type. This will always be 'transport'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - The name of the endpoint associated with this information. - - - - - - - Provide details about an endpoint section. - - - The object's type. This will always be 'endpoint'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - The aggregate device state for this endpoint. - - - The number of active channels associated with this endpoint. - - - - - - - - - - - - - Provide details about an Address of Record (AoR) section. - - - The object's type. This will always be 'aor'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Provide details about an Address of Record (Auth) section. - - - The object's type. This will always be 'auth'. - - - The name of this object. - - - - - - - - - - - - - - - - - - - - - - - - - Provide details about a contact section. - - - The object's type. This will always be 'contact'. - - - The name of this object. - - - IP address of the last Via header in REGISTER request. - Will only appear in the event if available. - - - Port number of the last Via header in REGISTER request. - Will only appear in the event if available. - - - The elapsed time in decimal seconds after which an OPTIONS - message is sent before the contact is considered unavailable. - - - Content of the Call-ID header in REGISTER request. - Will only appear in the event if available. - - - Asterisk Server name. - - - If true delete the contact on Asterisk restart/boot. - - - The Path header received on the REGISTER. - - - The name of the endpoint associated with this information. - - - A boolean indicating whether a qualify should be authenticated. - - - This contact's URI. - - - The interval in seconds at which the contact will be qualified. - - - Content of the User-Agent header in REGISTER request - - - Absolute time that this contact is no longer valid after - - - The contact's outbound proxy. - - - This contact's status. - - - - - - - - - The round trip time in microseconds. - - - - - - - Provide details about a contact's status. - - - The AoR that owns this contact. - - - This contact's URI. - - - This contact's status. - - - - - - - - - The round trip time in microseconds. - - - The name of the endpoint associated with this information. - - - Content of the User-Agent header in REGISTER request - - - Absolute time that this contact is no longer valid after - - - IP address:port of the last Via header in REGISTER request. - Will only appear in the event if available. - - - Content of the Call-ID header in REGISTER request. - Will only appear in the event if available. - - - The sorcery ID of the contact. - - - A boolean indicating whether a qualify should be authenticated. - - - The contact's outbound proxy. - - - The Path header received on the REGISTER. - - - The interval in seconds at which the contact will be qualified. - - - The elapsed time in decimal seconds after which an OPTIONS - message is sent before the contact is considered unavailable. - - - - - - - Provide details about a contact's status. - - - The object's type. This will always be 'endpoint'. - - - The name of this object. - - - The transport configurations associated with this endpoint. - - - The aor configurations associated with this endpoint. - - - The inbound authentication configurations associated with this endpoint. - - - The outbound authentication configurations associated with this endpoint. - - - The aggregate device state for this endpoint. - - - The number of active channels associated with this endpoint. - - - - - - - Lists PJSIP endpoints. - - - - - Provides a listing of all endpoints. For each endpoint an EndpointList event - is raised that contains relevant attributes and status information. Once all - endpoints have been listed an EndpointListComplete event is issued. - - - - - - - - - Provide final information about an endpoint list. - - - - - - - - - - - Detail listing of an endpoint and its objects. - - - - - The endpoint to list. - - - - - Provides a detailed listing of options for a given endpoint. Events are issued - showing the configuration and status of the endpoint and associated objects. These - events include EndpointDetail, AorDetail, - AuthDetail, TransportDetail, and - IdentifyDetail. Some events may be listed multiple times if multiple objects are - associated (for instance AoRs). Once all detail events have been raised a final - EndpointDetailComplete event is issued. - - - - - - - - - - - - - - Provide final information about endpoint details. - - - - - - - - - - - Lists PJSIP AORs. - - - - - Provides a listing of all AORs. For each AOR an AorList event - is raised that contains relevant attributes and status information. Once all - aors have been listed an AorListComplete event is issued. - - - - - - - - - Provide final information about an aor list. - - - - - - - - - - - Lists PJSIP Auths. - - - - Provides a listing of all Auths. For each Auth an AuthList event - is raised that contains relevant attributes and status information. Once all - auths have been listed an AuthListComplete event is issued. - - - - - - - - - Provide final information about an auth list. - - - - - - - - - - - Lists PJSIP Contacts. - - - - Provides a listing of all Contacts. For each Contact a ContactList - event is raised that contains relevant attributes and status information. - Once all contacts have been listed a ContactListComplete event - is issued. - - - - - - - - - Provide final information about a contact list. - - - - - - - - - - ***/ - #define MOD_DATA_CONTACT "contact" /*! Number of serializers in pool if one not supplied. */ diff --git a/res/res_pjsip/pjsip_config.xml b/res/res_pjsip/pjsip_config.xml new file mode 100644 index 0000000000..73b77db06f --- /dev/null +++ b/res/res_pjsip/pjsip_config.xml @@ -0,0 +1,2346 @@ + + + + + SIP Resource using PJProject + + + Endpoint + + The Endpoint is the primary configuration object. + It contains the core SIP related options only, endpoints are NOT + dialable entries of their own. Communication with another SIP device is + accomplished via Addresses of Record (AoRs) which have one or more + contacts associated with them. Endpoints NOT configured to + use a transport will default to first transport found + in pjsip.conf that matches its type. + + Example: An Endpoint has been configured with no transport. + When it comes time to call an AoR, PJSIP will find the + first transport that matches the type. A SIP URI of sip:5000@[11::33] + will use the first IPv6 transport and try to send the request. + + If the anonymous endpoint identifier is in use an endpoint with the name + "anonymous@domain" will be searched for as a last resort. If this is not found + it will fall back to searching for "anonymous". If neither endpoints are found + the anonymous endpoint identifier will not return an endpoint and anonymous + calling will not be possible. + + + + Allow support for RFC3262 provisional ACK tags + + + + + + + + + + Condense MWI notifications into a single NOTIFY. + When enabled, aggregate_mwi condenses message + waiting notifications from multiple mailboxes into a single NOTIFY. If it is disabled, + individual NOTIFYs are sent for each mailbox. + + + Media Codec(s) to allow + + + Codec negotiation prefs for incoming offers. + + + This is a string that describes how the codecs + specified on an incoming SDP offer (pending) are reconciled with the codecs specified + on an endpoint (configured) before being sent to the Asterisk core. + The string actually specifies 4 name:value pair parameters + separated by commas. Whitespace is ignored and they may be specified in any order. + Note that this option is reserved for future functionality. + + + + Parameters: + + + + + + + The codec list from the caller. (default) + The codec list from the endpoint. + + + + + + + Only common codecs with the preferred codecs first. (default) + Use only the preferred codecs. + Use only the non-preferred codecs. + + + + + + + After the operation, keep all codecs. (default) + After the operation, keep only the first codec. + + + + + + + Allow transcoding. (default) + Prevent transcoding. + + + + + + + codec_prefs_incoming_offer = prefer: pending, operation: intersect, keep: all, transcode: allow + + + Prefer the codecs coming from the caller. Use only the ones that are common. + keeping the order of the preferred list. Keep all codecs in the result. Allow transcoding. + + + + + Codec negotiation prefs for outgoing offers. + + + This is a string that describes how the codecs specified in the topology that + comes from the Asterisk core (pending) are reconciled with the codecs specified on an + endpoint (configured) when sending an SDP offer. + The string actually specifies 4 name:value pair parameters + separated by commas. Whitespace is ignored and they may be specified in any order. + Note that this option is reserved for future functionality. + + + + Parameters: + + + + + + + The codec list from the core. (default) + The codec list from the endpoint. + + + + + + + Merge the lists with the preferred codecs first. (default) + Only common codecs with the preferred codecs first. (default) + Use only the preferred codecs. + Use only the non-preferred codecs. + + + + + + + After the operation, keep all codecs. (default) + After the operation, keep only the first codec. + + + + + + + Allow transcoding. (default) + Prevent transcoding. + + + + + + + codec_prefs_outgoing_offer = prefer: configured, operation: union, keep: first, transcode: prevent + + + Prefer the codecs coming from the endpoint. Merge them with the codecs from the core + keeping the order of the preferred list. Keep only the first one. No transcoding allowed. + + + + + Codec negotiation prefs for incoming answers. + + + This is a string that describes how the codecs specified in an incoming SDP answer + (pending) are reconciled with the codecs specified on an endpoint (configured) + when receiving an SDP answer. + The string actually specifies 4 name:value pair parameters + separated by commas. Whitespace is ignored and they may be specified in any order. + Note that this option is reserved for future functionality. + + + + Parameters: + + + + + + + The codec list in the received SDP answer. (default) + The codec list from the endpoint. + + + + + + + Merge the lists with the preferred codecs first. + Only common codecs with the preferred codecs first. (default) + Use only the preferred codecs. + Use only the non-preferred codecs. + + + + + + + After the operation, keep all codecs. (default) + After the operation, keep only the first codec. + + + + + The transcode parameter is ignored when processing answers. + + + + + + + codec_prefs_incoming_answer = keep: first + + + Use the defaults but keep oinly the first codec. + + + + + Codec negotiation prefs for outgoing answers. + + + This is a string that describes how the codecs that come from the core (pending) + are reconciled with the codecs specified on an endpoint (configured) + when sending an SDP answer. + The string actually specifies 4 name:value pair parameters + separated by commas. Whitespace is ignored and they may be specified in any order. + Note that this option is reserved for future functionality. + + + + Parameters: + + + + + + + The codec list that came from the core. (default) + The codec list from the endpoint. + + + + + + + Merge the lists with the preferred codecs first. + Only common codecs with the preferred codecs first. (default) + Use only the preferred codecs. + Use only the non-preferred codecs. + + + + + + + After the operation, keep all codecs. (default) + After the operation, keep only the first codec. + + + + + The transcode parameter is ignored when processing answers. + + + + + + + codec_prefs_incoming_answer = keep: first + + + Use the defaults but keep oinly the first codec. + + + + + Enable RFC3578 overlap dialing support. + + + AoR(s) to be used with the endpoint + + List of comma separated AoRs that the endpoint should be associated with. + + + + Authentication Object(s) associated with the endpoint + + This is a comma-delimited list of auth sections defined + in pjsip.conf to be used to verify inbound connection attempts. + + Endpoints without an authentication object + configured will allow connections without verification. + + Using the same auth section for inbound and outbound + authentication is not recommended. There is a difference in + meaning for an empty realm setting between inbound and outbound + authentication uses. See the auth realm description for details. + + + + + CallerID information for the endpoint + + Must be in the format Name <Number>, + or only <Number>. + + + + Default privacy level + + + + + + + + + + + + + + + + Internal id_tag for the endpoint + + + Dialplan context for inbound sessions + + + Mitigation of direct media (re)INVITE glare + + + This setting attempts to avoid creating INVITE glare scenarios + by disabling direct media reINVITEs in one direction thereby allowing + designated servers (according to this option) to initiate direct + media reINVITEs without contention and significantly reducing call + setup time. + + + A more detailed description of how this option functions can be found on + the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance + + + + + + + + + + Direct Media method type + + Method for setting up Direct Media between endpoints. + + + + Alias for the invite value. + + + + + + + Accept Connected Line updates from this endpoint + + + Send Connected Line updates to this endpoint + + + Connected line method type + + Method used when updating connected line information. + + + When set to invite, check the remote's Allow header and + if UPDATE is allowed, send UPDATE instead of INVITE to avoid SDP + renegotiation. If UPDATE is not Allowed, send INVITE. + + + Alias for the invite value. + + + If set to update, send UPDATE regardless of what the remote + Allows. + + + + + + Determines whether media may flow directly between endpoints. + + + Disable direct media session refreshes when NAT obstructs the media session + + + Media Codec(s) to disallow + + + DTMF mode + + This setting allows to choose the DTMF mode for endpoint communication. + + + DTMF is sent out of band of the main audio stream. This + supercedes the older RFC-2833 used within + the older chan_sip. + + + DTMF is sent as part of audio stream. + + + DTMF is sent as SIP INFO packets. + + + DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not. + + + DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not. + + + + + + IP address used in SDP for media handling + + At the time of SDP creation, the IP address defined here will be used as + the media address for individual streams in the SDP. + + + Be aware that the external_media_address option, set in Transport + configuration, can also affect the final media address used in the SDP. + + + + + Bind the RTP instance to the media_address + + If media_address is specified, this option causes the RTP instance to be bound to the + specified ip address which causes the packets to be sent from that address. + + + + + Force use of return port + + + Enable the ICE mechanism to help traverse NAT + + + Way(s) for the endpoint to be identified + + Endpoints and AORs can be identified in multiple ways. This + option is a comma separated list of methods the endpoint can be + identified. + + + This option controls both how an endpoint is matched for incoming + traffic and also how an AOR is determined if a registration + occurs. You must list at least one method that also matches for + AORs or the registration will fail. + + + + Matches the endpoint or AOR ID based on the username + and domain in the From header (or To header for AORs). If + an exact match on both username and domain/realm fails, the + match is retried with just the username. + + + + Matches the endpoint or AOR ID based on the username + and realm in the Authentication header. If an exact match + on both username and domain/realm fails, the match is + retried with just the username. + + This method of identification has some security + considerations because an Authentication header is not + present on the first message of a dialog when digest + authentication is used. The client can't generate it until + the server sends the challenge in a 401 response. Since + Asterisk normally sends a security event when an incoming + request can't be matched to an endpoint, using this method + requires that the security event be deferred until a request + is received with the Authentication header and only + generated if the username doesn't result in a match. This + may result in a delay before an attack is recognized. You + can control how many unmatched requests are received from + a single ip address before a security event is generated + using the unidentified_request + parameters in the "global" configuration object. + + + + Matches the endpoint based on the source IP address. + + This method of identification is not configured here + but simply allowed by this configuration option. See the + documentation for the identify + configuration section for more details on this method of + endpoint identification. + + + + Matches the endpoint based on a configured SIP header + value. + + This method of identification is not configured here + but simply allowed by this configuration option. See the + documentation for the identify + configuration section for more details on this method of + endpoint identification. + + + + + + + How redirects received from an endpoint are handled + + When a redirect is received from an endpoint there are multiple ways it can be handled. + If this option is set to user the user portion of the redirect target + is treated as an extension within the dialplan and dialed using a Local channel. If this option + is set to uri_core the target URI is returned to the dialing application + which dials it using the PJSIP channel driver and endpoint originally used. If this option is + set to uri_pjsip the redirect occurs within chan_pjsip itself and is not exposed + to the core at all. The uri_pjsip option has the benefit of being more efficient + and also supporting multiple potential redirect targets. The con is that since redirection occurs + within chan_pjsip redirecting information is not forwarded and redirection can not be + prevented. + + + + + + + + + + NOTIFY the endpoint when state changes for any of the specified mailboxes + + Asterisk will send unsolicited MWI NOTIFY messages to the endpoint when state + changes happen for any of the specified mailboxes. More than one mailbox can be + specified with a comma-delimited string. app_voicemail mailboxes must be specified + as mailbox@context; for example: mailboxes=6001@default. For mailboxes provided by + external sources, such as through the res_mwi_external module, you must specify + strings supported by the external system. + + For endpoints that SUBSCRIBE for MWI, use the mailboxes option in your AOR + configuration. + + + + An MWI subscribe will replace sending unsolicited NOTIFYs + + + The voicemail extension to send in the NOTIFY Message-Account header + + + Default Music On Hold class + + + Authentication object(s) used for outbound requests + + This is a comma-delimited list of auth + sections defined in pjsip.conf used to respond + to outbound connection authentication challenges. + + Using the same auth section for inbound and outbound + authentication is not recommended. There is a difference in + meaning for an empty realm setting between inbound and outbound + authentication uses. See the auth realm description for details. + + + + + Full SIP URI of the outbound proxy used to send requests + + + Allow Contact header to be rewritten with the source IP address-port + + On inbound SIP messages from this endpoint, the Contact header or an + appropriate Record-Route header will be changed to have the source IP + address and port. This option does not affect outbound messages sent to + this endpoint. This option helps servers communicate with endpoints + that are behind NATs. This option also helps reuse reliable transport + connections such as TCP and TLS. + + + + Allow use of IPv6 for RTP traffic + + + Enforce that RTP must be symmetric + + + Send the Diversion header, conveying the diversion + information to the called user agent + + + Send the History-Info header, conveying the diversion + information to the called and calling user agents + + + Send the P-Asserted-Identity header + + + Send the Remote-Party-ID header + + + Immediately send connected line updates on unanswered incoming calls. + + When enabled, immediately send 180 Ringing + or 183 Progress response messages to the + caller if the connected line information is updated before + the call is answered. This can send a 180 Ringing + response before the call has even reached the far end. The + caller can start hearing ringback before the far end even gets + the call. Many phones tend to grab the first connected line + information and refuse to update the display if it changes. The + first information is not likely to be correct if the call + goes to an endpoint not under the control of this Asterisk + box. + When disabled, a connected line update must wait for + another reason to send a message with the connected line + information to the caller before the call is answered. You can + trigger the sending of the information by using an appropriate + dialplan application such as Ringing. + + + + Minimum session timers expiration period + + Minimum session timer expiration period. Time in seconds. + + + + Session timers for SIP packets + + + + + + + Alias of always + + + + + Maximum session timer expiration period + + Maximum session timer expiration period. Time in seconds. + + + + Explicit transport configuration to use + + This will force the endpoint to use the + specified transport configuration to send SIP messages. You need + to already know what kind of transport (UDP/TCP/IPv4/etc) the + endpoint device will use. + + Not specifying a transport will select the first + configured transport in pjsip.conf which is + compatible with the URI we are trying to contact. + + Transport configuration is not affected by reloads. In order to + change transports, a full Asterisk restart is required + + + + Accept identification information received from this endpoint + This option determines whether Asterisk will accept + identification from the endpoint from headers such as P-Asserted-Identity + or Remote-Party-ID header. This option applies both to calls originating from the + endpoint and calls originating from Asterisk. If no, the + configured Caller-ID from pjsip.conf will always be used as the identity for + the endpoint. + + + Send private identification details to the endpoint. + This option determines whether res_pjsip will send private + identification information to the endpoint. If no, + private Caller-ID information will not be forwarded to the endpoint. + "Private" in this case refers to any method of restricting identification. + Example: setting callerid_privacy to any + prohib variation. + Example: If trust_id_inbound is set to + yes, the presence of a Privacy: id + header in a SIP request or response would indicate the identification + provided in the request is private. + + + Must be of type 'endpoint'. + + + Use Endpoint's requested packetization interval + + + Determines whether res_pjsip will use and enforce usage of AVPF for this + endpoint. + + If set to yes, res_pjsip will use the AVPF or SAVPF RTP + profile for all media offers on outbound calls and media updates and will + decline media offers not using the AVPF or SAVPF profile. + + If set to no, res_pjsip will use the AVP or SAVP RTP + profile for all media offers on outbound calls and media updates, and will + decline media offers not using the AVP or SAVP profile. + + + + Determines whether res_pjsip will use and enforce usage of AVP, + regardless of the RTP profile in use for this endpoint. + + If set to yes, res_pjsip will use the AVP, AVPF, SAVP, or + SAVPF RTP profile for all media offers on outbound calls and media updates including + those for DTLS-SRTP streams. + + If set to no, res_pjsip will use the respective RTP profile + depending on configuration. + + + + Determines whether res_pjsip will use the media transport received in the + offer SDP in the corresponding answer SDP. + + If set to yes, res_pjsip will use the received media transport. + + If set to no, res_pjsip will use the respective RTP profile + depending on configuration. + + + + Determines whether res_pjsip will use and enforce usage of media encryption + for this endpoint. + + + + res_pjsip will offer no encryption and allow no encryption to be setup. + + + res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP + transport should be used in conjunction with this option to prevent + exposure of media encryption keys. + + + res_pjsip will offer DTLS-SRTP setup. + + + + + + Determines whether encryption should be used if possible but does not terminate the + session if not achieved. + + This option only applies if media_encryption is + set to sdes or dtls. + + + + Force g.726 to use AAL2 packing order when negotiating g.726 audio + + When set to "yes" and an endpoint negotiates g.726 audio then use g.726 for AAL2 + packing order instead of what is recommended by RFC3551. Since this essentially + replaces the underlying 'g726' codec with 'g726aal2' then 'g726aal2' needs to be + specified in the endpoint's allowed codec list. + + + + Determines whether chan_pjsip will indicate ringing using inband + progress. + + If set to yes, chan_pjsip will send a 183 Session Progress + when told to indicate ringing and will immediately start sending ringing + as audio. + + If set to no, chan_pjsip will send a 180 Ringing when told + 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 + + + + No error correction should be used. + + + Forward error correction should be used. + + + Redundancy error correction should be used. + + + + + + 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. + + + + How long into a call before fax_detect is disabled for the call + + The option determines how many seconds into a call before the + fax_detect option is disabled for the call. Setting the value + to zero disables the timeout. + + + + 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. + + + + Bind the UDPTL instance to the media_adress + + If media_address is specified, this option causes the UDPTL instance to be bound to + the specified ip address which causes the packets to be sent from that address. + + + + Set which country's indications to use for channels created for this endpoint. + + + Set the default language to use for channels created for this endpoint. + + + Determines whether one-touch recording is allowed for this endpoint. + + record_on_feature + record_off_feature + + + + 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 + feature will be enabled for the channel. The feature designated here can be any built-in + or dynamic feature defined in features.conf. + This setting has no effect if the endpoint's one_touch_recording option is disabled + + + one_touch_recording + record_off_feature + + + + 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 + feature will be enabled for the channel. The feature designated here can be any built-in + or dynamic feature defined in features.conf. + This setting has no effect if the endpoint's one_touch_recording option is disabled + + + one_touch_recording + record_on_feature + + + + Name of the RTP engine to use for channels created for this endpoint + + + Determines whether SIP REFER transfers are allowed for this endpoint + + + Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number + + + Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side + + + String placed as the username portion of an SDP origin (o=) line. + + + String used for the SDP session (s=) line. + + + DSCP TOS bits for audio streams + + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + + + + DSCP TOS bits for video streams + + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + + + + Priority for audio streams + + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for more information about QoS settings + + + + Priority for video streams + + 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. + + It can be one of the following values: + + + meaning no verification is done. + + + meaning to verify the remote fingerprint. + + + meaning to verify the remote certificate. + + + meaning to verify both the remote fingerprint and certificate. + + + + + + Interval at which to renegotiate the TLS session and rekey the SRTP session + + This option only applies if media_encryption is + set to dtls. + + If this is not set or the value provided is 0 rekeying will be disabled. + + + + Whether or not to automatically generate an ephemeral X.509 certificate + + + If enabled, Asterisk will generate an X.509 certificate for each DTLS session. + This option only applies if media_encryption is set + to dtls. This option will be automatically enabled if + webrtc is enabled and dtls_cert_file is + not specified. + + + + + 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 + set to dtls. + + Many options for acceptable ciphers. See link for more: + 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. + + + This option only applies if media_encryption is + set to dtls. + + + + res_pjsip will make a connection to the peer. + + + res_pjsip will accept connections from the peer. + + + res_pjsip will offer and accept connections from the peer. + + + + + + Type of hash to use for the DTLS fingerprint in the SDP. + + + This option only applies if media_encryption is + set to dtls. + + + + + + + + + Determines whether 32 byte tags should be used instead of 80 byte tags. + + This option only applies if media_encryption is + set to sdes or dtls. + + + + Variable set on a channel involving the endpoint. + + When a new channel is created using the endpoint set the specified + variable(s) on that channel. For multiple channel variables specify + multiple 'set_var'(s). + + + + Context to route incoming MESSAGE requests to. + + If specified, incoming MESSAGE requests will be routed to the indicated + dialplan context. If no message_context is + specified, then the context setting is used. + + + + An accountcode to set automatically on any channels created for this endpoint. + + If specified, any channel created for this endpoint will automatically + have this accountcode set on it. + + + + Respond to a SIP invite with the single most preferred codec (DEPRECATED) + Respond to a SIP invite with the single most preferred codec + rather than advertising all joint codec capabilities. This limits the other side's codec + choice to exactly what we prefer. + This option has been deprecated in favor of + incoming_call_offer_pref. Setting both options is unsupported. + + + + incoming_call_offer_pref + + + + Preferences for selecting codecs for an incoming call. + + Based on this setting, a joint list of preferred codecs between those + received in an incoming SDP offer (remote), and those specified in the + endpoint's "allow" parameter (local) es created and is passed to the Asterisk + core. + This list will consist of only those codecs found in both lists. + + + Include all codecs in the local list that are also in the remote list + preserving the local order. (default). + + + Include only the first codec in the local list that is also in the remote list. + + + Include all codecs in the remote list that are also in the local list + preserving the remote order. + + + Include only the first codec in the remote list that is also in the local list. + + + + + + Preferences for selecting codecs for an outgoing call. + + Based on this setting, a joint list of preferred codecs between + those received from the Asterisk core (remote), and those specified in + the endpoint's "allow" parameter (local) is created and is used to create + the outgoing SDP offer. + + + Include all codecs in the local list that are also in the remote list + preserving the local order. + + + Include all codecs in the local list preserving the local order. + + + Include only the first codec in the local list. + + + Include all codecs in the remote list that are also in the local list + preserving the remote order. + + + Include all codecs in the local list preserving the remote order. (default) + + + Include only the first codec in the remote list that is also in the local list. + + + + + + Number of seconds between RTP comfort noise keepalive packets. + + At the specified interval, Asterisk will send an RTP comfort noise frame. This may + be useful for situations where Asterisk is behind a NAT or firewall and must keep + a hole open in order to allow for media to arrive at Asterisk. + + + + Maximum number of seconds without receiving RTP (while off hold) before terminating call. + + This option configures the number of seconds without RTP (while off hold) before + considering a channel as dead. When the number of seconds is reached the underlying + channel is hung up. By default this option is set to 0, which means do not check. + + + + Maximum number of seconds without receiving RTP (while on hold) before terminating call. + + This option configures the number of seconds without RTP (while on hold) before + considering a channel as dead. When the number of seconds is reached the underlying + channel is hung up. By default this option is set to 0, which means do not check. + + + + List of IP ACL section names in acl.conf + + This matches sections configured in acl.conf. The value is + defined as a list of comma-delimited section names. + + + + List of IP addresses to deny access from + + The value is a comma-delimited list of IP addresses. IP addresses may + have a subnet mask appended. The subnet mask may be written in either + CIDR or dotted-decimal notation. Separate the IP address and subnet + mask with a slash ('/') + + + + List of IP addresses to permit access from + + The value is a comma-delimited list of IP addresses. IP addresses may + have a subnet mask appended. The subnet mask may be written in either + CIDR or dotted-decimal notation. Separate the IP address and subnet + mask with a slash ('/') + + + + List of Contact ACL section names in acl.conf + + This matches sections configured in acl.conf. The value is + defined as a list of comma-delimited section names. + + + + List of Contact header addresses to deny + + The value is a comma-delimited list of IP addresses. IP addresses may + have a subnet mask appended. The subnet mask may be written in either + CIDR or dotted-decimal notation. Separate the IP address and subnet + mask with a slash ('/') + + + + List of Contact header addresses to permit + + The value is a comma-delimited list of IP addresses. IP addresses may + have a subnet mask appended. The subnet mask may be written in either + CIDR or dotted-decimal notation. Separate the IP address and subnet + mask with a slash ('/') + + + + Context for incoming MESSAGE requests. + + If specified, incoming SUBSCRIBE requests will be searched for the matching + extension in the indicated context. + If no subscribe_context is specified, + then the context setting is used. + + + + Force the user on the outgoing Contact header to this value. + + On outbound requests, force the user portion of the Contact header to this value. + + + + Allow the sending and receiving RTP codec to differ + + When set to "yes" the codec in use for sending will be allowed to differ from + that of the received one. PJSIP will not automatically switch the sending one + to the receiving one. + + + + Enable RFC 5761 RTCP multiplexing on the RTP port + + With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" + attribute on all media streams. This will result in RTP and RTCP being sent and received + on the same port. This shifts the demultiplexing logic to the application rather than + the transport layer. This option is useful when interoperating with WebRTC endpoints + since they mandate this option's use. + + + + Whether to notifies all the progress details on blind transfer + + Some SIP phones (Mitel/Aastra, Snom) expect a sip/frag "200 OK" + after REFER has been accepted. If set to no then asterisk + will not send the progress details, but immediately will send "200 OK". + + + + Whether to notifies dialog-info 'early' on InUse&Ringing state + + Control whether dialog-info subscriptions get 'early' state + on Ringing when already INUSE. + + + + The maximum number of allowed audio streams for the endpoint + + This option enforces a limit on the maximum simultaneous negotiated audio + streams allowed for the endpoint. + + + + The maximum number of allowed video streams for the endpoint + + This option enforces a limit on the maximum simultaneous negotiated video + streams allowed for the endpoint. + + + + Enable RTP bundling + + With this option enabled, Asterisk will attempt to negotiate the use of bundle. + If negotiated this will result in multiple RTP streams being carried over the same + underlying transport. Note that enabling bundle will also enable the rtcp_mux option. + + + + Defaults and enables some options that are relevant to WebRTC + + When set to "yes" this also enables the following values that are needed in + order for basic WebRTC support to work: rtcp_mux, use_avpf, ice_support, and + use_received_transport. The following configuration settings also get defaulted + as follows: + media_encryption=dtls + dtls_auto_generate_cert=yes (if dtls_cert_file is not set) + dtls_verify=fingerprint + dtls_setup=actpass + + + + Mailbox name to use when incoming MWI NOTIFYs are received + + If an MWI NOTIFY is received from this endpoint, + this mailbox will be used when notifying other modules of MWI status + changes. If not set, incoming MWI NOTIFYs are ignored. + + + + Follow SDP forked media when To tag is different + + On outgoing calls, if the UAS responds with different SDP attributes + on subsequent 18X or 2XX responses (such as a port update) AND the + To tag on the subsequent response is different than that on the previous + one, follow it. This usually happens when the INVITE is forked to multiple + UASs and more than one sends an SDP answer. + + + This option must also be enabled in the system + section for it to take effect here. + + + + + Accept multiple SDP answers on non-100rel responses + + On outgoing calls, if the UAS responds with different SDP attributes + on non-100rel 18X or 2XX responses (such as a port update) AND the + To tag on the subsequent response is the same as that on the previous one, + process the updated SDP. This can happen when the UAS needs to change ports + for some reason such as using a separate port for custom ringback. + + + This option must also be enabled in the system + section for it to take effect here. + + + + + Suppress Q.850 Reason headers for this endpoint + + Some devices can't accept multiple Reason headers and get confused + when both 'SIP' and 'Q.850' Reason headers are received. This + option allows the 'Q.850' Reason header to be suppressed. + + + + Do not forward 183 when it doesn't contain SDP + + Certain SS7 internetworking scenarios can result in a 183 + to be generated for reasons other than early media. Forwarding + this 183 can cause loss of ringback tone. This flag emulates + the behavior of chan_sip and prevents these 183 responses from + being forwarded. + + + + Enable STIR/SHAKEN support on this endpoint + + Enable STIR/SHAKEN support on this endpoint. On incoming INVITEs, + the Identity header will be checked for validity. On outgoing + INVITEs, an Identity header will be added. + + + + Skip authentication when receiving OPTIONS requests + + RFC 3261 says that the response to an OPTIONS request MUST be the + same had the request been an INVITE. Some UAs use OPTIONS requests + like a 'ping' and the expectation is that they will return a + 200 OK. + Enabling allow_unauthenticated_options + will skip authentication of OPTIONS requests for the given + endpoint. + There are security implications to enabling this setting as + it can allow information disclosure to occur - specifically, if + enabled, an external party could enumerate and find the endpoint + name by sending OPTIONS requests and examining the + responses. + + + + + Authentication type + + Authentication objects hold the authentication information for use + by other objects such as endpoints or registrations. + This also allows for multiple objects to use a single auth object. See + the auth_type config option for password style choices. + + + Authentication type + + This option specifies which of the password style config options should be read + when trying to authenticate an endpoint inbound request. If set to userpass + then we'll read from the 'password' option. For md5 we'll read + from 'md5_cred'. If set to google_oauth then we'll read from the + refresh_token/oauth_clientid/oauth_secret fields. The following values are valid: + + + + + + + + + + + This setting only describes whether the password is in + plain text or has been pre-hashed with MD5. It doesn't describe + the acceptable digest algorithms we'll accept in a received + challenge. + + + + + + Lifetime of a nonce associated with this authentication config. + + + MD5 Hash used for authentication. + + Only used when auth_type is md5. + As an alternative to specifying a plain text password, + you can hash the username, realm and password + together one time and place the hash value here. + The input to the hash function must be in the + following format: + + + + + <username>:<realm>:<password> + + + + + For incoming authentication (asterisk is the server), + the realm must match either the realm set in this object + or the default_realm set in in the + global object. + + + + + For outgoing authentication (asterisk is the UAC), + the realm must match what the server will be sending + in their WWW-Authenticate header. It can't be blank + unless you expect the server to be sending a blank + realm in the header. You can't use pre-hashed + passwords with a wildcard auth object. + You can generate the hash with the following shell + command: + + + + + $ echo -n "myname:myrealm:mypassword" | md5sum + + + + + Note the '-n'. You don't want a newline to be part + of the hash. + + + + Plain text password used for authentication. + Only used when auth_type is userpass. + + + OAuth 2.0 refresh token + + + OAuth 2.0 application's client id + + + OAuth 2.0 application's secret + + + SIP realm for endpoint + + For incoming authentication (asterisk is the UAS), + this is the realm to be sent on WWW-Authenticate + headers. If not specified, the global + object's default_realm will be used. + + + + + For outgoing authentication (asterisk is the UAC), this + must either be the realm the server is expected to send, + or left blank or contain a single '*' to automatically + use the realm sent by the server. If you have multiple + auth objects for an endpoint, the realm is also used to + match the auth object to the realm the server sent. + + + + + + Using the same auth section for inbound and outbound + authentication is not recommended. There is a difference in + meaning for an empty realm setting between inbound and outbound + authentication uses. + + + + + + + If more than one auth object with the same realm or + more than one wildcard auth object associated to + an endpoint, we can only use the first one of + each defined on the endpoint. + + + + + + Must be 'auth' + + + Username to use for account + + + + Domain Alias + + Signifies that a domain is an alias. If the domain on a session is + not found to match an AoR then this object is used to see if we have + an alias for the AoR to which the endpoint is binding. This objects + name as defined in configuration should be the domain alias and a + config option is provided to specify the domain to be aliased. + + + Must be of type 'domain_alias'. + + + Domain to be aliased + + + + SIP Transport + + Transports + + There are different transports and protocol derivatives + supported by res_pjsip. They are in order of + preference: UDP, TCP, and WebSocket (WS). + Changes to transport configuration in pjsip.conf will only be + effected on a complete restart of Asterisk. A module reload + will not suffice. + + + Number of simultaneous Asynchronous Operations + + + IP Address and optional port to bind to for this transport + + + File containing a list of certificates to read (TLS ONLY, not WSS) + + + Path to directory containing a list of certificates to read (TLS ONLY, not WSS) + + + Certificate file for endpoint (TLS ONLY, not WSS) + + A path to a .crt or .pem file can be provided. However, only + the certificate is read from the file, not the private key. + The priv_key_file option must supply a + matching key file. + + + + Preferred cryptography cipher names (TLS ONLY, not WSS) + + Comma separated list of cipher names or numeric equivalents. + Numeric equivalents can be either decimal or hexadecimal (0xX). + + There are many cipher names. Use the CLI command + pjsip list ciphers to see a list of cipher + names available for your installation. See link for more: + http://www.openssl.org/docs/apps/ciphers.html#CIPHER_SUITE_NAMES + + + + + Domain the transport comes from + + + External IP address to use in RTP handling + + When a request or response is sent out, if the destination of the + message is outside the IP network defined in the option localnet, + and the media address in the SDP is within the localnet network, then the + media address in the SDP will be rewritten to the value defined for + external_media_address. + + + + External address for SIP signalling + + + External port for SIP signalling + + + Method of SSL transport (TLS ONLY, not WSS) + + + + The default as defined by PJSIP. This is currently TLSv1, but may change with future releases. + + + This option is equivalent to setting 'default' + + + + + + + + + + + + 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 ('/'). + + + Password required for transport + + + Private key file (TLS ONLY, not WSS) + + + Protocol to use for SIP traffic + + + + + + + + + + + + + Require client certificate (TLS ONLY, not WSS) + + + Must be of type 'transport'. + + + Require verification of client certificate (TLS ONLY, not WSS) + + + Require verification of server certificate (TLS ONLY, not WSS) + + + Enable TOS for the signalling sent over this transport + + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service + for more information on this parameter. + This option does not apply to the ws + or the wss protocols. + + + + Enable COS for the signalling sent over this transport + + See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service + for more information on this parameter. + This option does not apply to the ws + or the wss protocols. + + + + The timeout (in milliseconds) to set on WebSocket connections. + + If a websocket connection accepts input slowly, the timeout + for writes to it can be increased to keep it from being disconnected. + Value is in milliseconds. + + + + Allow this transport to be reloaded. + + Allow this transport to be reloaded when res_pjsip is reloaded. + This option defaults to "no" because reloading a transport may disrupt + in-progress calls. + + + + Use the same transport for outgoing requests as incoming ones. + + When a request from a dynamic contact + comes in on a transport with this option set to 'yes', + the transport name will be saved and used for subsequent + outgoing requests like OPTIONS, NOTIFY and INVITE. It's + saved as a contact uri parameter named 'x-ast-txp' and will + display with the contact uri in CLI, AMI, and ARI output. + On the outgoing request, if a transport wasn't explicitly + set on the endpoint AND the request URI is not a hostname, + the saved transport will be used and the 'x-ast-txp' + parameter stripped from the outgoing packet. + + + + + + A way of creating an aliased name to a SIP URI + + Contacts are a way to hide SIP URIs from the dialplan directly. + They are also used to make a group of contactable parties when + in use with AoR lists. + + + Must be of type 'contact'. + + + SIP URI to contact peer + + + Time to keep alive a contact + + Time to keep alive a contact. String style specification. + + + + Interval at which to qualify a contact + + Interval between attempts to qualify the contact for reachability. + If 0 never qualify. Time in seconds. + + + + Timeout for qualify + + If the contact doesn't respond to the OPTIONS request before the timeout, + the contact is marked unavailable. + If 0 no timeout. Time in fractional seconds. + + + + Authenticates a qualify challenge response if needed + + If true and a qualify request receives a challenge response then + authentication is attempted before declaring the contact available. + + This option does nothing as we will always complete + the challenge response authentication if the qualify request is + challenged. + + + + + Outbound proxy used when sending OPTIONS request + + If set the provided URI will be used as the outbound proxy when an + OPTIONS request is sent to a contact for qualify purposes. + + + + Stored Path vector for use in Route headers on outgoing requests. + + + User-Agent header from registration. + + The User-Agent is automatically stored based on data present in incoming SIP + REGISTER requests and is not intended to be configured manually. + + + + Endpoint name + + The name of the endpoint this contact belongs to + + + + Asterisk Server name + + Asterisk Server name on which SIP endpoint registered. + + + + IP-address of the last Via header from registration. + + The last Via header should contain the address of UA which sent the request. + The IP-address of the last Via header is automatically stored based on data present + in incoming SIP REGISTER requests and is not intended to be configured manually. + + + + IP-port of the last Via header from registration. + + The IP-port of the last Via header is automatically stored based on data present + in incoming SIP REGISTER requests and is not intended to be configured manually. + + + + Call-ID header from registration. + + The Call-ID header is automatically stored based on data present + in incoming SIP REGISTER requests and is not intended to be configured manually. + + + + A contact that cannot survive a restart/boot. + + The option is set if the incoming SIP REGISTER contact is rewritten + on a reliable transport and is not intended to be configured manually. + + + + + The configuration for a location of an endpoint + + An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no + AoRs are specified, an endpoint will not be reachable by Asterisk. + Beyond that, an AoR has other uses within Asterisk, such as inbound + registration. + + An AoR is a way to allow dialing a group + of Contacts that all use the same + endpoint for calls. + + This can be used as another way of grouping a list of contacts to dial + rather than specifying them each directly when dialing via the dialplan. + This must be used in conjunction with the PJSIP_DIAL_CONTACTS. + + Registrations: For Asterisk to match an inbound registration to an endpoint, + the AoR object name must match the user portion of the SIP URI in the "To:" + header of the inbound SIP registration. That will usually be equivalent + to the "user name" set in your hard or soft phones configuration. + + + Permanent contacts assigned to AoR + + Contacts specified will be called whenever referenced + by chan_pjsip. + + Use a separate "contact=" entry for each contact required. Contacts + are specified using a SIP URI. + + + + Default expiration time in seconds for contacts that are dynamically bound to an AoR. + + + Allow subscriptions for the specified mailbox(es) + This option applies when an external entity subscribes to an AoR + for Message Waiting Indications. The mailboxes specified will be subscribed to. + More than one mailbox can be specified with a comma-delimited string. + app_voicemail mailboxes must be specified as mailbox@context; + for example: mailboxes=6001@default. For mailboxes provided by external sources, + such as through the res_mwi_external module, you must specify strings supported by + the external system. + + For endpoints that cannot SUBSCRIBE for MWI, you can set the mailboxes option in your + endpoint configuration section to enable unsolicited MWI NOTIFYs to the endpoint. + + + + The voicemail extension to send in the NOTIFY Message-Account header + + + Maximum time to keep an AoR + + Maximum time to keep a peer with explicit expiration. Time in seconds. + + + + Maximum number of contacts that can bind to an AoR + + Maximum number of contacts that can associate with this AoR. This value does + not affect the number of contacts that can be added with the "contact" option. + It only limits contacts added through external interaction, such as + registration. + + The rewrite_contact option + registers the source address as the contact address to help with + NAT and reusing connection oriented transports such as TCP and + TLS. Unfortunately, refreshing a registration may register a + different contact address and exceed + max_contacts. The + remove_existing and + remove_unavailable options can help by + removing either the soonest to expire or unavailable contact(s) over + max_contacts which is likely the + old rewrite_contact contact source + address being refreshed. + + This should be set to 1 and + remove_existing set to yes if you + wish to stick with the older chan_sip behaviour. + + + + + Minimum keep alive time for an AoR + + Minimum time to keep a peer with an explicit expiration. Time in seconds. + + + + Determines whether new contacts replace existing ones. + + On receiving a new registration to the AoR should it remove enough + existing contacts not added or updated by the registration to + satisfy max_contacts? Any removed + contacts will expire the soonest. + + The rewrite_contact option + registers the source address as the contact address to help with + NAT and reusing connection oriented transports such as TCP and + TLS. Unfortunately, refreshing a registration may register a + different contact address and exceed + max_contacts. The + remove_existing option can help by + removing the soonest to expire contact(s) over + max_contacts which is likely the + old rewrite_contact contact source + address being refreshed. + + This should be set to yes and + max_contacts set to 1 if you + wish to stick with the older chan_sip behaviour. + + + + + Determines whether new contacts should replace unavailable ones. + + The effect of this setting depends on the setting of + remove_existing. + If remove_existing is set to + no (default), setting remove_unavailable to + yes will remove only unavailable contacts that exceed + max_contacts to allow an incoming + REGISTER to complete sucessfully. + If remove_existing is set to + yes, setting remove_unavailable to + yes will prioritize unavailable contacts for removal + instead of just removing the contact that expires the soonest. + See remove_existing and + max_contacts for further information about how + these 3 settings interact. + + + + + Must be of type 'aor'. + + + Interval at which to qualify an AoR + + Interval between attempts to qualify the AoR for reachability. + If 0 never qualify. Time in seconds. + + + + Timeout for qualify + + If the contact doesn't respond to the OPTIONS request before the timeout, + the contact is marked unavailable. + If 0 no timeout. Time in fractional seconds. + + + + Authenticates a qualify challenge response if needed + + If true and a qualify request receives a challenge response then + authentication is attempted before declaring the contact available. + + This option does nothing as we will always complete + the challenge response authentication if the qualify request is + challenged. + + + + + Outbound proxy used when sending OPTIONS request + + If set the provided URI will be used as the outbound proxy when an + OPTIONS request is sent to a contact for qualify purposes. + + + + Enables Path support for REGISTER requests and Route support for other requests. + + When this option is enabled, the Path headers in register requests will be saved + and its contents will be used in Route headers for outbound out-of-dialog requests + and in Path headers for outbound 200 responses. Path support will also be indicated + in the Supported header. + + + + + Options that apply to the SIP stack as well as other system-wide settings + + The settings in this section are global. In addition to being global, the values will + not be re-evaluated when a reload is performed. This is because the values must be set + 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 + requests that receive no response when using an unreliable transport (e.g. UDP). + 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 + request before terminating the transaction. It is recommended that this be set + to 64 * Timer T1, but it may be set higher if desired. For more information on + this timer, see RFC 3261, Section 17.1.1.1. + + + + Use the short forms of common SIP header names. + + + Initial number of threads in the res_pjsip threadpool. + + + The amount by which the number of threads is incremented when necessary. + + + Number of seconds before an idle thread should be disposed of. + + + Maximum number of threads in the res_pjsip threadpool. + A value of 0 indicates no maximum. + + + Disable automatic switching from UDP to TCP transports. + + Disable automatic switching from UDP to TCP transports if outgoing + request is too large. See RFC 3261 section 18.1.1. + + + + Follow SDP forked media when To tag is different + + On outgoing calls, if the UAS responds with different SDP attributes + on subsequent 18X or 2XX responses (such as a port update) AND the + To tag on the subsequent response is different than that on the previous + one, follow it. + + + This option must also be enabled on endpoints that require + this functionality. + + + + + Follow SDP forked media when To tag is the same + + On outgoing calls, if the UAS responds with different SDP attributes + on non-100rel 18X or 2XX responses (such as a port update) AND the + To tag on the subsequent response is the same as that on the previous one, + process the updated SDP. + + + This option must also be enabled on endpoints that require + this functionality. + + + + + Disable the use of rport in outgoing requests. + + Remove "rport" parameter from the outgoing requests. + + + + Must be of type 'system' UNLESS the object name is 'system'. + + + + Options that apply globally to all SIP communications + + 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. + + + The interval (in seconds) to send keepalives to active connection-oriented transports. + + + The interval (in seconds) to check for expired contacts. + + + Disable Multi Domain support + + If disabled it can improve realtime performance by reducing the number of database requests. + + + + The maximum amount of time from startup that qualifies should be attempted on all contacts. + If greater than the qualify_frequency for an aor, qualify_frequency will be used instead. + + + The number of seconds over which to accumulate unidentified requests. + + If unidentified_request_count unidentified requests are received + during unidentified_request_period, a security event will be generated. + + + + The number of unidentified requests from a single IP to allow. + + If unidentified_request_count unidentified requests are received + during unidentified_request_period, a security event will be generated. + + + + The interval at which unidentified requests are older than + twice the unidentified_request_period are pruned. + + + Must be of type 'global' UNLESS the object name is 'global'. + + + Value used in User-Agent header for SIP requests and Server header for SIP responses. + + + When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given + peer who registers or unregisters with us. + + + Endpoint to use when sending an outbound request to a URI without a specified endpoint. + + + The voicemail extension to send in the NOTIFY Message-Account header if not specified on endpoint or aor + + + Enable/Disable SIP debug logging. Valid options include yes, no, or + a host address + + + The order by which endpoint identifiers are processed and checked. + Identifier names are usually derived from and can be found in the endpoint + identifier module itself (res_pjsip_endpoint_identifier_*). + You can use the CLI command "pjsip show identifiers" to see the + identifiers currently available. + + + One of the identifiers is "auth_username" which matches on the username in + an Authentication header. This method has some security considerations because an + Authentication header is not present on the first message of a dialog when + digest authentication is used. The client can't generate it until the server + sends the challenge in a 401 response. Since Asterisk normally sends a security + event when an incoming request can't be matched to an endpoint, using auth_username + requires that the security event be deferred until a request is received with + the Authentication header and only generated if the username doesn't result in a + match. This may result in a delay before an attack is recognized. You can control + how many unmatched requests are received from a single ip address before a security + event is generated using the unidentified_request parameters. + + + + + When Asterisk generates an outgoing SIP request, the From header username will be + set to this value if there is no better option (such as CallerID) to be + used. + + + When Asterisk generates a challenge, the digest realm will be + set to this value if there is no better option (such as auth/realm) to be + used. + + + MWI taskprocessor high water alert trigger level. + + On a heavily loaded system you may need to adjust the + taskprocessor queue limits. If any taskprocessor queue size + reaches its high water level then pjsip will stop processing + new requests until the alert is cleared. The alert clears + when all alerting taskprocessor queues have dropped to their + low water clear level. + + + + + MWI taskprocessor low water clear alert level. + + On a heavily loaded system you may need to adjust the + taskprocessor queue limits. If any taskprocessor queue size + reaches its high water level then pjsip will stop processing + new requests until the alert is cleared. The alert clears + when all alerting taskprocessor queues have dropped to their + low water clear level. + + Set to -1 for the low water level to be 90% of + the high water level. + + + + Enable/Disable sending unsolicited MWI to all endpoints on startup. + + When the initial unsolicited MWI notification are + enabled on startup then the initial notifications + get sent at startup. If you have a lot of endpoints + (thousands) that use unsolicited MWI then you may + want to consider disabling the initial startup + notifications. + + When the initial unsolicited MWI notifications are + disabled on startup then the notifications will start + on the endpoint's next contact update. + + + + + Enable/Disable ignoring SIP URI user field options. + + If you have this option enabled and there are semicolons + in the user field of a SIP URI then the field is truncated + at the first semicolon. This effectively makes the semicolon + a non-usable character for PJSIP endpoint names, extensions, + and AORs. This can be useful for improving compatibility with + an ITSP that likes to use user options for whatever reason. + + + sip:1235557890;phone-context=national@x.x.x.x;user=phone + + + 1235557890;phone-context=national + + + 1235557890 + + The caller-id and redirecting number strings + obtained from incoming SIP URI user fields are always truncated + at the first semicolon. + + + + Place caller-id information into Contact header + + This option will cause Asterisk to place caller-id information into + generated Contact headers. + + + + Enable sending AMI ContactStatus event when a device refreshes its registration. + + + Trigger scope for taskprocessor overloads + + This option specifies the trigger the distributor will use for + detecting taskprocessor overloads. When it detects an overload condition, + the distrubutor will stop accepting new requests until the overload is + cleared. + + + (default) Any taskprocessor overload will trigger. + Only pjsip taskprocessor overloads will trigger. + No overload detection will be performed. + + + The "none" and "pjsip_only" options should be used + with extreme caution and only to mitigate specific issues. + Under certain conditions they could make things worse. + + + + + Advertise support for RFC4488 REFER subscription suppression + + + + + diff --git a/res/res_pjsip/pjsip_manager.xml b/res/res_pjsip/pjsip_manager.xml new file mode 100644 index 0000000000..a0047aaf97 --- /dev/null +++ b/res/res_pjsip/pjsip_manager.xml @@ -0,0 +1,900 @@ + + + + + + Qualify a chan_pjsip endpoint. + + + + + The endpoint you want to qualify. + + + + Qualify a chan_pjsip endpoint. + + + + + Provide details about an identify section. + + + The object's type. This will always be 'identify'. + + + The name of this object. + + + + + + + + + + + + + + + The name of the endpoint associated with this information. + + + + + + + Provide details about an Address of Record (AoR) section. + + + The object's type. This will always be 'aor'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + The total number of contacts associated with this AoR. + + + The number of non-permanent contacts associated with this AoR. + + + The name of the endpoint associated with this information. + + + + + + + Provide details about an authentication section. + + + The object's type. This will always be 'auth'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + The name of the endpoint associated with this information. + + + + + + + Provide details about an authentication section. + + + The object's type. This will always be 'transport'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + The name of the endpoint associated with this information. + + + + + + + Provide details about an endpoint section. + + + The object's type. This will always be 'endpoint'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + The aggregate device state for this endpoint. + + + The number of active channels associated with this endpoint. + + + + + + + + + + + + + Provide details about an Address of Record (AoR) section. + + + The object's type. This will always be 'aor'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + Provide details about an Address of Record (Auth) section. + + + The object's type. This will always be 'auth'. + + + The name of this object. + + + + + + + + + + + + + + + + + + + + + + + + + Provide details about a contact section. + + + The object's type. This will always be 'contact'. + + + The name of this object. + + + IP address of the last Via header in REGISTER request. + Will only appear in the event if available. + + + Port number of the last Via header in REGISTER request. + Will only appear in the event if available. + + + The elapsed time in decimal seconds after which an OPTIONS + message is sent before the contact is considered unavailable. + + + Content of the Call-ID header in REGISTER request. + Will only appear in the event if available. + + + Asterisk Server name. + + + If true delete the contact on Asterisk restart/boot. + + + The Path header received on the REGISTER. + + + The name of the endpoint associated with this information. + + + A boolean indicating whether a qualify should be authenticated. + + + This contact's URI. + + + The interval in seconds at which the contact will be qualified. + + + Content of the User-Agent header in REGISTER request + + + Absolute time that this contact is no longer valid after + + + The contact's outbound proxy. + + + This contact's status. + + + + + + + + + The round trip time in microseconds. + + + + + + + Provide details about a contact's status. + + + The AoR that owns this contact. + + + This contact's URI. + + + This contact's status. + + + + + + + + + The round trip time in microseconds. + + + The name of the endpoint associated with this information. + + + Content of the User-Agent header in REGISTER request + + + Absolute time that this contact is no longer valid after + + + IP address:port of the last Via header in REGISTER request. + Will only appear in the event if available. + + + Content of the Call-ID header in REGISTER request. + Will only appear in the event if available. + + + The sorcery ID of the contact. + + + A boolean indicating whether a qualify should be authenticated. + + + The contact's outbound proxy. + + + The Path header received on the REGISTER. + + + The interval in seconds at which the contact will be qualified. + + + The elapsed time in decimal seconds after which an OPTIONS + message is sent before the contact is considered unavailable. + + + + + + + Provide details about a contact's status. + + + The object's type. This will always be 'endpoint'. + + + The name of this object. + + + The transport configurations associated with this endpoint. + + + The aor configurations associated with this endpoint. + + + The inbound authentication configurations associated with this endpoint. + + + The outbound authentication configurations associated with this endpoint. + + + The aggregate device state for this endpoint. + + + The number of active channels associated with this endpoint. + + + + + + + Lists PJSIP endpoints. + + + + + Provides a listing of all endpoints. For each endpoint an EndpointList event + is raised that contains relevant attributes and status information. Once all + endpoints have been listed an EndpointListComplete event is issued. + + + + + + + + + Provide final information about an endpoint list. + + + + + + + + + + + Detail listing of an endpoint and its objects. + + + + + The endpoint to list. + + + + + Provides a detailed listing of options for a given endpoint. Events are issued + showing the configuration and status of the endpoint and associated objects. These + events include EndpointDetail, AorDetail, + AuthDetail, TransportDetail, and + IdentifyDetail. Some events may be listed multiple times if multiple objects are + associated (for instance AoRs). Once all detail events have been raised a final + EndpointDetailComplete event is issued. + + + + + + + + + + + + + + Provide final information about endpoint details. + + + + + + + + + + + Lists PJSIP AORs. + + + + + Provides a listing of all AORs. For each AOR an AorList event + is raised that contains relevant attributes and status information. Once all + aors have been listed an AorListComplete event is issued. + + + + + + + + + Provide final information about an aor list. + + + + + + + + + + + Lists PJSIP Auths. + + + + Provides a listing of all Auths. For each Auth an AuthList event + is raised that contains relevant attributes and status information. Once all + auths have been listed an AuthListComplete event is issued. + + + + + + + + + Provide final information about an auth list. + + + + + + + + + + + Lists PJSIP Contacts. + + + + Provides a listing of all Contacts. For each Contact a ContactList + event is raised that contains relevant attributes and status information. + Once all contacts have been listed a ContactListComplete event + is issued. + + + + + + + + + Provide final information about a contact list. + + + + + + + + +