diff --git a/extensions.conf b/extensions.conf index d2496ba..983891c 100644 --- a/extensions.conf +++ b/extensions.conf @@ -801,7 +801,12 @@ same => n,Hangup() [local] exten => _22X,1,Dial(SIP/${EXTEN:1}) +same => n,Hangup +exten => _X.,1,Dial(IAX2/hetzner/${EXTEN}) +same => n,Hangup + +[oldlocal] exten => _0X.,1,Goto(to-eventphone,${EXTEN:1},1) ; Open bidding, read start bid @@ -845,3 +850,8 @@ same => n,Hangup() [to-eventphone] exten => _X.,1,SET(CALLERID(all)="Teleshop" <7467>) same => n,Dial(SIP/${EXTEN}@eventphone) + +[fromhetzner] +exten => _2X,1,Originate(Console/dsp,exten,auction-spy,s,1,,a) +same => n,Dial(SIP/${EXTEN}) +same => n,Hangup() diff --git a/http.conf b/http.conf index 1fac8a3..a97998f 100644 --- a/http.conf +++ b/http.conf @@ -26,13 +26,13 @@ servername=Asterisk ; Whether HTTP/HTTPS interface is enabled or not. Default is no. ; This also affects manager/rawman/mxml access (see manager.conf) ; -enabled=yes +;enabled=yes ; ; Address to bind to, both for HTTP and HTTPS. You MUST specify ; a bindaddr in order for the HTTP server to run. There is no ; default value. ; -bindaddr=0.0.0.0 +;bindaddr=0.0.0.0 ; ; Port to bind to for HTTP sessions (default is 8088) ; diff --git a/iax.conf b/iax.conf new file mode 100644 index 0000000..e4c31b8 --- /dev/null +++ b/iax.conf @@ -0,0 +1,680 @@ +; +; Inter-Asterisk eXchange v2 (IAX2) Channel Driver configuration +; +; This configuration is read when the chan_iax2.so module is loaded, and is +; re-read when the module is reloaded, such as when invoking the CLI command: +; +; *CLI> iax2 reload +; + +; General settings, like port number to bind to, and an option address (the +; default is to bind to all local addresses). + +[general] + +; Listener Addresses +; +; Use the 'bindaddr' and 'bindport' options to specify on which address and port +; the IAX2 channel driver will listen for incoming requests. +; +; + +;bindport=4569 ; The default port to listen on + ; NOTE: bindport must be specified BEFORE bindaddr or + ; may be specified on a specific bindaddr if followed by + ; colon and port (e.g. bindaddr=192.168.0.1:4569) or for + ; IPv6 the address needs to be in brackets then colon + ; and port (e.g. bindaddr=[2001:db8::1]:4569). + +bindaddr=172.19.27.1 ; You can specify 'bindaddr' more than once to bind to + ; multiple addresses, but the first will be the + ; default. IPv6 addresses are accepted. + +; +; Set 'iaxcompat' to yes if you plan to use layered switches or some other +; scenario which may cause some delay when doing a lookup in the dialplan. It +; incurs a small performance hit to enable it. This option causes Asterisk to +; spawn a separate thread when it receives an IAX2 DPREQ (Dialplan Request) +; instead of blocking while it waits for a response. +; +; Accepted values: yes, no +; Default value: no +; +;iaxcompat=yes +; + +; +; Disable UDP checksums (if nochecksums is set, then no checkums will +; be calculated/checked on systems supporting this feature) +; +; Accepted values: yes, no +; Default value: no +; +;nochecksums=yes +; + +; +; For increased security against brute force password attacks enable +; 'delayreject' which will delay the sending of authentication reject for REGREQ +; or AUTHREP if there is a password. +; +; Accepted values: yes, no +; Default value: no +; +;delayreject=yes +; + +; +; You may specify a global default AMA flag for iaxtel calls. These flags are +; used in the generation of call detail records. +; +; Accepted values: default, omit, billing, documentation +; Default value: default +; +;amaflags=billing +; + +; +; ADSI (Analog Display Services Interface) can be enabled if you have (or may +; have) ADSI compatible CPE equipment. +; +; Accepted values: yes, no +; Default value: no +; +;adsi=yes +; + +; +; Whether or not to perform an SRV lookup on outbound calls. +; +; Accepted values: yes, no +; Default value: no +; +;srvlookup=yes +; + +; +; You may specify a default account for Call Detail Records (CDRs) in addition to +; specifying on a per-user basis. +; +; Accepted values: Any string value up to 19 characters in length +; Default value: +; +;accountcode=lss0101 +; + +; +; You may specify a global default language for users. This can be specified +; also on a per-user basis. If omitted, will fallback to English (en). +; +; Accepted values: A language tag such as 'en' or 'es' +; Default value: en +; +;language=en +; + +; +; 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. +; +; If this option is set to "passthrough", then the hold message will always be +; passed through as signalling instead of generating hold music locally. +; +; This option may be specified globally, or on a per-user or per-peer basis. +; +; Accepted values: passthrough, or any music-on-hold class name +; Default value: +; +;mohinterpret=default +; + +; +; The 'mohsuggest' 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 +; + +; +; Specify bandwidth of low, medium, or high to control which codecs are used +; in general. +; +bandwidth=low +; + +; +; You can also fine tune codecs here using "allow" and "disallow" clauses with +; specific codecs. Use "all" to represent all formats. +; +disallow=all +allow=alaw +;disallow=g723.1 +;disallow=lpc10 +;allow=gsm +; + +;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; +; Jitter Buffer +;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; + +; +; You can adjust several parameters relating to the jitter buffer. The jitter +; buffer's function is to compensate for varying network delay. +; +; All of the jitter buffer settings are in milliseconds. The jitter buffer +; works for INCOMING audio only - the outbound audio will be dejittered by the +; jitter buffer at the other end. +; +; jitterbuffer=yes|no: global default as to whether you want +; the jitter buffer at all. +; +; maxjitterbuffer: a maximum size for the jitter buffer. +; Setting a reasonable maximum here will prevent the call delay +; from rising to silly values in extreme situations; you'll hear +; SOMETHING, even though it will be jittery. +; +; resyncthreshold: when the jitterbuffer notices a significant change in delay +; that continues over a few frames, it will resync, assuming that the change in +; delay was caused by a timestamping mix-up. The threshold for noticing a +; change in delay is measured as twice the measured jitter plus this resync +; threshold. +; Resyncing can be disabled by setting this parameter to -1. +; +; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer +; should return in a row. Since some clients do not send CNG/DTX frames to +; indicate silence, the jitterbuffer will assume silence has begun after +; returning this many interpolations. This prevents interpolating throughout +; a long silence. +; +; jittertargetextra: number of milliseconds by which the new jitter buffer +; will pad its size. the default is 40, so without modification, the new +; jitter buffer will set its size to the jitter value plus 40 milliseconds. +; increasing this value may help if your network normally has low jitter, +; but occasionally has spikes. +; + +jitterbuffer=no +;maxjitterbuffer=1000 +;maxjitterinterps=10 +;resyncthreshold=1000 +;jittertargetextra=40 + +;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; +; IAX2 Encryption +;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; + +; +; Enable IAX2 encryption. The default is no. +; +;encryption=yes +; + +; +; Force encryption insures no connection is established unless both sides +; support encryption. By turning this option on, encryption is automatically +; turned on as well. The default is no. +; +;forceencryption=yes +; + +; This option defines the maximum payload in bytes an IAX2 trunk can support at +; a given time. The best way to explain this is to provide an example. If the +; maximum number of calls to be supported is 800, and each call transmits 20ms +; frames of audio using ulaw: +; +; (8000hz / 1000ms) * 20ms * 1 byte per sample = 160 bytes per frame +; +; The maximum load in bytes is: +; +; (160 bytes per frame) * (800 calls) = 128000 bytes +; +; Once this limit is reached, calls may be dropped or begin to lose audio. +; Depending on the codec in use and number of channels to be supported this value +; may need to be raised, but in most cases the default value is large enough. +; +; trunkmaxsize = 128000 ; defaults to 128000 bytes, which supports up to 800 + ; calls of ulaw at 20ms a frame. + +; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice +; quality when allowing the Linux system to handle fragmentation of UDP packets. +; Depending on the size of each payload, allowing the OS to handle fragmentation +; may not be very efficient. This setting sets the maximum transmission unit for +; IAX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload +; is over 1240 bytes for every 20ms it will be broken into multiple 1240 byte +; messages. Zero disables this functionality and let's the OS handle +; fragmentation. +; +; trunkmtu = 1240 ; trunk data will be sent in 1240 byte messages. + +; trunkfreq sets how frequently trunk messages are sent in milliseconds. This +; value is 20ms by default, which means the trunk will send all the data queued +; to it in the past 20ms. By increasing the time between sending trunk messages, +; the trunk's payload size will increase as well. Note, depending on the size +; set by trunkmtu, messages may be sent more often than specified. For example +; if a trunk's message size grows to the trunkmtu size before 20ms is reached +; that message will be sent immediately. Acceptable values are between 10ms and +; 1000ms. +; +; trunkfreq=20 ; How frequently to send trunk msgs (in ms). This is 20ms by + ; default. + +; Should we send timestamps for the individual sub-frames within trunk frames? +; There is a small bandwidth use for these (less than 1kbps/call), but they +; ensure that frame timestamps get sent end-to-end properly. If both ends of +; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame +; length for your codecs, you can probably suppress these. The receiver must +; also support this feature, although they do not also need to have it enabled. +; +; trunktimestamps=yes + +; Minimum and maximum amounts of time that IAX2 peers can request as a +; registration expiration interval (in seconds). +; minregexpire = 60 +; maxregexpire = 60 + +; IAX2 helper threads + +; Establishes the number of iax helper threads to handle I/O. +; iaxthreadcount = 10 + +; Establishes the number of extra dynamic threads that may be spawned to handle I/O +; iaxmaxthreadcount = 100 + +; +; We can register with another IAX2 server to let him know where we are +; in case we have a dynamic IP address for example +; +; Register with tormenta using username marko and password secretpass +; +;register => marko:secretpass@tormenta.linux-support.net +; +; Register joe at remote host with no password +; +;register => joe@remotehost:5656 +; +; Register marko at tormenta.linux-support.net using RSA key "torkey" +; +;register => marko:[torkey]@tormenta.linux-support.net +; +; Sample Registration for iaxtel +; +; Visit http://www.iaxtel.com to register with iaxtel. Replace "user" +; and "pass" with your username and password for iaxtel. Incoming +; calls arrive at the "s" extension of "default" context. +; +;register => user:pass@iaxtel.com +; +; Sample Registration for IAX2 + FWD +; +; To register using IAX2 with FWD, it must be enabled by visiting the URL +; http://www.fwdnet.net/index.php?section_id=112 +; +; Note that you need an extension in you default context which matches +; your free world dialup number. Please replace "FWDNumber" with your +; FWD number and "passwd" with your password. +; +;register => FWDNumber:passwd@iax.fwdnet.net +; +; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the +; perceived external network address has changed. When the stun_monitor is installed and +; configured, chan_iax will renew all outbound registrations when the monitor detects any sort +; of network change has occurred. By default this option is enabled, but only takes effect once +; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not +; generate all outbound registrations on a network change, use the option below to disable +; this feature. +; +; subscribe_network_change_event = yes ; on by default +; +; You can enable authentication debugging to increase the amount of +; debugging traffic. +; +;authdebug = yes +; +; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters. +;tos=ef +;cos=5 +; +; 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. The actual extension is the 'regexten' parameter of the registering +; peer or its name if 'regexten' is not provided. More than one regexten +; may be supplied if they are separated by '&'. Patterns may be used in +; regexten. +; +;regcontext=iaxregistrations +; +; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes, +; then we cancel the whole thing (that's enough time for one retransmission +; only). This is used to keep things from stalling for a long time for a host +; that is not available, but would be ill advised for bad connections. In +; addition to 'yes' or 'no' you can also specify a number of milliseconds. +; See 'qualify' for individual peers to turn on for just a specific peer. +; +autokill=yes +; +; codecpriority controls the codec negotiation of an inbound IAX2 call. +; This option is inherited to all user entities. It can also be defined +; in each user entity separately which will override the setting in general. +; +; The valid values are: +; +; caller - Consider the callers preferred order ahead of the host's. +; host - Consider the host's preferred order ahead of the caller's. +; disabled - Disable the consideration of codec preference altogether. +; (this is the original behaviour before preferences were added) +; reqonly - Same as disabled, only do not consider capabilities if +; the requested format is not available the call will only +; be accepted if the requested format is available. +; +; The default value is 'host' +; +;codecpriority=host +; +; allowfwdownload controls whether this host will serve out firmware to +; IAX2 clients which request it. This has only been used for the IAXy, +; and it has been recently proven that this firmware distribution method +; can be used as a source of traffic amplification attacks. Also, the +; IAXy firmware has not been updated for at least 18 months, so unless +; you are provisioning IAXys in a secure network, we recommend that you +; leave this option to the default, off. +; +;allowfwdownload=yes + +;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 IAX2 peer registers successfully, + ; the IP address, the origination port, the registration period, + ; and the username of the peer will be set to database via realtime. + ; If not present, defaults to 'yes'. + +;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule + ; as if it had just registered? (yes|no|) + ; 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. + +;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration + ; has expired based on its registration interval, used the stored + ; address information regardless. (yes|no) + +;parkinglot=edvina ; Default parkinglot for IAX2 peers and users + ; This can also be configured per device + ; Parkinglots are defined in features.conf + +; +; The following two options are used to disable call token validation for the +; purposes of interoperability with IAX2 endpoints that do not yet support it. +; +; Call token validation can be set as optional for a single IP address or IP +; address range by using the 'calltokenoptional' option. 'calltokenoptional' is +; only a global option. +; +;calltokenoptional=209.16.236.73/255.255.255.0 +; +; By setting 'requirecalltoken=no', call token validation becomes optional for +; that peer/user. By setting 'requirecalltoken=auto', call token validation +; is optional until a call token supporting peer registers successfully using +; call token validation. This is used as an indication that from now on, we +; can require it from this peer. So, requirecalltoken is internally set to yes. +; requirecalltoken may only be used in peer/user/friend definitions, +; not in the global scope. +; By default, 'requirecalltoken=yes'. +; +;requirecalltoken=no +; +; Maximum time allowed for call token authentication handshaking. Default is 10 seconds. +; Use higher values in lagged or high packet loss networks. +; +;calltokenexpiration=10 + +; +; These options are used to limit the amount of call numbers allocated to a +; single IP address. Before changing any of these values, it is highly encouraged +; to read the user guide associated with these options first. In most cases, the +; default values for these options are sufficient. +; +; The 'maxcallnumbers' option limits the amount of call numbers allowed for each +; individual remote IP address. Once an IP address reaches it's call number +; limit, no more new connections are allowed until the previous ones close. This +; option can be used in a peer definition as well, but only takes effect for +; the IP of a dynamic peer after it completes registration. +; +;maxcallnumbers=512 +; +; The 'maxcallnumbers_nonvalidated' is used to set the combined number of call +; numbers that can be allocated for connections where call token validation +; has been disabled. Unlike the 'maxcallnumbers' option, this limit is not +; separate for each individual IP address. Any connection resulting in a +; non-call token validated call number being allocated contributes to this +; limit. For use cases, see the call token user guide. This option's +; default value of 8192 should be sufficient in most cases. +; +;maxcallnumbers_nonvalidated=1024 +; +; The [callnumberlimits] section allows custom call number limits to be set +; for specific IP addresses and IP address ranges. These limits take precedence +; over the global 'maxcallnumbers' option, but may still be overridden by a +; peer defined 'maxcallnumbers' entry. Note that these limits take effect +; for every individual address within the range, not the range as a whole. +; +;[callnumberlimits] +;10.1.1.0/255.255.255.0 = 24 +;10.1.2.0/255.255.255.0 = 32 +; + +; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not +; in square brackets. For example, the Caller*ID value 555.5555 becomes 5555555 +; when this option is enabled. Disabling this option results in no modification +; of the Caller*ID value, which is necessary when the Caller*ID represents something +; that must be preserved. This option can only be used in the [general] section. +; By default this option is on. +; +;shrinkcallerid=yes ; on by default + +;; Guest sections for unauthenticated connection attempts. Just specify an +;; empty secret, or provide no secret section. +;; +;[guest] +;type=user +;context=public +;callerid="Guest IAX User" +; +;; +;; Trust Caller*ID Coming from iaxtel.com +;; +;[iaxtel] +;type=user +;context=default +;auth=rsa +;inkeys=iaxtel +; +;; +;; Trust Caller*ID Coming from iax.fwdnet.net +;; +;[iaxfwd] +;type=user +;context=default +;auth=rsa +;inkeys=freeworlddialup +; +; +; Trust Caller*ID delivered over DUNDi/e164 +; +;[dundi] +;type=user +;dbsecret=dundi/secret +;context=dundi-e164-local + +; +; Further user sections may be added, specifying a context and a secret used +; for connections with that given authentication name. Limited IP based +; access control is allowed by use of "permit", "deny", and "acl" keywords. +; Multiple rules are permitted. Multiple permitted contexts may be specified, +; in which case the first will be the default. You can also override +; Caller*ID so that when you receive a call you set the Caller*ID to be what +; you want instead of trusting what the remote user provides +; +; There are three authentication methods that are supported: md5, plaintext, +; and rsa. The least secure is "plaintext", which sends passwords cleartext +; across the net. "md5" uses a challenge/response md5 sum arrangement, but +; still requires both ends have plain text access to the secret. "rsa" allows +; unidirectional secret knowledge through public/private keys. If "rsa" +; authentication is used, "inkeys" is a list of acceptable public keys on the +; local system that can be used to authenticate the remote peer, separated by +; the ":" character. "outkey" is a single, private key to use to authenticate +; to the other side. Public keys are named /var/lib/asterisk/keys/.pub +; while private keys are named /var/lib/asterisk/keys/.key. Private +; keys should always be 3DES encrypted. +; +; +; NOTE: All hostnames and IP addresses in this file are for example purposes +; only; you should not expect any of them to actually be available for +; your use. +; +;[markster] +;type=user +;context=default +;context=local +;auth=md5,plaintext,rsa +;secret=markpasswd +;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer to the + ; target of the transfer. +;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too +;transfer=no ; Disable IAX2 native transfer +;transfer=mediaonly ; When doing IAX2 native transfers, transfer only + ; the media stream +;jitterbuffer=yes ; Override the global setting and enable the jitter + ; buffer for this user +;maxauthreq=10 ; Set the maximum number of outstanding AUTHREQs + ; waiting for replies. If this limit is reached, + ; any further authentication will be blocked, until + ; the pending requests expire or a reply is + ; received. +;callerid="Mark Spencer" <(256) 428-6275> +;deny=0.0.0.0/0.0.0.0 +;accountcode=markster0101 +;permit=209.16.236.73/255.255.255.0 +;language=en ; Use english as default language +;encryption=yes ; Enable IAX2 encryption. The default is no. +;keyrotate=off ; This is a compatibility option for older versions + ; of IAX2 that do not support key rotation with + ; encryption. This option will disable the + ; IAX_COMMAND_RTENC message. The default is on. + +; +; Peers may also be specified, with a secret and a remote hostname. +; +;[demo] +;type=peer +;username=asterisk +;secret=supersecret +;host=192.168.10.10 +;description=My IAX2 Peer ; Description of this peer, as listed by + ; 'iax2 show peers' +;sendani=no +;host=asterisk.linux-support.net +;port=5036 +;mask=255.255.255.255 +;qualify=yes ; Make sure this peer is alive. +;qualifysmoothing = yes ; Use an average of the last two PONG results to + ; reduce falsely detected LAGGED hosts. The default + ; is 'no.' +;qualifyfreqok = 60000 ; How frequently to ping the peer when everything + ; seems to be OK, in milliseconds. +;qualifyfreqnotok = 10000 ; How frequently to ping the peer when it's either + ; LAGGED or UNAVAILABLE, in milliseconds. +;jitterbuffer=no ; Turn off jitter buffer for this peer +; +;encryption=yes ; Enable IAX2 encryption. The default is no. +;keyrotate=off ; This is a compatibility option for older versions + ; of IAX2 that do not support key rotation with + ; encryption. This option will disable the + ; IAX_COMMAND_RTENC message. The default is 'on.' + +; Peers can remotely register as well, so that they can be mobile. Default +; IPs can also optionally be given but are not required. Caller*ID can be +; suggested to the other side as well if it is for example a phone instead of +; another PBX. +;connectedline=yes ; Set if connected line and redirecting information updates +; ; are passed between Asterisk servers for this peer. +; ; yes - Sending and receiving updates are enabled. +; ; send - Only send updates. +; ; receive - Only process received updates. +; ; no - Sending and receiving updates are disabled. +; ; Default is "no". +; ; +; ; Note: Because of an incompatibility between Asterisk v1.4 +; ; and Asterisk v1.8 or later, this option must be set +; ; to "no" toward the Asterisk v1.4 peer. A symptom of the +; ; incompatibility is the call gets disconnected unexpectedly. + + +;[dynamichost] +;host=dynamic +;secret=mysecret +; Note: app_voicemail mailboxes must be in the form of mailbox@context. +;mailbox=1234 ; Notify about mailbox 1234 +;inkeys=key1:key2 +;peercontext=local ; Default context to request for calls to peer +;defaultip=216.207.245.34 +;callerid="Some Host" <(256) 428-6011> + +;[biggateway] +;type=peer +;host=192.168.0.1 +;description=Gateway to PSTN +;context=* +;secret=myscret +;trunk=yes ; Use IAX2 trunking with this host +;timezone=America/New_York ; Set a timezone for the date/time IE + +; +; Friends are a shortcut for creating a user and a peer with the same values. +; + +;[marko] +;type=friend +;host=dynamic +;regexten=1234 +;secret=moofoo ; Multiple secrets may be specified. For a "user", all +;secret=foomoo ; specified entries will be accepted as valid. For a "peer", +;secret=shazbot ; only the last specified secret will be used. +;context=default +;permit=0.0.0.0/0.0.0.0 +;acl=example_named_acl + +; +; With immediate=yes, an IAX2 phone or a phone on an IAXy acts as a hot-line +; which goes immediately to the s extension when picked up. Useful for +; elevator phones, manual service, or other similar applications. +; +;[manual] +;type=friend +;host=dynamic +;immediate=yes ; go immediately to s extension when picked up +;secret=moofoo ; when immediate=yes is specified, secret is required +;context=number-please ; we start at the s extension in this context +; + +[hetzner] +type=friend +host=172.19.27.2 +username=stubnitz +secret=f9DuF9Sh0GRKUFrF +context=fromhetzner +qualify=yes +qualifyfreqok=10000 +qualifyfreqnotok=10000 diff --git a/iaxprov.conf b/iaxprov.conf new file mode 100644 index 0000000..8b7d8e0 --- /dev/null +++ b/iaxprov.conf @@ -0,0 +1,80 @@ +; +; IAX2 Provisioning Information +; +; Contains provisioning information for templates and for specific service +; entries. +; +; Templates provide a group of settings from which provisioning takes place. +; A template may be based upon any template that has been specified before +; it. If the template that an entry is based on is not specified then it is +; presumed to be 'default' (unless it is the first of course). +; +; Templates which begin with 'si-' are used for provisioning units with +; specific service identifiers. For example the entry "si-000364000126" +; would be used when the device with the corresponding service identifier of +; "000364000126" attempts to register or make a call. +; +[default] +; +; The port number the device should use to bind to. The default is 4569. +; +;port=4569 +; +; server is our PRIMARY server for registration and placing calls +; +;server=192.168.69.3 +; +; altserver is the BACKUP server for registration and placing calls in the +; event the primary server is unavailable. +; +;altserver=192.168.69.4 +; +; port is the port number to use for IAX2 outbound. The connections to the +; server and altserver -- default is of course 4569. +;serverport=4569 +; +; language is the preferred language for the device +; +;language=en +; +; codec is the requested codec. The iaxy supports ulaw and adpcm +; +codec=ulaw +; +; flags is a comma separated list of flags which the device should +; use and may contain any of the following keywords: +; +; "register" - Register with server +; "secure" - Do not accept calls / provisioning not originated by the server +; "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255 +; "debug" - Output extra debugging to port 9999 +; +; Note that use can use += and -= to adjust parameters +; +flags=register,heartbeat +; +; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter. +;tos=ef +; +; Example iaxy provisioning +; +;[si-000364000126] +;user=iaxy +;pass=bitsy +;flags += debug + +;[si-000364000127] +;user=iaxy2 +;pass=bitsy2 +;template=si-000364000126 +;flags += debug + +; +;[*] +; +; If specified, the '*' provisioning is used for all devices which do not +; have another provisioning entry within the file. If unspecified, no +; provisioning will take place for devices which have no entry. DO NOT +; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING. +; +;template=default diff --git a/oss.conf b/oss.conf index 78c1c1c..efa7735 100644 --- a/oss.conf +++ b/oss.conf @@ -29,7 +29,7 @@ autoanswer = yes ; no autoanswer on call autohangup = yes ; hangup when other party closes ; extension = s ; default extension to call - ; context = default ; default context for outgoing calls + context = local ; default context for outgoing calls ; language = "" ; default language ; If you set overridecontext to 'yes', then the whole dial string diff --git a/sip.conf b/sip.conf index 96da34b..3021119 100644 --- a/sip.conf +++ b/sip.conf @@ -818,8 +818,10 @@ qualify=yes ; Examples: ; ;register => 1234:password@mysipprovider.com -register => 7467:12zX22bH02q2@eventphone.de -register => 7467:COwcNVlnErGf@voip.eventphone.de + +;register => 7467:12zX22bH02q2@eventphone.de +;register => 7467:COwcNVlnErGf@voip.eventphone.de + ; ; This will pass incoming calls to the 's' extension ; @@ -1621,32 +1623,32 @@ registerattempts=0 ; Number of registration attempts before we give u ; external IP address of the remote device. If port forwarding is done at the client side ; then UDPTL will flow to the remote device. -[eventphone] -type=friend -host=148.251.63.154 -defaultuser=7467 -secret=12zX22bH02q2 -context=from-eventphone -qualify=yes -disallow=all -allow=alaw -direct-media=no -canrewrite=no -dtmfmode=rfc2833 -insecure=invite,port -qualify=yes - -[eventphone-event] -type=friend -host=93.180.79.35 -defaultuser=7467 -secret=COwcNVlnErGf -context=from-eventphone -qualify=yes -disallow=all -allow=alaw -direct-media=no -canrewrite=no -dtmfmode=rfc2833 -insecure=invite,port -qualify=yes +;[eventphone] +;type=friend +;host=148.251.63.154 +;defaultuser=7467 +;secret=12zX22bH02q2 +;context=from-eventphone +;qualify=yes +;disallow=all +;allow=alaw +;direct-media=no +;canrewrite=no +;dtmfmode=rfc2833 +;insecure=invite,port +;qualify=yes +; +;[eventphone-event] +;type=friend +;host=93.180.79.35 +;defaultuser=7467 +;secret=COwcNVlnErGf +;context=from-eventphone +;qualify=yes +;disallow=all +;allow=alaw +;direct-media=no +;canrewrite=no +;dtmfmode=rfc2833 +;insecure=invite,port +;qualify=yes