; SIP Configuration for Asterisk [general] context=default ; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes) ; If your Asterisk is connected to the Internet ; and you have allowguest=yes ; you want to check which services you offer everyone ; out there, by enabling them in the default context (see below). ;match_auth_username=yes ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowoverlap=yes ; Enable RFC3578 overlap dialing support. ; Can use the Incomplete application to collect the ; needed digits from an ambiguous dialplan match. ;allowoverlap=dtmf ; Enable overlap dialing support using DTMF delivery ; methods (inband, RFC2833, SIP INFO) in the early ; media phase. Uses the Incomplete application to ; collect the needed digits. ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled. The Dial() options 't' and 'T' are not ; related as to whether SIP transfers are allowed or not. ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk". If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name ;domainsasrealm=no ; Use domains list as realms ; You can serve multiple Realms specifying several ; 'domain=...' directives (see below). ; In this case Realm will be based on request 'From'/'To' header ; and should match one of domain names. ; Otherwise default 'realm=...' will be used. udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no) ;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) ; Remember that the IP address must match the common name (hostname) in the ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. ; For details how to construct a certificate for SIP see ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs ;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number ; of seconds a client has to authenticate. If ; the client does not authenticate beofre this ; timeout expires, the client will be ; disconnected. (default: 30 seconds) ;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of ; unauthenticated sessions that will be allowed ; to connect at any given time. (default: 100) transport=udp ; Set the default transports. The order determines the primary default transport. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP. srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. ;pedantic=yes ; Enable checking of tags in headers, ; international character conversions in URIs ; and multiline formatted headers for strict ; SIP compatibility (defaults to "yes") ; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets. ;cos_video=4 ; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets. ;maxexpiry=3600 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) ;defaultexpiry=120 ; Default length of incoming/outgoing registration ;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions ;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention) ; Default value is 70 ;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds ; and reported in milliseconds with sip show settings. ; Set to low value if you use low timeout for NAT of UDP sessions ; Default: 60 ;qualifygap=100 ; Number of milliseconds between each group of peers being qualified ; Default: 100 ;qualifypeers=1 ; Number of peers in a group to be qualified at the same time ; Default: 1 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC ; fully. Enable this option to not get error messages ; when sending MWI to phones with this bug. ;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in ; the From: header as the "name" portion. Also fill the ; "user" portion of the URI in the From: header with this ; value if no fromuser is set ; Default: empty ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to "asterisk" ; Codec negotiation ; ; When Asterisk is receiving a call, the codec will initially be set to the ; first codec in the allowed codecs defined for the user receiving the call ; that the caller also indicates that it supports. But, after the caller ; starts sending RTP, Asterisk will switch to using whatever codec the caller ; is sending. ; ; When Asterisk is placing a call, the codec used will be the first codec in ; the allowed codecs that the callee indicates that it supports. Asterisk will ; *not* switch to whatever codec the callee is sending. ; preferred_codec_only=yes ; 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. ;disallow=all ; First disallow all codecs allow=alaw allow=ulaw ; Allow codecs in order of preference allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization ; for framing options ; ; This option specifies a preference for which music on hold class this channel ; should listen to when put on hold if the music class has not been set on the ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer ; channel putting this one on hold did not suggest a music class. ; ; This option may be specified globally, or on a per-user or per-peer basis. ; ;mohinterpret=default ; ; This option specifies which music on hold class to suggest to the peer channel ; when this channel places the peer on hold. It may be specified globally or on ; a per-user or per-peer basis. ; ;mohsuggest=default ; ;parkinglot=plaza ; Sets the default parking lot for call parking ; This may also be set for individual users/peers ; Parkinglots are configured in features.conf ;language=en ; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;trustrpid = no ; If Remote-Party-ID should be trusted ;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) ;sendrpid = rpid ; Use the "Remote-Party-ID" header ; to send the identity of the remote party ; This is identical to sendrpid=yes ;sendrpid = pai ; Use the "P-Asserted-Identity" header ; to send the identity of the remote party ;rpid_update = no ; In certain cases, the only method by which a connected line ; change may be immediately transmitted is with a SIP UPDATE request. ; If communicating with another Asterisk server, and you wish to be able ; transmit such UPDATE messages to it, then you must enable this option. ; Otherwise, we will have to wait until we can send a reinvite to ; transmit the information. ;prematuremedia=no ; Some ISDN links send empty media frames before ; the call is in ringing or progress state. The SIP ; channel will then send 183 indicating early media ; which will be empty - thus users get no ring signal. ; Setting this to "yes" will stop any media before we have ; call progress (meaning the SIP channel will not send 183 Session ; Progress for early media). Default is "yes". Also make sure that ; the SIP peer is configured with progressinband=never. ; ; In order for "noanswer" applications to work, you need to run ; the progress() application in the priority before the app. ;progressinband=never ; If we should generate in-band ringing always ; use 'never' to never use in-band signalling, even in cases ; where some buggy devices might not render it ; Valid values: yes, no, never Default: never useragent=N39 Door PBX ; Allows you to change the user agent string ; The default user agent string also contains the Asterisk ; version. If you don't want to expose this, change the ; useragent string. ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Note that promiscredir when redirects are made to the ; local system will cause loops since Asterisk is incapable ; of performing a "hairpin" call. ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains ; a valid phone number ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 ; Other options: ; info : SIP INFO messages (application/dtmf-relay) ; shortinfo : SIP INFO messages (application/dtmf) ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) ; auto : Use rfc2833 if offered, inband otherwise ;compactheaders = yes ; send compact sip headers. ; ;videosupport=yes ; Turn on support for SIP video. You need to turn this ; on in this section to get any video support at all. ; You can turn it off on a per peer basis if the general ; video support is enabled, but you can't enable it for ; one peer only without enabling in the general section. ; If you set videosupport to "always", then RTP ports will ; always be set up for video, even on clients that don't ; support it. This assists callfile-derived calls and ; certain transferred calls to use always use video when ; available. [yes|NO|always] ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) ; Videosupport and maxcallbitrate is settable ; for peers and users as well ;callevents=no ; generate manager events when sip ua ; performs events (e.g. hold) ;authfailureevents=no ; generate manager "peerstatus" events when peer can't ; authenticate with Asterisk. Peerstatus will be "rejected". alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ; for any reason, always reject with an identical response ; equivalent to valid username and invalid password/hash ; instead of letting the requester know whether there was ; a matching user or peer for their request. This reduces ; the ability of an attacker to scan for valid SIP usernames. ; This option is set to "yes" by default. ;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like ; INVITE requests are. By default this option is disabled. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing ; order instead of RFC3551 packing order (this is required ; for Sipura and Grandstream ATAs, among others). This is ; contrary to the RFC3551 specification, the peer _should_ ; be negotiating AAL2-G726-32 instead :-( ;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering ; as any IP address used for staticly defined ; hosts. This helps avoid the configuration ; error of allowing your users to register at ; the same address as a SIP provider. ;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to ;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may ; register their phones. contactpermit=172.23.48.0/255.255.252.0 ;engine=asterisk ; RTP engine to use when communicating with the device ; ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with ; us and have a "regexten=" configuration item. ; Multiple contexts may be specified by separating them with '&'. The ; actual extension is the 'regexten' parameter of the registering peer or its ; name if 'regexten' is not provided. If more than one context is provided, ; the context must be specified within regexten by appending the desired ; context after '@'. More than one regexten may be supplied if they are ; separated by '&'. Patterns may be used in regexten. ; ;regcontext=sipregistrations ;regextenonqualify=yes ; Default "no" ; If you have qualify on and the peer becomes unreachable ; this setting will enforce inactivation of the regexten ; extension for the peer ;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons ; in the user field of a sip URI, the field be truncated ; at the first semicolon seen. This effectively makes ; semicolon a non-usable character for peer names, extensions, ; and maybe other, less tested things. This can be useful ; for improving compatability with devices that like to use ; user options for whatever reason. The behavior is similar to ; how SIP URI's were typically handled in 1.6.2, hence the name. ; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not ; in square brackets. For example, the caller id value 555.5555 becomes 5555555 ; when this option is enabled. Disabling this option results in no modification ; of the caller id value, which is necessary when the caller id represents something ; that must be preserved. This option can only be used in the [general] section. ; By default this option is on. ; ;shrinkcallerid=yes ; on by default ;use_q850_reason = no ; Default "no" ; Set to yes add Reason header and use Reason header if it is available. ;--------------------------- SIP timers ---------------------------------------------------- ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500 ; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 ;--------------------------- RTP timers ---------------------------------------------------- ; These timers are currently used for both audio and video streams. The RTP timeouts ; are only applied to the audio channel. ; The settings are settable in the global section as well as per device ; ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity ; on the audio channel ; when we're on hold (must be > rtptimeout) ;rtpkeepalive= ; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) ;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------ ; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions. ; This mechanism can detect and reclaim SIP channels that do not terminate through normal ; signaling procedures. Session-Timers can be configured globally or at a user/peer level. ; The operation of Session-Timers is driven by the following configuration parameters: ; ; * session-timers - Session-Timers feature operates in the following three modes: ; originate : Request and run session-timers always ; accept : Run session-timers only when requested by other UA ; refuse : Do not run session timers in any case ; The default mode of operation is 'accept'. ; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs. ; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs. ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'. ; ;session-timers=originate ;session-expires=600 ;session-minse=90 ;session-refresher=uas ; ;--------------------------- SIP DEBUGGING --------------------------------------------------- ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;dumphistory=yes ; Dump SIP history at end of SIP dialogue ; SIP history is output to the DEBUG logging channel ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- ; You can subscribe to the status of extensions with a "hint" priority ; (See extensions.conf.sample for examples) ; chan_sip support two major formats for notifications: dialog-info and SIMPLE ; ; You will get more detailed reports (busy etc) if you have a call counter enabled ; for a device. ; ; If you set the busylevel, we will indicate busy when we have a number of calls that ; matches the busylevel treshold. ; ; For queues, you will need this level of detail in status reporting, regardless ; if you use SIP subscriptions. Queues and manager use the same internal interface ; for reading status information. ; ; Note: Subscriptions does not work if you have a realtime dialplan and use the ; realtime switch. ; ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests ; Useful to limit subscriptions to local extensions ; Settable per peer/user also ;notifyringing = no ; Control whether subscriptions already INUSE get sent ; RINGING when another call is sent (default: yes) ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no) ; Turning on notifyringing and notifyhold will add a lot ; more database transactions if you are using realtime. ;notifycid = yes ; Control whether caller ID information is sent along with ; dialog-info+xml notifications (supported by snom phones). ; Note that this feature will only work properly when the ; incoming call is using the same extension and context that ; is being used as the hint for the called extension. This means ; that it won't work when using subscribecontext for your sip ; user or peer (if subscribecontext is different than context). ; This is also limited to a single caller, meaning that if an ; extension is ringing because multiple calls are incoming, ; only one will be used as the source of caller ID. Specify ; 'ignore-context' to ignore the called context when looking ; for the caller's channel. The default value is 'no.' Setting ; notifycid to 'ignore-context' also causes call-pickups attempted ; via SNOM's NOTIFY mechanism to set the context for the call pickup ; to PICKUPMARK. ;callcounter = yes ; Enable call counters on devices. This can be set per ; device too. ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ ; Asterisk can register as a SIP user agent to a SIP proxy (provider) register => {{ gatekeeper_sip_registration }}/s ; ; This will pass incoming calls to the 's' extension ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's ; no reason for Asterisk to stay in the media path, the media will be redirected. ; This does not really work well in the case where Asterisk is outside and the ; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. ; ;directmedia=yes ; Asterisk by default tries to redirect the ; RTP media stream to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients ; behind a NAT, or for some other reason want Asterisk to ; stay in the audio path, you may want to turn this off. ; This setting also affect direct RTP ; at call setup (a new feature in 1.4 - setting up the ; call directly between the endpoints instead of sending ; a re-INVITE). ; Additionally this option does not disable all reINVITE operations. ; It only controls Asterisk generating reINVITEs for the specific ; purpose of setting up a direct media path. If a reINVITE is ; needed to switch a media stream to inactive (when placed on ; hold) or to T.38, it will still be done, regardless of this ; setting. Note that direct T.38 is not supported. ;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). ;directmedia=update ; Yet a third option... use UPDATE for media path redirection, ; instead of INVITE. This can be combined with 'nonat', as ; 'directmedia=update,nonat'. It implies 'yes'. ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the ; callers INVITE. This will also fail if directmedia is enabled when ; the device is actually behind NAT. ;directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict ;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other ; (There is no default setting, this is just an example) ; Use this if some of your phones are on IP addresses that ; can not reach each other directly. This way you can force ; RTP to always flow through asterisk in such cases. ;ignoresdpversion=yes ; By default, Asterisk will honor the session version ; number in SDP packets and will only modify the SDP ; session if the version number changes. This option will ; force asterisk to ignore the SDP session version number ; and treat all SDP data as new data. This is required ; for devices that send us non standard SDP packets ; (observed with Microsoft OCS). By default this option is ; off. ;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=) ; Like the useragent parameter, the default user agent string ; also contains the Asterisk version. ;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=) ; This field MUST NOT contain spaces ;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media) ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if ; the peer does not support SRTP. Defaults to no. ;----------------------------------------- REALTIME SUPPORT ------------------------ ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ; ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ;rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ;ignoreregexpire=yes ; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' ; domains, each of which can direct the call to a specific context if desired. ; By default, all domains are accepted and sent to the default context or the ; context associated with the user/peer placing the call. ; REGISTER to non-local domains will be automatically denied if a domain ; list is configured. ; ; Domains can be specified using: ; domain=[,] ; Examples: ; domain=myasterisk.dom ; domain=customer.com,customer-context ; ; In addition, all the 'default' domains associated with a server should be ; added if incoming request filtering is desired. ; autodomain=yes ; ; To disallow requests for domains not serviced by this server: ; allowexternaldomains=no ;domain=mydomain.tld,mydomain-incoming ; Add domain and configure incoming context ; for external calls to this domain ;domain=1.2.3.4 ; Add IP address as local domain ; You can have several "domain" settings ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to ; non-peers, use your primary domain "identity" ; for From: headers instead of just your IP ; address. This is to be polite and ; it may be a mandatory requirement for some ; destinations which do not have a prior ; account relationship with your server. ;------------------------------ Advice of Charge CONFIGURATION -------------------------- ; snom_aoc_enabled = yes; ; This options turns on and off support for sending AOC-D and ; AOC-E to snom endpoints. This option can be used both in the ; peer and global scope. The default for this option is off. ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to "no". ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. ; The option represents the number of milliseconds by which the new jitter buffer ; will pad its size. the default is 40, so without modification, the new ; jitter buffer will set its size to the jitter value plus 40 milliseconds. ; increasing this value may help if your network normally has low jitter, ; but occasionally has spikes. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------- SIP_CAUSE reporting --------------------------------- ; storesipcause = no ; This option causes chan_sip to set the ; HASH(SIP_CAUSE,) channel variable ; to the value of the last sip response. ; WARNING: enabling this option carries a ; significant performance burden. It should only ; be used in low call volume situations. This ; option defaults to "no". ;----------------------------------------------------------------------------------- [authentication] ; Global credentials for outbound calls, i.e. when a proxy challenges your ; Asterisk server for authentication. These credentials override ; any credentials in peer/register definition if realm is matched. ; ; This way, Asterisk can authenticate for outbound calls to other ; realms. We match realm on the proxy challenge and pick an set of ; credentials from this list ; Syntax: ; auth = :@ ; auth = #@ ; Example: ;auth=mark:topsecret@digium.com ; ; You may also add auth= statements to [peer] definitions ; Peer auth= override all other authentication settings if we match on realm [basic-options](!) ; a template dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options directmedia=yes [my-codecs](!) ; a template for my preferred codecs disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw [ulaw-phone](!) ; and another one for ulaw-only disallow=all allow=ulaw