diff options
Diffstat (limited to '1.4.23-rc4/configs/sip.conf.sample')
| -rw-r--r-- | 1.4.23-rc4/configs/sip.conf.sample | 685 |
1 files changed, 0 insertions, 685 deletions
diff --git a/1.4.23-rc4/configs/sip.conf.sample b/1.4.23-rc4/configs/sip.conf.sample deleted file mode 100644 index 25bbb7735..000000000 --- a/1.4.23-rc4/configs/sip.conf.sample +++ /dev/null @@ -1,685 +0,0 @@ -; -; SIP Configuration example for Asterisk -; -; Syntax for specifying a SIP device in extensions.conf is -; SIP/devicename where devicename is defined in a section below. -; -; You may also use -; SIP/username@domain to call any SIP user on the Internet -; (Don't forget to enable DNS SRV records if you want to use this) -; -; 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 -; -; Useful CLI commands to check peers/users: -; sip show peers Show all SIP peers (including friends) -; sip show users Show all SIP users (including friends) -; sip show registry Show status of hosts we register with -; -; sip debug Show all SIP messages -; -; module reload chan_sip.so Reload configuration file -; Active SIP peers will not be reconfigured -; - -[general] -context=default ; Default context for incoming calls -;allowguest=no ; Allow or reject guest calls (default is yes) -allowoverlap=no ; Disable overlap dialing support. (Default is yes) -;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) - ; Default is enabled -;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 -bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) - ; bindport is the local UDP port that Asterisk will listen on -bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) -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 - -;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 doc/ip-tos.txt 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. - -;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 -;t1min=100 ; Minimum roundtrip time for messages to monitored hosts - ; Defaults to 100 ms -;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY -;checkmwi=10 ; Default time between mailbox checks for peers -;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" -;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 -; -;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 -;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 -;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 - ; 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 the 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. -;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) -;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, - ; for any reason, always reject with '401 Unauthorized' - ; instead of letting the requester know whether there was - ; a matching user or peer for their request - -;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 :-( - -;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. - -; -; 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 -; -;--------------------------- 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 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 limit set -; for a device. When the call limit is filled, we will indicate busy. Note that -; you need at least 2 in order to be able to do attended transfers. -; -; 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 = yes ; Control whether subscriptions already INUSE get sent - ; RINGING when another call is sent (default: no) -;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. -;limitonpeers = yes ; Apply call limits on peers only. This will improve - ; status notification when you are using type=friend - ; Inbound calls, that really apply to the user part - ; of a friend will now be added to and compared with - ; the peer limit instead of applying two call limits, - ; one for the peer and one for the user. - ; "sip show inuse" will only show active calls on - ; the peer side of a "type=friend" object if this - ; setting is turned on. - -;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- -; -; This setting is available in the [general] section as well as in device configurations. -; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided -; both parties have T38 support enabled in their Asterisk configuration -; This has to be enabled in the general section for all devices to work. You can then -; disable it on a per device basis. -; -; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used. -; -; t38pt_udptl = yes ; Default false -; -;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ -; Asterisk can register as a SIP user agent to a SIP proxy (provider) -; Format for the register statement is: -; register => user[:secret[:authuser]]@host[:port][/extension] -; -; 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). -; -; host is either a host name defined in DNS or the name of a section defined -; below. -; -; Examples: -; -;register => 1234:password@mysipprovider.com -; -; This will pass incoming calls to the 's' extension -; -; -;register => 2345:password@sip_proxy/1234 -; -; Register 2345 at sip provider 'sip_proxy'. Calls from this provider -; connect to local extension 1234 in extensions.conf, default context, -; unless you configure a [sip_proxy] section below, and configure a -; context. -; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] -; Tip 2: Use separate type=peer and type=user sections for SIP providers -; (instead of type=friend) if you have calls in both directions - -;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 - -;----------------------------------------- NAT SUPPORT ------------------------ -; The externip, externhost and localnet settings are used if you use Asterisk -; behind a NAT device to communicate with services on the outside. - -;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP - ; messages if we're behind a NAT - - ; The externip and localnet is used - ; when registering and communicating with other proxies - ; that we're registered with -;externhost=foo.dyndns.net ; Alternatively you can specify an - ; external host, and Asterisk will - ; perform DNS queries periodically. Not - ; recommended for production - ; environments! Use externip instead -;externrefresh=10 ; How often to refresh externhost if - ; used - ; You may add multiple local networks. A reasonable - ; set of defaults are: -;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks -;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 nat= setting is used when Asterisk is on a public IP, communicating with -; devices hidden behind a NAT device (broadband router). If you have one-way -; audio problems, you usually have problems with your NAT configuration or your -; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP -; ports for incoming audio in rtp.conf -; -;nat=no ; Global NAT settings (Affects all peers and users) - ; yes = Always ignore info and assume NAT - ; no = Use NAT mode only according to RFC3581 (;rport) - ; never = Never attempt NAT mode or RFC3581 support - ; route = Assume NAT, don't send rport - ; (work around more UNIDEN bugs) - -;----------------------------------- MEDIA HANDLING -------------------------------- -; By default, Asterisk tries to re-invite the audio 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 with in the case where Asterisk is outside and have -; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat -; -;canreinvite=yes ; 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). - ; The default setting is YES. If you have all clients - ; behind a NAT, or for some other reason wants Asterisk to - ; stay in the audio path, you may want to turn this off. - - ; In Asterisk 1.4 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 canreinvite is enabled when - ; the device is actually behind NAT. - -;canreinvite=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). - -;canreinvite=update ; Yet a third option... use UPDATE for media path redirection, - ; instead of INVITE. This can be combined with 'nonat', as - ; 'canreinvite=update,nonat'. It implies 'yes'. - -;----------------------------------------- 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. - -; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". -;----------------------------------------------------------------------------------- - -[authentication] -; Global credentials for outbound calls, i.e. when a proxy challenges your -; Asterisk server for authentication. These credentials override -; any credentials in peer/register definition if realm is matched. -; -; This way, Asterisk can authenticate for outbound calls to other -; realms. We match realm on the proxy challenge and pick an set of -; credentials from this list -; Syntax: -; auth = <user>:<secret>@<realm> -; auth = <user>#<md5secret>@<realm> -; Example: -;auth=mark:topsecret@digium.com -; -; You may also add auth= statements to [peer] definitions -; Peer auth= override all other authentication settings if we match on realm - -;------------------------------------------------------------------------------ -; Users and peers have different settings available. Friends have all settings, -; since a friend is both a peer and a user -; -; User config options: Peer configuration: -; -------------------- ------------------- -; context context -; callingpres callingpres -; permit permit -; deny deny -; secret secret -; md5secret md5secret -; dtmfmode dtmfmode -; canreinvite canreinvite -; nat nat -; callgroup callgroup -; pickupgroup pickupgroup -; language language -; allow allow -; disallow disallow -; insecure insecure -; trustrpid trustrpid -; progressinband progressinband -; promiscredir promiscredir -; useclientcode useclientcode -; accountcode accountcode -; setvar setvar -; callerid callerid -; amaflags amaflags -; call-limit call-limit -; allowoverlap allowoverlap -; allowsubscribe allowsubscribe -; allowtransfer allowtransfer -; subscribecontext subscribecontext -; videosupport videosupport -; maxcallbitrate maxcallbitrate -; rfc2833compensate mailbox -; t38pt_usertpsource username -; template -; fromdomain -; regexten -; fromuser -; host -; port -; qualify -; defaultip -; rtptimeout -; rtpholdtimeout -; sendrpid -; outboundproxy -; rfc2833compensate -; 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 -;secret=guessit -;username=yourusername ; Authentication user for outbound proxies -;fromuser=yourusername ; Many SIP providers require this! -;fromdomain=provider.sip.domain -;host=box.provider.com -;usereqphone=yes ; This provider requires ";user=phone" on URI -;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer -;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer - ; Call-limits will not be enforced on real-time peers, - ; since they are not stored in-memory -;port=80 ; The port number we want to connect to on the remote side - ; Also used as "defaultport" in combination with "defaultip" settings - -;------------------------------------------------------------------------------ -; Definitions of locally connected SIP devices -; -; type = user a device that authenticates to us by "from" field to place calls -; type = peer a device we place calls to or that calls us and we match by host -; type = friend two configurations (peer+user) in one -; -; 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 - -;[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 -;canreinvite=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 - ; 1 for the explicit peer, 1 for the explicit user, - ; remember that a friend equals 1 peer and 1 user in - ; memory - ; This will affect your subscriptions as well. - ; 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 doc/callingpres.txt 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 -;canreinvite=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 - - -;[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 -;username=polly ; Username to use in INVITE until peer registers - ; 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 -; -; 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 - -;[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 -;canreinvite=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 -;username=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 this device - -;[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. |