;

; 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::[email protected]:5062
;
SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:[email protected]
;
; 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@[email protected]
;
; 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:[email protected]
;
; 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:[email protected]
;
; 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:[email protected]/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:[email protected]
;
; 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.

context=users

[6001] ;Tom Smith

type=friend
host=dynamic
secret=xlite
context=users
mailbox=6001@default

[6002] ;Joe Jones

type=friend
host=dynamic
secret=xlite
context=users
mailbox=6002@default

[6003] ;Julio Jaramillo

type=friend
host=dynamic
secret=xlite
context=users
mailbox=6003@default

[4002]
type=friend
host=dynamic
secret=xlite
context=users

[4003]
type=friend
host=dynamic
secret=xlite
context=users

register=>bootcamp1:[email protected]

[to_sipprovider]
type=peer
context=users
username=bootcamp1
fromuser=bootcamp1
fromdomain=bootcamp.com
secret=bootcamp
canreinvite=no
insecure=invite,port
host=192.168.1.1
deny=0.0.0.0/0
permit=1.1.1.1/255.255.255.255
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
nat=no