;
; SIP Configuration example for Asterisk
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several 
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
;        SIP/devicename/extension
;
;
; Devicename
;        devicename is defined as a peer in a section below.
;
; username@domain
;        Call any SIP user on the Internet
;        (Don't forget to enable DNS SRV records if you want to use this)
; 
; devicename/extension
;        If you define a SIP proxy as a peer below, you may call
;        SIP/proxyhostname/user or SIP/user@proxyhostname 
;        where the proxyhostname is defined in a section below 
;        This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
;        This form allows you to specify password or md5secret and authname
;        without altering any authentication data in config.
;        Examples:
;
;        SIP/*98@mysipproxy
;        SIP/sales:topsecret::account02@domain.com:5062
;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
;         SIP/sales@mysipproxy!sales@edvina.net
;
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
;   sip show peers               Show all SIP peers (including friends)
;   sip show registry            Show status of hosts we register with
;
;   sip set debug on             Show all SIP messages
;
;   module reload chan_sip.so    Reload configuration file
;
;------- Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
;	1. Asterisk checks the SIP From: address username and matches against
;	   names of devices with type=user 
;	   The name is the text between square brackets [name]
;	2. Asterisk checks the From: addres and matches the list of devices
;	   with a type=peer
;	3. Asterisk checks the IP address (and port number) that the INVITE
;	   was sent from and matches against any devices with type=peer
;
; Don't mix extensions with the names of the devices. Devices need a unique
; name. The device name is *not* used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
; 
; When setting up trunks, make sure there's no risk that any From: username
; (caller ID) will match any of your device names, because then Asterisk 
; might match the wrong device.
;
; Note: The parameter "username" is not the username and in most cases is
;       not needed at all. Check below. In later releases, it's renamed
;       to "defaultuser" which is a better name, since it is used in 
;       combination with the "defaultip" setting.
;-----------------------------------------------------------------------------

; ** Deprecated configuration options **
; The "call-limit" configuation option is deprecated. It still works in
; this version of Asterisk, but will disappear in the next version.
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
;
; You can still set limits per device in sip.conf or in a database by using 
; "setvar" to set variables that can be used in the dialplan for various limits.

[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
;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)
;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
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)

;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental.  Since it is new, all of the related configuration options are
; subject to change in any release.  If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
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

;tlscertfile=asterisk.pem       ; Certificate file (*.pem only) to use for TLS connections 
                                ; default is to look for "asterisk.pem" in current directory

;tlscafile=</path/to/certificate>
;        If the server your connecting to uses a self signed certificate
;        you should have their certificate installed here so the code can 
;        verify the authenticity of their certificate.

;tlscadir=</path/to/ca/dir>
;        A directory full of CA certificates.  The files must be named with 
;        the CA subject name hash value. 
;        (see man SSL_CTX_load_verify_locations for more info) 

;tlsdontverifyserver=[yes|no]
;        If set to yes, don't verify the servers certificate when acting as 
;        a client.  If you don't have the server's CA certificate you can
;        set this and it will connect without requiring tlscafile to be set.
;        Default is no.

;tlscipher=<SSL cipher string>
;        A string specifying which SSL ciphers to use or not use
;        A list of valid SSL cipher strings can be found at: 
;                http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS

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 "no")

; See qos.tex or Quality of Service section of asterisk.pdf 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
;qualifyfreq=60                 ; Qualification: How often to check for the 
                                ; host to be up in seconds
                                ; Set to low value if you use low timeout for
                                ; NAT of UDP sessions
;qualifygap=100			; Number of milliseconds between each group of peers being qualified
;qualifypeers=1			; Number of peers in a group to be qualified at the same time
;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.
;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.
;
;disallow=all                   ; First disallow all codecs
;allow=ulaw                     ; Allow codecs in order of preference
;allow=ilbc                     ; see doc/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
;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.

;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=Asterisk 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.
;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
;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.

;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 :-(
;outboundproxy=proxy.provider.domain            ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls 
;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
;                                               ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
                                ; your localnet setting. Unless you have some sort of strange network
                                ; setup you will not need to enable this.

;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.
;forwardloopdetected=no         ; Attempt to forward a call locally if the
                                ; destination replies with 482 Loop Detected
                                ; default = yes

; 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

;
; 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
;
;--------------------------- 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=<secs>            ; 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.

;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
;
; t38pt_udptl = yes            ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,fec        ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
; t38pt_udptl = yes,none       ; Enables T.38 with no error correction.
;
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
;                                       ; the other endpoint's provided value to assume we can
;                                       ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
; based one or more events being detected. The events that can be detected are an incoming
; CNG tone or an incoming T.38 re-INVITE request.
;
; faxdetect = yes		; Default 'no', 'yes' enables both CNG and T.38 detection
; faxdetect = cng		; Enables only CNG detection
; faxdetect = t38		; Enables only T.38 detection
; faxdetect = both		; Enables both CNG and T.38 detection (same as 'yes')
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
; 
;
; domain is either 
;	- domain in DNS
; 	- host name in DNS
;	- the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to 
; is username@domain
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
;        register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Note that a register= line doesn't mean that we will match the incoming call in any
; other way than described above. If you want to control where the call enters your
; dialplan, which context, you want to define a peer with the hostname of the provider's
; server. If the provider has multiple servers to place calls to your system, you need
; a peer for each server.
;
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
; contain a port number. Since the logical separator between a host and port number is a
; ':' character, and this character is already used to separate between the optional "secret"
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
; they are blank. See the third example below for an illustration.
;
;
; Examples:
;
;register => 1234:password@mysipprovider.com        
;
;     This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
;    Register 2345 at sip provider 'sip_proxy'.  Calls from this provider
;    connect to local extension 1234 in extensions.conf, default context,
;    unless you configure a [sip_proxy] section below, and configure a
;    context.
;    Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;    Tip 2: Use separate inbound and outbound sections for SIP providers
;           (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
;    Note that in this example, the optional authuser and secret portions have
;    been left blank because we have specified a port in the user section
;
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
;
;    The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
;    Using 'udp://' explicitly is also useful in case the username part
;    contains a '/' ('user/name').

;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
                                ; 0 = continue forever, hammering the other server
                                ; until it accepts the registration
                                ; Default is 0 tries, continue forever
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones.
; Format for the mwi register statement is:
;       mwi => user[:secret[:authuser]]@host[:port][/mailbox]
;
; Examples:
;mwi => 1234:password@mysipprovider.com/1234
;
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
; mailbox=1234@SIP_Remote
;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the "local" address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone "inside" or "outside" of the NATted network.
;   This is configured by assigning the "localnet" parameter with a list
;   of network addresses that are considered "inside" of the NATted network.
;   IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
;   Multiple entries are allowed, e.g. a reasonable set is the following:
;
;      localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
;      localnet=10.0.0.0/255.0.0.0      ; Also RFC1918
;      localnet=172.16.0.0/12           ; Another RFC1918 with CIDR notation
;      localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
;
; + the "externally visible" address and port number to be used when talking
;   to a host outside the NAT. This information is derived by one of the
;   following (mutually exclusive) config file parameters:
;
;   a. "externip = hostname[:port]" specifies a static address[:port] to
;      be used in SIP and SDP messages.
;      The hostname is looked up only once, when [re]loading sip.conf .
;      If a port number is not present, use the "bindport" value (which is
;      not guaranteed to work correctly, because a NAT box might remap the
;      port number as well as the address).
;      This approach can be useful if you have a NAT device where you can
;      configure the mapping statically. Examples:
;
;        externip = 12.34.56.78          ; use this address.
;        externip = 12.34.56.78:9900     ; use this address and port.
;        externip = mynat.my.org:12600   ; Public address of my nat box.
;
;   b. "externhost = hostname[:port]" is similar to "externip" except
;      that the hostname is looked up every "externrefresh" seconds
;      (default 10s). This can be useful when your NAT device lets you choose
;      the port mapping, but the IP address is dynamic.
;      Beware, you might suffer from service disruption when the name server
;      resolution fails. Examples:
;
;        externhost=foo.dyndns.net       ; refreshed periodically
;        externrefresh=180               ; change the refresh interval
;
;   c. "stunaddr = stun.server[:port]" queries the STUN server specified
;      as an argument to obtain the external address/port.
;      Queries are also sent periodically every "externrefresh" seconds
;      (as a side effect, sending the query also acts as a keepalive for
;      the state entry on the nat box):
;
;        stunaddr = foo.stun.com:3478
;        externrefresh = 15
;
;   Note that at the moment all these mechanism work only for the SIP socket.
;   The IP address discovered with externip/externhost/STUN is reused for
;   media sessions as well, but the port numbers are not remapped so you
;   may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externip" and
; "externhost" might not help you configure addresses properly, and you
; really need to use STUN.
;
; NOTE 2: when using "externip" or "externhost", the address part is
; also used as the external address for media sessions. Even if you 
; use "stunaddr", STUN queries will be sent only from the SIP port,
; not from media sockets. Thus, the port information in the SDP may be wrong!
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ' settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
;        nat = no                ; default. Use NAT mode only according to RFC3581 (;rport)
;        nat = yes               ; Always ignore info and assume NAT
;        nat = never             ; Never attempt NAT mode or RFC3581 support
;        nat = route             ; route = Assume NAT, don't send rport 
;                                ; (work around more UNIDEN bugs)

;----------------------------------- 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).

;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.

                                ; 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'.

;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.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;
;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|<seconds>)
                                ; 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=<domain>[,<context>]
; 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. 

;------------------------------ 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".
;-----------------------------------------------------------------------------------

[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 = <user>:<secret>@<realm>
;        auth = <user>#<md5secret>@<realm>
; 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

;------------------------------------------------------------------------------
; DEVICE CONFIGURATION
; 
; The SIP channel has two types of devices, the friend and the peer.
; * The type=friend is a device type that accepts both incoming and outbound calls,
;   where Asterisk match on the From: username on incoming calls.
;   (A synonym for friend is "user"). This is a type you use for your local
;   SIP phones.
; * The type=peer also handles both incoming and outbound calls. On inbound calls,
;   Asterisk only matches on IP/port, not on names. This is mostly used for SIP
;   trunks.
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
; 
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you probably have NAT problems. 
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
; 
; Configuration options available 
; --------------------     
; context
; callingpres
; permit
; deny
; secret
; md5secret
; remotesecret
; transport
; dtmfmode
; directmedia
; nat
; callgroup
; pickupgroup
; language
; allow
; disallow
; insecure
; trustrpid
; progressinband
; promiscredir
; useclientcode
; accountcode
; setvar
; callerid
; amaflags
; callcounter
; busylevel
; allowoverlap
; allowsubscribe
; allowtransfer
; ignoresdpversion
; subscribecontext
; template
; videosupport
; maxcallbitrate
; rfc2833compensate
; mailbox
; session-timers
; session-expires
; session-minse
; session-refresher
; t38pt_usertpsource
; regexten
; fromdomain
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; registertrying
; timert1
; timerb
; qualifyfreq
; t38pt_usertpsource
; contactpermit         ; Limit what a host may register as (a neat trick
; contactdeny           ; is to register at the same IP as a SIP provider,
;                       ; then call oneself, and get redirected to that
;                       ; same location).

;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls 
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com

;[sip_proxy-out]
;type=peer                        ; we only want to call out, not be called
;remotesecret=guessit             ; Our password to their service
;defaultuser=yourusername         ; Authentication user for outbound proxies
;fromuser=yourusername            ; Many SIP providers require this!
;fromdomain=provider.sip.domain 
;host=box.provider.com
;transport=udp,tcp                ; This sets the default transport type to udp for outgoing, and will
;                                 ; accept both tcp and udp. The default transport type is only used for
;                                 ; outbound messages until a Registration takes place.  During the
;                                 ; peer Registration the transport type may change to another supported
;                                 ; type if the peer requests so.

;usereqphone=yes                  ; This provider requires ";user=phone" on URI
;callcounter=yes                  ; Enable call counter
;busylevel=2                      ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain  ; send outbound signaling to this proxy, not directly to the peer
;port=80                          ; The port number we want to connect to on the remote side
                                  ; Also used as "defaultport" in combination with "defaultip" settings

;--- sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
;fromuser=4015552299              ; how your provider knows you
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
;secret=gissadetdu                ; The password they use to contact us
;callbackextension=123            ; Register with this server and require calls coming back to this extension
;transport=udp,tcp                ; This sets the transport type to udp for outgoing, and will
;                                 ;   accept both tcp and udp. Default is udp. The first transport
;                                 ;   listed will always be used for outgoing connections.

;
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        nat=yes
        directmedia=no
        host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
        nat=no
        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

; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
;        secret = peekaboo
; [2134](natted-phone,ulaw-phone)
;        secret = not_very_secret
; [2136](public-phone,ulaw-phone)
;        secret = not_very_secret_either
; ...
;

; Standard configurations not using templates look like this:
;
;[grandstream1]
;type=friend                         
;context=from-sip                ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234>        ; Full caller ID, to override the phones config
                                 ; on incoming calls to Asterisk
;host=192.168.0.23               ; we have a static but private IP address
                                 ; No registration allowed
;nat=no                          ; there is not NAT between phone and Asterisk
;directmedia=yes                 ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info                   ; either RFC2833 or INFO for the BudgeTone
;call-limit=1                    ; permit only 1 outgoing call and 1 incoming call at a time
                                 ; from the phone to asterisk (deprecated)
                                 ; 1 for the explicit peer, 1 for the explicit user,
                                 ; remember that a friend equals 1 peer and 1 user in
                                 ; memory
                                 ; There is no combined call counter for a "friend"
                                 ; so there's currently no way in sip.conf to limit
                                 ; to one inbound or outbound call per phone. Use
                                 ; the group counters in the dial plan for that.
                                 ;
;mailbox=1234@default            ; mailbox 1234 in voicemail context "default"
;disallow=all                    ; need to disallow=all before we can use allow=
;allow=ulaw                      ; Note: In user sections the order of codecs
                                 ; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
;allow=g729                      ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
                                 ; See README.callingpres for more information

;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234                   ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic                    ; This device needs to register
;nat=yes                         ; X-Lite is behind a NAT router
;directmedia=no                  ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm                       ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes              ; Send a 100 Trying when the device registers.

;[snom]
;type=friend                     ; Friends place calls and receive calls
;context=from-sip                ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de                     ; Use German prompts for this user 
;host=dynamic                    ; This peer register with us
;dtmfmode=inband                 ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59          ; IP used until peer registers
;mailbox=1234@context,2345       ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes                ; Only send notifications if this phone 
                                 ; subscribes for mailbox notification
;vmexten=voicemail               ; dialplan extension to reach mailbox 
                                 ; sets the Message-Account in the MWI notify message
                                 ; defaults to global vmexten which defaults to "asterisk"
;disallow=all
;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!


;[polycom]
;type=friend                     ; Friends place calls and receive calls
;context=from-sip                ; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic                    ; This peer register with us
;dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info
;defaultuser=polly               ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
                                 ; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw                      ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no               ; Polycom phones don't work properly with "never"


;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;insecure=port                   ; Allow matching of peer by IP address without 
                                 ; matching port number
;insecure=invite                 ; Do not require authentication of incoming INVITEs
;insecure=port,invite            ; (both)
;qualify=1000                    ; Consider it down if it's 1 second to reply
                                 ; Helps with NAT session
                                 ; qualify=yes uses default value
;qualifyfreq=60                  ; Qualification: How often to check for the 
                                 ; host to be up in seconds
                                 ; Set to low value if you use low timeout for
                                 ; NAT of UDP sessions
;
; Call group and Pickup group should be in the range from 0 to 63
;
;callgroup=1,3-4                 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5               ; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60          ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0            ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0
;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks

;[cisco1]
;type=friend
;secret=blah
;qualify=200                     ; Qualify peer is no more than 200ms away
;nat=yes                         ; This phone may be natted
                                 ; Send SIP and RTP to the IP address that packet is 
                                 ; received from instead of trusting SIP headers 
;host=dynamic                    ; This device registers with us
;directmedia=no                  ; Asterisk by default tries to redirect the
                                 ; RTP media stream (audio) to go directly from
                                 ; the caller to the callee.  Some devices do not
                                 ; support this (especially if one of them is 
                                 ; behind a NAT).
;defaultip=192.168.0.4           ; IP address to use until registration
;defaultuser=goran               ; Username to use when calling this device before registration
                                 ; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678              ; Channel variable to be set for all calls from or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
                                                ; cause the given audio file to
                                                ; be played upon completion of
                                                ; an attended transfer.

;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes          ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
                                ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
                                ; external IP address of the remote device. If port forwarding is done at the client side
                                ; then UDPTL will flow to the remote device.
