comparison: asterisk/sip.conf
asterisk/sip.conf
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| 1 ; | |
| 2 ; SIP Configuration example for Asterisk | |
| 3 ; | |
| 4 ; Syntax for specifying a SIP device in extensions.conf is | |
| 5 ; SIP/devicename where devicename is defined in a section below. | |
| 6 ; | |
| 7 ; You may also use | |
| 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) | |
| 10 ; | |
| 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 | |
| 14 ; | |
| 15 ; Useful CLI commands to check peers/users: | |
| 16 ; sip show peers Show all SIP peers (including friends) | |
| 17 ; sip show users Show all SIP users (including friends) | |
| 18 ; sip show registry Show status of hosts we register with | |
| 19 ; | |
| 20 ; sip debug Show all SIP messages | |
| 21 ; | |
| 22 ; reload chan_sip.so Reload configuration file | |
| 23 ; Active SIP peers will not be reconfigured | |
| 24 ; | |
| 25 | |
| 26 ;[general] | |
| 27 ;context=default ; Default context for incoming calls | |
| 28 ;allowguest=no ; Allow or reject guest calls (default is yes) | |
| 29 ;allowoverlap=no ; Disable overlap dialing support. (Default is yes) | |
| 30 ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) | |
| 31 ; Default is enabled | |
| 32 ;realm=mydomain.tld ; Realm for digest authentication | |
| 33 ; defaults to "asterisk". If you set a system name in | |
| 34 ; asterisk.conf, it defaults to that system name | |
| 35 ; Realms MUST be globally unique according to RFC 3261 | |
| 36 ; Set this to your host name or domain name | |
| 37 ;bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) | |
| 38 ; bindport is the local UDP port that Asterisk will listen on | |
| 39 ;bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) | |
| 40 ;srvlookup=yes ; Enable DNS SRV lookups on outbound calls | |
| 41 ; Note: Asterisk only uses the first host | |
| 42 ; in SRV records | |
| 43 ; Disabling DNS SRV lookups disables the | |
| 44 ; ability to place SIP calls based on domain | |
| 45 ; names to some other SIP users on the Internet | |
| 46 | |
| 47 ;domain=mydomain.tld ; Set default domain for this host | |
| 48 ; If configured, Asterisk will only allow | |
| 49 ; INVITE and REFER to non-local domains | |
| 50 ; Use "sip show domains" to list local domains | |
| 51 ;pedantic=yes ; Enable checking of tags in headers, | |
| 52 ; international character conversions in URIs | |
| 53 ; and multiline formatted headers for strict | |
| 54 ; SIP compatibility (defaults to "no") | |
| 55 | |
| 56 ; See doc/README.tos for a description of these parameters. | |
| 57 ;tos_sip=cs3 ; Sets TOS for SIP packets. | |
| 58 ;tos_audio=ef ; Sets TOS for RTP audio packets. | |
| 59 ;tos_video=af41 ; Sets TOS for RTP video packets. | |
| 60 | |
| 61 ;maxexpiry=3600 ; Maximum allowed time of incoming registrations | |
| 62 ; and subscriptions (seconds) | |
| 63 ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) | |
| 64 ;defaultexpiry=120 ; Default length of incoming/outgoing registration | |
| 65 ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts | |
| 66 ; Defaults to 100 ms | |
| 67 ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY | |
| 68 ;checkmwi=10 ; Default time between mailbox checks for peers | |
| 69 ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC | |
| 70 ; fully. Enable this option to not get error messages | |
| 71 ; when sending MWI to phones with this bug. | |
| 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 | |
| 78 ; | |
| 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. | |
| 83 ; | |
| 84 ; This option may be specified globally, or on a per-user or per-peer basis. | |
| 85 ; | |
| 86 ;mohinterpret=default | |
| 87 ; | |
| 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. | |
| 91 ; | |
| 92 ;mohsuggest=default | |
| 93 ; | |
| 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 ;trustrpid = no ; If Remote-Party-ID should be trusted | |
| 98 ;sendrpid = yes ; If Remote-Party-ID should be sent | |
| 99 ;progressinband=never ; If we should generate in-band ringing always | |
| 100 ; use 'never' to never use in-band signalling, even in cases | |
| 101 ; where some buggy devices might not render it | |
| 102 ; Valid values: yes, no, never Default: never | |
| 103 ;useragent=Asterisk PBX ; Allows you to change the user agent string | |
| 104 ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address | |
| 105 ; Note that promiscredir when redirects are made to the | |
| 106 ; local system will cause loops since Asterisk is incapable | |
| 107 ; of performing a "hairpin" call. | |
| 108 ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains | |
| 109 ; a valid phone number | |
| 110 ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 | |
| 111 ; Other options: | |
| 112 ; info : SIP INFO messages | |
| 113 ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) | |
| 114 ; auto : Use rfc2833 if offered, inband otherwise | |
| 115 | |
| 116 ;compactheaders = yes ; send compact sip headers. | |
| 117 ; | |
| 118 ;videosupport=yes ; Turn on support for SIP video. You need to turn this on | |
| 119 ; in the this section to get any video support at all. | |
| 120 ; You can turn it off on a per peer basis if the general | |
| 121 ; video support is enabled, but you can't enable it for | |
| 122 ; one peer only without enabling in the general section. | |
| 123 ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s) | |
| 124 ; Videosupport and maxcallbitrate is settable | |
| 125 ; for peers and users as well | |
| 126 ;callevents=no ; generate manager events when sip ua | |
| 127 ; performs events (e.g. hold) | |
| 128 ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, | |
| 129 ; for any reason, always reject with '401 Unauthorized' | |
| 130 ; instead of letting the requester know whether there was | |
| 131 ; a matching user or peer for their request | |
| 132 | |
| 133 ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing | |
| 134 ; order instead of RFC3551 packing order (this is required | |
| 135 ; for Sipura and Grandstream ATAs, among others). This is | |
| 136 ; contrary to the RFC3551 specification, the peer _should_ | |
| 137 ; be negotiating AAL2-G726-32 instead :-( | |
| 138 | |
| 139 ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches | |
| 140 ; your localnet setting. Unless you have some sort of strange network | |
| 141 ; setup you will not need to enable this. | |
| 142 | |
| 143 ; | |
| 144 ; If regcontext is specified, Asterisk will dynamically create and destroy a | |
| 145 ; NoOp priority 1 extension for a given peer who registers or unregisters with | |
| 146 ; us and have a "regexten=" configuration item. | |
| 147 ; Multiple contexts may be specified by separating them with '&'. The | |
| 148 ; actual extension is the 'regexten' parameter of the registering peer or its | |
| 149 ; name if 'regexten' is not provided. If more than one context is provided, | |
| 150 ; the context must be specified within regexten by appending the desired | |
| 151 ; context after '@'. More than one regexten may be supplied if they are | |
| 152 ; separated by '&'. Patterns may be used in regexten. | |
| 153 ; | |
| 154 ;regcontext=sipregistrations | |
| 155 ; | |
| 156 ;--------------------------- RTP timers ---------------------------------------------------- | |
| 157 ; These timers are currently used for both audio and video streams. The RTP timeouts | |
| 158 ; are only applied to the audio channel. | |
| 159 ; The settings are settable in the global section as well as per device | |
| 160 ; | |
| 161 ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity | |
| 162 ; on the audio channel | |
| 163 ; when we're not on hold. This is to be able to hangup | |
| 164 ; a call in the case of a phone disappearing from the net, | |
| 165 ; like a powerloss or grandma tripping over a cable. | |
| 166 ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity | |
| 167 ; on the audio channel | |
| 168 ; when we're on hold (must be > rtptimeout) | |
| 169 ;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open | |
| 170 ; (default is off - zero) | |
| 171 ;--------------------------- SIP DEBUGGING --------------------------------------------------- | |
| 172 ;sipdebug = yes ; Turn on SIP debugging by default, from | |
| 173 ; the moment the channel loads this configuration | |
| 174 ;recordhistory=yes ; Record SIP history by default | |
| 175 ; (see sip history / sip no history) | |
| 176 ;dumphistory=yes ; Dump SIP history at end of SIP dialogue | |
| 177 ; SIP history is output to the DEBUG logging channel | |
| 178 | |
| 179 | |
| 180 ;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ---------------------------- | |
| 181 ; You can subscribe to the status of extensions with a "hint" priority | |
| 182 ; (See extensions.conf.sample for examples) | |
| 183 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE | |
| 184 ; | |
| 185 ; You will get more detailed reports (busy etc) if you have a call limit set | |
| 186 ; for a device. When the call limit is filled, we will indicate busy. Note that | |
| 187 ; you need at least 2 in order to be able to do attended transfers. | |
| 188 ; | |
| 189 ; For queues, you will need this level of detail in status reporting, regardless | |
| 190 ; if you use SIP subscriptions. Queues and manager use the same internal interface | |
| 191 ; for reading status information. | |
| 192 ; | |
| 193 ; Note: Subscriptions does not work if you have a realtime dialplan and use the | |
| 194 ; realtime switch. | |
| 195 ; | |
| 196 ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) | |
| 197 ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests | |
| 198 ; Useful to limit subscriptions to local extensions | |
| 199 ; Settable per peer/user also | |
| 200 ;notifyringing = yes ; Notify subscriptions on RINGING state (default: no) | |
| 201 ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no) | |
| 202 ; Turning on notifyringing and notifyhold will add a lot | |
| 203 ; more database transactions if you are using realtime. | |
| 204 ;limitonpeers = yes ; Apply call limits on peers only. This will improve | |
| 205 ; status notification when you are using type=friend | |
| 206 ; Inbound calls, that really apply to the user part | |
| 207 ; of a friend will now be added to and compared with | |
| 208 ; the peer limit instead of applying two call limits, | |
| 209 ; one for the peer and one for the user. | |
| 210 ; "sip show inuse" will only show active calls on | |
| 211 ; the peer side of a "type=friend" object if this | |
| 212 ; setting is turned on. | |
| 213 | |
| 214 ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- | |
| 215 ; | |
| 216 ; This setting is available in the [general] section as well as in device configurations. | |
| 217 ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided | |
| 218 ; both parties have T38 support enabled in their Asterisk configuration | |
| 219 ; This has to be enabled in the general section for all devices to work. You can then | |
| 220 ; disable it on a per device basis. | |
| 221 ; | |
| 222 ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used. | |
| 223 ; | |
| 224 ; t38pt_udptl = yes ; Default false | |
| 225 ; | |
| 226 ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ | |
| 227 ; Asterisk can register as a SIP user agent to a SIP proxy (provider) | |
| 228 ; Format for the register statement is: | |
| 229 ; register => user[:secret[:authuser]]@host[:port][/extension] | |
| 230 ; | |
| 231 ; If no extension is given, the 's' extension is used. The extension needs to | |
| 232 ; be defined in extensions.conf to be able to accept calls from this SIP proxy | |
| 233 ; (provider). | |
| 234 ; | |
| 235 ; host is either a host name defined in DNS or the name of a section defined | |
| 236 ; below. | |
| 237 ; | |
| 238 ; Examples: | |
| 239 ; | |
| 240 ;register => 1234:password@mysipprovider.com | |
| 241 ; | |
| 242 ; This will pass incoming calls to the 's' extension | |
| 243 ; | |
| 244 ; | |
| 245 ;register => 2345:password@sip_proxy/1234 | |
| 246 ; | |
| 247 ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider | |
| 248 ; connect to local extension 1234 in extensions.conf, default context, | |
| 249 ; unless you configure a [sip_proxy] section below, and configure a | |
| 250 ; context. | |
| 251 ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] | |
| 252 ; Tip 2: Use separate type=peer and type=user sections for SIP providers | |
| 253 ; (instead of type=friend) if you have calls in both directions | |
| 254 | |
| 255 ;registertimeout=20 ; retry registration calls every 20 seconds (default) | |
| 256 ;registerattempts=10 ; Number of registration attempts before we give up | |
| 257 ; 0 = continue forever, hammering the other server | |
| 258 ; until it accepts the registration | |
| 259 ; Default is 0 tries, continue forever | |
| 260 | |
| 261 ;----------------------------------------- NAT SUPPORT ------------------------ | |
| 262 ; The externip, externhost and localnet settings are used if you use Asterisk | |
| 263 ; behind a NAT device to communicate with services on the outside. | |
| 264 | |
| 265 ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP | |
| 266 ; messages if we're behind a NAT | |
| 267 | |
| 268 ; The externip and localnet is used | |
| 269 ; when registering and communicating with other proxies | |
| 270 ; that we're registered with | |
| 271 ;externhost=foo.dyndns.net ; Alternatively you can specify an | |
| 272 ; external host, and Asterisk will | |
| 273 ; perform DNS queries periodically. Not | |
| 274 ; recommended for production | |
| 275 ; environments! Use externip instead | |
| 276 ;externrefresh=10 ; How often to refresh externhost if | |
| 277 ; used | |
| 278 ; You may add multiple local networks. A reasonable | |
| 279 ; set of defaults are: | |
| 280 ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks | |
| 281 ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 | |
| 282 ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation | |
| 283 ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network | |
| 284 | |
| 285 ; The nat= setting is used when Asterisk is on a public IP, communicating with | |
| 286 ; devices hidden behind a NAT device (broadband router). If you have one-way | |
| 287 ; audio problems, you usually have problems with your NAT configuration or your | |
| 288 ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP | |
| 289 ; ports for incoming audio in rtp.conf | |
| 290 ; | |
| 291 ;nat=no ; Global NAT settings (Affects all peers and users) | |
| 292 ; yes = Always ignore info and assume NAT | |
| 293 ; no = Use NAT mode only according to RFC3581 (;rport) | |
| 294 ; never = Never attempt NAT mode or RFC3581 support | |
| 295 ; route = Assume NAT, don't send rport | |
| 296 ; (work around more UNIDEN bugs) | |
| 297 | |
| 298 ;----------------------------------- MEDIA HANDLING -------------------------------- | |
| 299 ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's | |
| 300 ; no reason for Asterisk to stay in the media path, the media will be redirected. | |
| 301 ; This does not really work with in the case where Asterisk is outside and have | |
| 302 ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat | |
| 303 ; | |
| 304 ;canreinvite=yes ; Asterisk by default tries to redirect the | |
| 305 ; RTP media stream (audio) to go directly from | |
| 306 ; the caller to the callee. Some devices do not | |
| 307 ; support this (especially if one of them is behind a NAT). | |
| 308 ; The default setting is YES. If you have all clients | |
| 309 ; behind a NAT, or for some other reason wants Asterisk to | |
| 310 ; stay in the audio path, you may want to turn this off. | |
| 311 | |
| 312 ; In Asterisk 1.4 this setting also affect direct RTP | |
| 313 ; at call setup (a new feature in 1.4 - setting up the | |
| 314 ; call directly between the endpoints instead of sending | |
| 315 ; a re-INVITE). | |
| 316 | |
| 317 ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up | |
| 318 ; the call directly with media peer-2-peer without re-invites. | |
| 319 ; Will not work for video and cases where the callee sends | |
| 320 ; RTP payloads and fmtp headers in the 200 OK that does not match the | |
| 321 ; callers INVITE. This will also fail if canreinvite is enabled when | |
| 322 ; the device is actually behind NAT. | |
| 323 | |
| 324 ;canreinvite=nonat ; An additional option is to allow media path redirection | |
| 325 ; (reinvite) but only when the peer where the media is being | |
| 326 ; sent is known to not be behind a NAT (as the RTP core can | |
| 327 ; determine it based on the apparent IP address the media | |
| 328 ; arrives from). | |
| 329 | |
| 330 ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, | |
| 331 ; instead of INVITE. This can be combined with 'nonat', as | |
| 332 ; 'canreinvite=update,nonat'. It implies 'yes'. | |
| 333 | |
| 334 ;----------------------------------------- REALTIME SUPPORT ------------------------ | |
| 335 ; For additional information on ARA, the Asterisk Realtime Architecture, | |
| 336 ; please read realtime.txt and extconfig.txt in the /doc directory of the | |
| 337 ; source code. | |
| 338 ; | |
| 339 ;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list | |
| 340 ; just like friends added from the config file only on a | |
| 341 ; as-needed basis? (yes|no) | |
| 342 | |
| 343 ;rtsavesysname=yes ; Save systemname in realtime database at registration | |
| 344 ; Default= no | |
| 345 | |
| 346 ;rtupdate=yes ; Send registry updates to database using realtime? (yes|no) | |
| 347 ; If set to yes, when a SIP UA registers successfully, the ip address, | |
| 348 ; the origination port, the registration period, and the username of | |
| 349 ; the UA will be set to database via realtime. | |
| 350 ; If not present, defaults to 'yes'. | |
| 351 ;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule | |
| 352 ; as if it had just registered? (yes|no|<seconds>) | |
| 353 ; If set to yes, when the registration expires, the friend will | |
| 354 ; vanish from the configuration until requested again. If set | |
| 355 ; to an integer, friends expire within this number of seconds | |
| 356 ; instead of the registration interval. | |
| 357 | |
| 358 ;ignoreregexpire=yes ; Enabling this setting has two functions: | |
| 359 ; | |
| 360 ; For non-realtime peers, when their registration expires, the | |
| 361 ; information will _not_ be removed from memory or the Asterisk database | |
| 362 ; if you attempt to place a call to the peer, the existing information | |
| 363 ; will be used in spite of it having expired | |
| 364 ; | |
| 365 ; For realtime peers, when the peer is retrieved from realtime storage, | |
| 366 ; the registration information will be used regardless of whether | |
| 367 ; it has expired or not; if it expires while the realtime peer | |
| 368 ; is still in memory (due to caching or other reasons), the | |
| 369 ; information will not be removed from realtime storage | |
| 370 | |
| 371 ;----------------------------------------- SIP DOMAIN SUPPORT ------------------------ | |
| 372 ; Incoming INVITE and REFER messages can be matched against a list of 'allowed' | |
| 373 ; domains, each of which can direct the call to a specific context if desired. | |
| 374 ; By default, all domains are accepted and sent to the default context or the | |
| 375 ; context associated with the user/peer placing the call. | |
| 376 ; Domains can be specified using: | |
| 377 ; domain=<domain>[,<context>] | |
| 378 ; Examples: | |
| 379 ; domain=myasterisk.dom | |
| 380 ; domain=customer.com,customer-context | |
| 381 ; | |
| 382 ; In addition, all the 'default' domains associated with a server should be | |
| 383 ; added if incoming request filtering is desired. | |
| 384 ; autodomain=yes | |
| 385 ; | |
| 386 ; To disallow requests for domains not serviced by this server: | |
| 387 ; allowexternaldomains=no | |
| 388 | |
| 389 ;domain=mydomain.tld,mydomain-incoming | |
| 390 ; Add domain and configure incoming context | |
| 391 ; for external calls to this domain | |
| 392 ;domain=1.2.3.4 ; Add IP address as local domain | |
| 393 ; You can have several "domain" settings | |
| 394 ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains | |
| 395 ; Default is yes | |
| 396 ;autodomain=yes ; Turn this on to have Asterisk add local host | |
| 397 ; name and local IP to domain list. | |
| 398 | |
| 399 ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to | |
| 400 ; non-peers, use your primary domain "identity" | |
| 401 ; for From: headers instead of just your IP | |
| 402 ; address. This is to be polite and | |
| 403 ; it may be a mandatory requirement for some | |
| 404 ; destinations which do not have a prior | |
| 405 ; account relationship with your server. | |
| 406 | |
| 407 ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- | |
| 408 ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a | |
| 409 ; SIP channel. Defaults to "no". An enabled jitterbuffer will | |
| 410 ; be used only if the sending side can create and the receiving | |
| 411 ; side can not accept jitter. The SIP channel can accept jitter, | |
| 412 ; thus a jitterbuffer on the receive SIP side will be used only | |
| 413 ; if it is forced and enabled. | |
| 414 | |
| 415 ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP | |
| 416 ; channel. Defaults to "no". | |
| 417 | |
| 418 ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. | |
| 419 | |
| 420 ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is | |
| 421 ; resynchronized. Useful to improve the quality of the voice, with | |
| 422 ; big jumps in/broken timestamps, usually sent from exotic devices | |
| 423 ; and programs. Defaults to 1000. | |
| 424 | |
| 425 ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP | |
| 426 ; channel. Two implementations are currently available - "fixed" | |
| 427 ; (with size always equals to jbmaxsize) and "adaptive" (with | |
| 428 ; variable size, actually the new jb of IAX2). Defaults to fixed. | |
| 429 | |
| 430 ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". | |
| 431 ;----------------------------------------------------------------------------------- | |
| 432 | |
| 433 ;[authentication] | |
| 434 ; Global credentials for outbound calls, i.e. when a proxy challenges your | |
| 435 ; Asterisk server for authentication. These credentials override | |
| 436 ; any credentials in peer/register definition if realm is matched. | |
| 437 ; | |
| 438 ; This way, Asterisk can authenticate for outbound calls to other | |
| 439 ; realms. We match realm on the proxy challenge and pick an set of | |
| 440 ; credentials from this list | |
| 441 ; Syntax: | |
| 442 ; auth = <user>:<secret>@<realm> | |
| 443 ; auth = <user>#<md5secret>@<realm> | |
| 444 ; Example: | |
| 445 ;auth=mark:topsecret@digium.com | |
| 446 ; | |
| 447 ; You may also add auth= statements to [peer] definitions | |
| 448 ; Peer auth= override all other authentication settings if we match on realm | |
| 449 | |
| 450 ;------------------------------------------------------------------------------ | |
| 451 ; Users and peers have different settings available. Friends have all settings, | |
| 452 ; since a friend is both a peer and a user | |
| 453 ; | |
| 454 ; User config options: Peer configuration: | |
| 455 ; -------------------- ------------------- | |
| 456 ; context context | |
| 457 ; callingpres callingpres | |
| 458 ; permit permit | |
| 459 ; deny deny | |
| 460 ; secret secret | |
| 461 ; md5secret md5secret | |
| 462 ; dtmfmode dtmfmode | |
| 463 ; canreinvite canreinvite | |
| 464 ; nat nat | |
| 465 ; callgroup callgroup | |
| 466 ; pickupgroup pickupgroup | |
| 467 ; language language | |
| 468 ; allow allow | |
| 469 ; disallow disallow | |
| 470 ; insecure insecure | |
| 471 ; trustrpid trustrpid | |
| 472 ; progressinband progressinband | |
| 473 ; promiscredir promiscredir | |
| 474 ; useclientcode useclientcode | |
| 475 ; accountcode accountcode | |
| 476 ; setvar setvar | |
| 477 ; callerid callerid | |
| 478 ; amaflags amaflags | |
| 479 ; call-limit call-limit | |
| 480 ; allowoverlap allowoverlap | |
| 481 ; allowsubscribe allowsubscribe | |
| 482 ; allowtransfer allowtransfer | |
| 483 ; subscribecontext subscribecontext | |
| 484 ; videosupport videosupport | |
| 485 ; maxcallbitrate maxcallbitrate | |
| 486 ; rfc2833compensate mailbox | |
| 487 ; username | |
| 488 ; template | |
| 489 ; fromdomain | |
| 490 ; regexten | |
| 491 ; fromuser | |
| 492 ; host | |
| 493 ; port | |
| 494 ; qualify | |
| 495 ; defaultip | |
| 496 ; rtptimeout | |
| 497 ; rtpholdtimeout | |
| 498 ; sendrpid | |
| 499 ; outboundproxy | |
| 500 ; rfc2833compensate | |
| 501 | |
| 502 ;[sip_proxy] | |
| 503 ; For incoming calls only. Example: FWD (Free World Dialup) | |
| 504 ; We match on IP address of the proxy for incoming calls | |
| 505 ; since we can not match on username (caller id) | |
| 506 ;type=peer | |
| 507 ;context=from-fwd | |
| 508 ;host=fwd.pulver.com | |
| 509 | |
| 510 ;[sip_proxy-out] | |
| 511 ;type=peer ; we only want to call out, not be called | |
| 512 ;secret=guessit | |
| 513 ;username=yourusername ; Authentication user for outbound proxies | |
| 514 ;fromuser=yourusername ; Many SIP providers require this! | |
| 515 ;fromdomain=provider.sip.domain | |
| 516 ;host=box.provider.com | |
| 517 ;usereqphone=yes ; This provider requires ";user=phone" on URI | |
| 518 ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer | |
| 519 ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer | |
| 520 ; Call-limits will not be enforced on real-time peers, | |
| 521 ; since they are not stored in-memory | |
| 522 ;port=80 ; The port number we want to connect to on the remote side | |
| 523 ; Also used as "defaultport" in combination with "defaultip" settings | |
| 524 | |
| 525 ;------------------------------------------------------------------------------ | |
| 526 ; Definitions of locally connected SIP devices | |
| 527 ; | |
| 528 ; type = user a device that authenticates to us by "from" field to place calls | |
| 529 ; type = peer a device we place calls to or that calls us and we match by host | |
| 530 ; type = friend two configurations (peer+user) in one | |
| 531 ; | |
| 532 ; For device names, we recommend using only a-z, numerics (0-9) and underscore | |
| 533 ; | |
| 534 ; For local phones, type=friend works most of the time | |
| 535 ; | |
| 536 ; If you have one-way audio, you probably have NAT problems. | |
| 537 ; If Asterisk is on a public IP, and the phone is inside of a NAT device | |
| 538 ; you will need to configure nat option for those phones. | |
| 539 ; Also, turn on qualify=yes to keep the nat session open | |
| 540 | |
| 541 ;[grandstream1] | |
| 542 ;type=friend | |
| 543 ;context=from-sip ; Where to start in the dialplan when this phone calls | |
| 544 ;callerid=John Doe <1234> ; Full caller ID, to override the phones config | |
| 545 ; on incoming calls to Asterisk | |
| 546 ;host=192.168.0.23 ; we have a static but private IP address | |
| 547 ; No registration allowed | |
| 548 ;nat=no ; there is not NAT between phone and Asterisk | |
| 549 ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk | |
| 550 ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone | |
| 551 ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time | |
| 552 ; from the phone to asterisk | |
| 553 ; 1 for the explicit peer, 1 for the explicit user, | |
| 554 ; remember that a friend equals 1 peer and 1 user in | |
| 555 ; memory | |
| 556 ; This will affect your subscriptions as well. | |
| 557 ; There is no combined call counter for a "friend" | |
| 558 ; so there's currently no way in sip.conf to limit | |
| 559 ; to one inbound or outbound call per phone. Use | |
| 560 ; the group counters in the dial plan for that. | |
| 561 ; | |
| 562 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" | |
| 563 ;disallow=all ; need to disallow=all before we can use allow= | |
| 564 ;allow=ulaw ; Note: In user sections the order of codecs | |
| 565 ; listed with allow= does NOT matter! | |
| 566 ;allow=alaw | |
| 567 ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! | |
| 568 ;allow=g729 ; Pass-thru only unless g729 license obtained | |
| 569 ;callingpres=allowed_passed_screen ; Set caller ID presentation | |
| 570 ; See README.callingpres for more information | |
| 571 | |
| 572 | |
| 573 ;[xlite1] | |
| 574 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! | |
| 575 ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed | |
| 576 ;type=friend | |
| 577 ;regexten=1234 ; When they register, create extension 1234 | |
| 578 ;callerid="Jane Smith" <5678> | |
| 579 ;host=dynamic ; This device needs to register | |
| 580 ;nat=yes ; X-Lite is behind a NAT router | |
| 581 ;canreinvite=no ; Typically set to NO if behind NAT | |
| 582 ;disallow=all | |
| 583 ;allow=gsm ; GSM consumes far less bandwidth than ulaw | |
| 584 ;allow=ulaw | |
| 585 ;allow=alaw | |
| 586 ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes | |
| 587 | |
| 588 | |
| 589 ;[snom] | |
| 590 ;type=friend ; Friends place calls and receive calls | |
| 591 ;context=from-sip ; Context for incoming calls from this user | |
| 592 ;secret=blah | |
| 593 ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions | |
| 594 ;language=de ; Use German prompts for this user | |
| 595 ;host=dynamic ; This peer register with us | |
| 596 ;dtmfmode=inband ; Choices are inband, rfc2833, or info | |
| 597 ;defaultip=192.168.0.59 ; IP used until peer registers | |
| 598 ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator | |
| 599 ;subscribemwi=yes ; Only send notifications if this phone | |
| 600 ; subscribes for mailbox notification | |
| 601 ;vmexten=voicemail ; dialplan extension to reach mailbox | |
| 602 ; sets the Message-Account in the MWI notify message | |
| 603 ; defaults to global vmexten which defaults to "asterisk" | |
| 604 ;disallow=all | |
| 605 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! | |
| 606 | |
| 607 | |
| 608 ;[polycom] | |
| 609 ;type=friend ; Friends place calls and receive calls | |
| 610 ;context=from-sip ; Context for incoming calls from this user | |
| 611 ;secret=blahpoly | |
| 612 ;host=dynamic ; This peer register with us | |
| 613 ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info | |
| 614 ;username=polly ; Username to use in INVITE until peer registers | |
| 615 ; Normally you do NOT need to set this parameter | |
| 616 ;disallow=all | |
| 617 ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw! | |
| 618 ;progressinband=no ; Polycom phones don't work properly with "never" | |
| 619 | |
| 620 | |
| 621 ;[pingtel] | |
| 622 ;type=friend | |
| 623 ;secret=blah | |
| 624 ;host=dynamic | |
| 625 ;insecure=port ; Allow matching of peer by IP address without | |
| 626 ; matching port number | |
| 627 ;insecure=invite ; Do not require authentication of incoming INVITEs | |
| 628 ;insecure=port,invite ; (both) | |
| 629 ;qualify=1000 ; Consider it down if it's 1 second to reply | |
| 630 ; Helps with NAT session | |
| 631 ; qualify=yes uses default value | |
| 632 ; | |
| 633 ; Call group and Pickup group should be in the range from 0 to 63 | |
| 634 ; | |
| 635 ;callgroup=1,3-4 ; We are in caller groups 1,3,4 | |
| 636 ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5 | |
| 637 ;defaultip=192.168.0.60 ; IP address to use if peer has not registered | |
| 638 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address | |
| 639 ;permit=192.168.0.60/255.255.255.0 | |
| 640 | |
| 641 ;[cisco1] | |
| 642 ;type=friend | |
| 643 ;secret=blah | |
| 644 ;qualify=200 ; Qualify peer is no more than 200ms away | |
| 645 ;nat=yes ; This phone may be natted | |
| 646 ; Send SIP and RTP to the IP address that packet is | |
| 647 ; received from instead of trusting SIP headers | |
| 648 ;host=dynamic ; This device registers with us | |
| 649 ;canreinvite=no ; Asterisk by default tries to redirect the | |
| 650 ; RTP media stream (audio) to go directly from | |
| 651 ; the caller to the callee. Some devices do not | |
| 652 ; support this (especially if one of them is | |
| 653 ; behind a NAT). | |
| 654 ;defaultip=192.168.0.4 ; IP address to use until registration | |
| 655 ;username=goran ; Username to use when calling this device before registration | |
| 656 ; Normally you do NOT need to set this parameter | |
| 657 ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device | |
| 658 | |
| 659 ;[pre14-asterisk] | |
| 660 ;type=friend | |
| 661 ;secret=digium | |
| 662 ;host=dynamic | |
| 663 ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine. | |
| 664 ; You must have this turned on or DTMF reception will work improperly. |
