2 ; SIP Configuration example for Asterisk
4 ; Syntax for specifying a SIP device in extensions.conf is
5 ; SIP/devicename where devicename is defined in a section below.
8 ; SIP/username@domain to call any SIP user on the Internet
9 ; (Don't forget to enable DNS SRV records if you want to use this)
11 ; If you define a SIP proxy as a peer below, you may call
12 ; SIP/proxyhostname/user or SIP/user@proxyhostname
13 ; where the proxyhostname is defined in a section below
15 ; Useful CLI commands to check peers/users:
16 ; sip list peers Show all SIP peers (including friends)
17 ; sip list users Show all SIP users (including friends)
18 ; sip list registry Show status of hosts we register with
20 ; sip debug Show all SIP messages
22 ; sip reload Reload configuration file
23 ; Active SIP peers will not be reconfigured
27 context=default ; Default context for incoming calls
28 ;allowguest=no ; Allow or reject guest calls (default is yes)
29 ;match_auth_username=yes ; if available, match user entry using the
30 ; 'username' field from the authentication line
31 ; instead of the From: field.
33 allowoverlap=no ; Disable overlap dialing support. (Default is yes)
34 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
36 ;realm=mydomain.tld ; Realm for digest authentication
37 ; defaults to "asterisk". If you set a system name in
38 ; asterisk.conf, it defaults to that system name
39 ; Realms MUST be globally unique according to RFC 3261
40 ; Set this to your host name or domain name
41 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
42 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
43 srvlookup=yes ; Enable DNS SRV lookups on outbound calls
44 ; Note: Asterisk only uses the first host
46 ; Disabling DNS SRV lookups disables the
47 ; ability to place SIP calls based on domain
48 ; names to some other SIP users on the Internet
50 ;domain=mydomain.tld ; Set default domain for this host
51 ; If configured, Asterisk will only allow
52 ; INVITE and REFER to non-local domains
53 ; Use "sip show domains" to list local domains
54 ;pedantic=yes ; Enable checking of tags in headers,
55 ; international character conversions in URIs
56 ; and multiline formatted headers for strict
57 ; SIP compatibility (defaults to "no")
59 ; See doc/README.tos for a description of these parameters.
60 ;tos_sip=cs3 ; Sets TOS for SIP packets.
61 ;tos_audio=ef ; Sets TOS for RTP audio packets.
62 ;tos_video=af41 ; Sets TOS for RTP video packets.
64 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
65 ; and subscriptions (seconds)
66 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
67 ;defaultexpiry=120 ; Default length of incoming/outgoing registration
68 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
70 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
71 ;checkmwi=10 ; Default time between mailbox checks for peers
72 ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
73 ; Message-Account in the MWI notify message
74 ; defaults to "asterisk"
75 ;disallow=all ; First disallow all codecs
76 ;allow=ulaw ; Allow codecs in order of preference
77 ;allow=ilbc ; see doc/rtp-packetization for framing options
79 ; This option specifies a preference for which music on hold class this channel
80 ; should listen to when put on hold if the music class has not been set on the
81 ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
82 ; channel putting this one on hold did not suggest a music class.
84 ; This option may be specified globally, or on a per-user or per-peer basis.
88 ; This option specifies which music on hold class to suggest to the peer channel
89 ; when this channel places the peer on hold. It may be specified globally or on
90 ; a per-user or per-peer basis.
94 ;language=en ; Default language setting for all users/peers
95 ; This may also be set for individual users/peers
96 ;relaxdtmf=yes ; Relax dtmf handling
97 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
98 ; when we're not on hold. This is to be able to hangup
99 ; a call in the case of a phone disappearing from the net,
100 ; like a powerloss or grandma tripping over a cable.
101 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
102 ; when we're on hold (must be > rtptimeout)
103 ;trustrpid = no ; If Remote-Party-ID should be trusted
104 ;sendrpid = yes ; If Remote-Party-ID should be sent
105 ;progressinband=never ; If we should generate in-band ringing always
106 ; use 'never' to never use in-band signalling, even in cases
107 ; where some buggy devices might not render it
108 ; Valid values: yes, no, never Default: never
109 ;useragent=Asterisk PBX ; Allows you to change the user agent string
110 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
111 ; Note that promiscredir when redirects are made to the
112 ; local system will cause loops since Asterisk is incapable
113 ; of performing a "hairpin" call.
114 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
115 ; a valid phone number
116 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
118 ; info : SIP INFO messages
119 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
120 ; auto : Use rfc2833 if offered, inband otherwise
122 ;compactheaders = yes ; send compact sip headers.
124 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
125 ; in the this section to get any video support at all.
126 ; You can turn it off on a per peer basis if the general
127 ; video support is enabled, but you can't enable it for
128 ; one peer only without enabling in the general section.
129 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
130 ; Videosupport and maxcallbitrate is settable
131 ; for peers and users as well
132 ;callevents=no ; generate manager events when sip ua
133 ; performs events (e.g. hold)
134 ;limitpeersonly=no ; Apply all call limits ("limit=") only to peers, never
135 ; to users. This improves handling of call limits
136 ; and device states in certain situations. The user part
137 ; of a type=friend will still be affected by the call
138 ; limit, but Asterisk will only use one object for
139 ; counting the simultaneous calls.
140 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
141 ; for any reason, always reject with '401 Unauthorized'
142 ; instead of letting the requester know whether there was
143 ; a matching user or peer for their request
145 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
146 ; order instead of RFC3551 packing order (this is required
147 ; for Sipura and Grandstream ATAs, among others). This is
148 ; contrary to the RFC3551 specification, the peer _should_
149 ; be negotiating AAL2-G726-32 instead :-(
152 ; If regcontext is specified, Asterisk will dynamically create and destroy a
153 ; NoOp priority 1 extension for a given peer who registers or unregisters with
154 ; us and have a "regexten=" configuration item.
155 ; Multiple contexts may be specified by separating them with '&'. The
156 ; actual extension is the 'regexten' parameter of the registering peer or its
157 ; name if 'regexten' is not provided. If more than one context is provided,
158 ; the context must be specified within regexten by appending the desired
159 ; context after '@'. More than one regexten may be supplied if they are
160 ; separated by '&'. Patterns may be used in regexten.
162 ;regcontext=sipregistrations
164 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
165 ;sipdebug = yes ; Turn on SIP debugging by default, from
166 ; the moment the channel loads this configuration
167 ;recordhistory=yes ; Record SIP history by default
168 ; (see sip history / sip no history)
169 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue
170 ; SIP history is output to the DEBUG logging channel
173 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
174 ; You can subscribe to the status of extensions with a "hint" priority
175 ; (See extensions.conf.sample for examples)
176 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE
177 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
180 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
181 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
182 ; Useful to limit subscriptions to local extensions
183 ; Settable per peer/user also
184 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
185 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
186 ; Turning on notifyringing and notifyhold will add a lot
187 ; more database transactions if you are using realtime.
189 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
191 ; This setting is available in the [general] section as well as in device configurations.
192 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
193 ; both parties have T38 support enabled in their Asterisk configuration (either general or
194 ; peer/user/friend sections)
196 ; t38pt_udptl = yes ; Default false
198 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
199 ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
200 ; Format for the register statement is:
201 ; register => user[:secret[:authuser]]@host[:port][/extension]
203 ; If no extension is given, the 's' extension is used. The extension needs to
204 ; be defined in extensions.conf to be able to accept calls from this SIP proxy
207 ; host is either a host name defined in DNS or the name of a section defined
210 ; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
211 ; this is equivalent to having the following line in the general section:
213 ; register => username:secret@host/callbackextension
215 ; and more readable because you don't have to write the parameters in two places
216 ; (note that the "port" is ignored - this is a bug that should be fixed).
220 ;register => 1234:password@mysipprovider.com
222 ; This will pass incoming calls to the 's' extension
225 ;register => 2345:password@sip_proxy/1234
227 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
228 ; connect to local extension 1234 in extensions.conf, default context,
229 ; unless you configure a [sip_proxy] section below, and configure a
231 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
232 ; Tip 2: Use separate type=peer and type=user sections for SIP providers
233 ; (instead of type=friend) if you have calls in both directions
235 ;registertimeout=20 ; retry registration calls every 20 seconds (default)
236 ;registerattempts=10 ; Number of registration attempts before we give up
237 ; 0 = continue forever, hammering the other server
238 ; until it accepts the registration
239 ; Default is 0 tries, continue forever
241 ;----------------------------------------- NAT SUPPORT ------------------------
242 ; The externip, externhost and localnet settings are used if you use Asterisk
243 ; behind a NAT device to communicate with services on the outside.
245 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
246 ; messages if we're behind a NAT
248 ; The externip and localnet is used
249 ; when registering and communicating with other proxies
250 ; that we're registered with
251 ;externhost=foo.dyndns.net ; Alternatively you can specify an
252 ; external host, and Asterisk will
253 ; perform DNS queries periodically. Not
254 ; recommended for production
255 ; environments! Use externip instead
256 ;externrefresh=10 ; How often to refresh externhost if
258 ; You may add multiple local networks. A reasonable
259 ; set of defaults are:
260 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
261 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
262 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
263 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
265 ; The nat= setting is used when Asterisk is on a public IP, communicating with
266 ; devices hidden behind a NAT device (broadband router). If you have one-way
267 ; audio problems, you usually have problems with your NAT configuration or your
268 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
269 ; ports for incoming audio in rtp.conf
271 ;nat=no ; Global NAT settings (Affects all peers and users)
272 ; yes = Always ignore info and assume NAT
273 ; no = Use NAT mode only according to RFC3581 (;rport)
274 ; never = Never attempt NAT mode or RFC3581 support
275 ; route = Assume NAT, don't send rport
276 ; (work around more UNIDEN bugs)
278 ;----------------------------------- MEDIA HANDLING --------------------------------
279 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
280 ; no reason for Asterisk to stay in the media path, the media will be redirected.
281 ; This does not really work with in the case where Asterisk is outside and have
282 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
284 ;canreinvite=yes ; Asterisk by default tries to redirect the
285 ; RTP media stream (audio) to go directly from
286 ; the caller to the callee. Some devices do not
287 ; support this (especially if one of them is behind a NAT).
288 ; The default setting is YES. If you have all clients
289 ; behind a NAT, or for some other reason wants Asterisk to
290 ; stay in the audio path, you may want to turn this off.
292 ; This setting also affect direct RTP
293 ; at call setup (a new feature in 1.4 - setting up the
294 ; call directly between the endpoints instead of sending
297 ;canreinvite=nonat ; An additional option is to allow media path redirection
298 ; (reinvite) but only when the peer where the media is being
299 ; sent is known to not be behind a NAT (as the RTP core can
300 ; determine it based on the apparent IP address the media
303 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
304 ; instead of INVITE. This can be combined with 'nonat', as
305 ; 'canreinvite=update,nonat'. It implies 'yes'.
307 ;----------------------------------------- REALTIME SUPPORT ------------------------
308 ; For additional information on ARA, the Asterisk Realtime Architecture,
309 ; please read realtime.txt and extconfig.txt in the /doc directory of the
312 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
313 ; just like friends added from the config file only on a
314 ; as-needed basis? (yes|no)
316 ;rtsavesysname=yes ; Save systemname in realtime database at registration
319 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
320 ; If set to yes, when a SIP UA registers successfully, the ip address,
321 ; the origination port, the registration period, and the username of
322 ; the UA will be set to database via realtime.
323 ; If not present, defaults to 'yes'.
324 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
325 ; as if it had just registered? (yes|no|<seconds>)
326 ; If set to yes, when the registration expires, the friend will
327 ; vanish from the configuration until requested again. If set
328 ; to an integer, friends expire within this number of seconds
329 ; instead of the registration interval.
331 ;ignoreregexpire=yes ; Enabling this setting has two functions:
333 ; For non-realtime peers, when their registration expires, the
334 ; information will _not_ be removed from memory or the Asterisk database
335 ; if you attempt to place a call to the peer, the existing information
336 ; will be used in spite of it having expired
338 ; For realtime peers, when the peer is retrieved from realtime storage,
339 ; the registration information will be used regardless of whether
340 ; it has expired or not; if it expires while the realtime peer
341 ; is still in memory (due to caching or other reasons), the
342 ; information will not be removed from realtime storage
344 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
345 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
346 ; domains, each of which can direct the call to a specific context if desired.
347 ; By default, all domains are accepted and sent to the default context or the
348 ; context associated with the user/peer placing the call.
349 ; Domains can be specified using:
350 ; domain=<domain>[,<context>]
352 ; domain=myasterisk.dom
353 ; domain=customer.com,customer-context
355 ; In addition, all the 'default' domains associated with a server should be
356 ; added if incoming request filtering is desired.
359 ; To disallow requests for domains not serviced by this server:
360 ; allowexternaldomains=no
362 ;domain=mydomain.tld,mydomain-incoming
363 ; Add domain and configure incoming context
364 ; for external calls to this domain
365 ;domain=1.2.3.4 ; Add IP address as local domain
366 ; You can have several "domain" settings
367 ;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
369 ;autodomain=yes ; Turn this on to have Asterisk add local host
370 ; name and local IP to domain list.
372 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
373 ; non-peers, use your primary domain "identity"
374 ; for From: headers instead of just your IP
375 ; address. This is to be polite and
376 ; it may be a mandatory requirement for some
377 ; destinations which do not have a prior
378 ; account relationship with your server.
380 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
381 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
382 ; SIP channel. Defaults to "no". An enabled jitterbuffer will
383 ; be used only if the sending side can create and the receiving
384 ; side can not accept jitter. The SIP channel can accept jitter,
385 ; thus a jitterbuffer on the receive SIP side will be used only
386 ; if it is forced and enabled.
388 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
389 ; channel. Defaults to "no".
391 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
393 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
394 ; resynchronized. Useful to improve the quality of the voice, with
395 ; big jumps in/broken timestamps, usually sent from exotic devices
396 ; and programs. Defaults to 1000.
398 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
399 ; channel. Two implementations are currently available - "fixed"
400 ; (with size always equals to jbmaxsize) and "adaptive" (with
401 ; variable size, actually the new jb of IAX2). Defaults to fixed.
403 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
404 ;-----------------------------------------------------------------------------------
407 ; Global credentials for outbound calls, i.e. when a proxy challenges your
408 ; Asterisk server for authentication. These credentials override
409 ; any credentials in peer/register definition if realm is matched.
411 ; This way, Asterisk can authenticate for outbound calls to other
412 ; realms. We match realm on the proxy challenge and pick an set of
413 ; credentials from this list
415 ; auth = <user>:<secret>@<realm>
416 ; auth = <user>#<md5secret>@<realm>
418 ;auth=mark:topsecret@digium.com
420 ; You may also add auth= statements to [peer] definitions
421 ; Peer auth= override all other authentication settings if we match on realm
423 ;------------------------------------------------------------------------------
424 ; Users and peers have different settings available. Friends have all settings,
425 ; since a friend is both a peer and a user
427 ; User config options: Peer configuration:
428 ; -------------------- -------------------
430 ; callingpres callingpres
434 ; md5secret md5secret
436 ; canreinvite canreinvite
438 ; callgroup callgroup
439 ; pickupgroup pickupgroup
444 ; trustrpid trustrpid
445 ; progressinband progressinband
446 ; promiscredir promiscredir
447 ; useclientcode useclientcode
448 ; accountcode accountcode
452 ; call-limit call-limit
453 ; allowoverlap allowoverlap
454 ; allowsubscribe allowsubscribe
455 ; allowtransfer allowtransfer
456 ; subscribecontext subscribecontext
457 ; videosupport videosupport
458 ; maxcallbitrate maxcallbitrate
459 ; rfc2833compensate mailbox
477 ; For incoming calls only. Example: FWD (Free World Dialup)
478 ; We match on IP address of the proxy for incoming calls
479 ; since we can not match on username (caller id)
485 ;type=peer ; we only want to call out, not be called
487 ;username=yourusername ; Authentication user for outbound proxies
488 ;fromuser=yourusername ; Many SIP providers require this!
489 ;fromdomain=provider.sip.domain
490 ;host=box.provider.com
491 ;usereqphone=yes ; This provider requires ";user=phone" on URI
492 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
493 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
494 ; Call-limits will not be enforced on real-time peers,
495 ; since they are not stored in-memory
497 ;--- sample definition for a provider
500 ;host=sip.provider1.com
501 ;username=4015552299 ; how your provider knows you
502 ;secret=youwillneverguessit
503 ;callbackextension=123 ; Register with this server and require calls coming back to this extension
505 ;------------------------------------------------------------------------------
506 ; Definitions of locally connected SIP devices
508 ; type = user a device that authenticates to us by "from" field to place calls
509 ; type = peer a device we place calls to or that calls us and we match by host
510 ; type = friend two configurations (peer+user) in one
512 ; For device names, we recommend using only a-z, numerics (0-9) and underscore
514 ; For local phones, type=friend works most of the time
516 ; If you have one-way audio, you probably have NAT problems.
517 ; If Asterisk is on a public IP, and the phone is inside of a NAT device
518 ; you will need to configure nat option for those phones.
519 ; Also, turn on qualify=yes to keep the nat session open
521 ; Because you might have a large number of similar sections, it is generally
522 ; convenient to use templates for the common parameters, and add them
523 ; the the various sections. Examples are below, and we can even leave
524 ; the templates uncommented as they will not harm:
526 [basic-options](!) ; a template
531 [natted-phone](!,basic-options) ; another template inheriting basic-options
536 [public-phone](!,basic-options) ; another template inheriting basic-options
540 [my-codecs](!) ; a template for my preferred codecs
548 [ulaw-phone](!) ; and another one for ulaw-only
552 ; and finally instantiate a few phones
554 ; [2133](natted-phone,my-codecs)
556 ; [2134](natted-phone,ulaw-hone)
557 ; secret = not_very_secret
558 ; [2136](public-phone,ulaw-hone)
559 ; secret = not_very_secret_either
563 ; Standard configurations not using templates look like this:
567 ;context=from-sip ; Where to start in the dialplan when this phone calls
568 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
569 ; on incoming calls to Asterisk
570 ;host=192.168.0.23 ; we have a static but private IP address
571 ; No registration allowed
572 ;nat=no ; there is not NAT between phone and Asterisk
573 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
574 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
575 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
576 ; from the phone to asterisk
577 ; 1 for the explicit peer, 1 for the explicit user,
578 ; remember that a friend equals 1 peer and 1 user in
580 ; This will affect your subscriptions as well.
581 ; There is no combined call counter for a "friend"
582 ; so there's currently no way in sip.conf to limit
583 ; to one inbound or outbound call per phone. Use
584 ; the group counters in the dial plan for that.
586 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
587 ;disallow=all ; need to disallow=all before we can use allow=
588 ;allow=ulaw ; Note: In user sections the order of codecs
589 ; listed with allow= does NOT matter!
591 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
592 ;allow=g729 ; Pass-thru only unless g729 license obtained
593 ;callingpres=allowed_passed_screen ; Set caller ID presentation
594 ; See README.callingpres for more information
598 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
599 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
601 ;regexten=1234 ; When they register, create extension 1234
602 ;callerid="Jane Smith" <5678>
603 ;host=dynamic ; This device needs to register
604 ;nat=yes ; X-Lite is behind a NAT router
605 ;canreinvite=no ; Typically set to NO if behind NAT
607 ;allow=gsm ; GSM consumes far less bandwidth than ulaw
610 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
614 ;type=friend ; Friends place calls and receive calls
615 ;context=from-sip ; Context for incoming calls from this user
617 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
618 ;language=de ; Use German prompts for this user
619 ;host=dynamic ; This peer register with us
620 ;dtmfmode=inband ; Choices are inband, rfc2833, or info
621 ;defaultip=192.168.0.59 ; IP used until peer registers
622 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
623 ;subscribemwi=yes ; Only send notifications if this phone
624 ; subscribes for mailbox notification
625 ;vmexten=voicemail ; dialplan extension to reach mailbox
626 ; sets the Message-Account in the MWI notify message
627 ; defaults to global vmexten which defaults to "asterisk"
629 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
633 ;type=friend ; Friends place calls and receive calls
634 ;context=from-sip ; Context for incoming calls from this user
636 ;host=dynamic ; This peer register with us
637 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
638 ;username=polly ; Username to use in INVITE until peer registers
639 ; Normally you do NOT need to set this parameter
641 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
642 ;progressinband=no ; Polycom phones don't work properly with "never"
649 ;insecure=port ; Allow matching of peer by IP address without
650 ; matching port number
651 ;insecure=invite ; Do not require authentication of incoming INVITEs
652 ;insecure=port,invite ; (both)
653 ;qualify=1000 ; Consider it down if it's 1 second to reply
654 ; Helps with NAT session
655 ; qualify=yes uses default value
657 ; Call group and Pickup group should be in the range from 0 to 63
659 ;callgroup=1,3-4 ; We are in caller groups 1,3,4
660 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
661 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
662 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
663 ;permit=192.168.0.60/255.255.255.0
668 ;qualify=200 ; Qualify peer is no more than 200ms away
669 ;nat=yes ; This phone may be natted
670 ; Send SIP and RTP to the IP address that packet is
671 ; received from instead of trusting SIP headers
672 ;host=dynamic ; This device registers with us
673 ;canreinvite=no ; Asterisk by default tries to redirect the
674 ; RTP media stream (audio) to go directly from
675 ; the caller to the callee. Some devices do not
676 ; support this (especially if one of them is
678 ;defaultip=192.168.0.4 ; IP address to use until registration
679 ;username=goran ; Username to use when calling this device before registration
680 ; Normally you do NOT need to set this parameter
681 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
687 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
688 ; You must have this turned on or DTMF reception will work improperly.
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