I have a big problem with asterisk. Some days ago i have installed asterisk on my vps ( fortunately without any exit trunk). Today, when i open the log i see something that i don’t understand:
[Aug 10 12:55:20] VERBOSE loader.c: pbx_config.so => (Text Extension Configuration) [Aug 10 12:55:20] VERBOSE pbx.c: == Registered application 'Exec' [Aug 10 12:55:20] VERBOSE pbx.c: == Registered application 'TryExec' [Aug 10 12:55:20] VERBOSE pbx.c: == Registered application 'ExecIf' [Aug 10 12:55:20] VERBOSE loader.c: app_exec.so => (Executes dialplan applications) [Aug 10 12:55:20] VERBOSE config.c: == Parsing '/etc/asterisk/cli_permissions.conf': [Aug 10 12:55:20] VERBOSE config.c: == Found [Aug 10 12:55:20] VERBOSE asterisk.c: Asterisk Ready. [Aug 10 12:55:20] VERBOSE config.c: == Parsing '/etc/asterisk/cli.conf': [Aug 10 12:55:20] VERBOSE config.c: == Found [Aug 10 12:55:21] VERBOSE asterisk.c: -- Remote UNIX connection disconnected [Aug 10 12:56:50] VERBOSE netsock.c: == Using SIP RTP CoS mark 5 [Aug 10 12:56:50] VERBOSE pbx.c: -- Executing [00972592113512@default:1] Answer("SIP/VPS_IP-00000000", "") in new stack [Aug 10 12:56:50] VERBOSE pbx.c: == Spawn extension (default, 00972592113512, 1) exited non-zero on 'SIP/VPS_IP-00000000' [Aug 10 12:56:50] VERBOSE pbx.c: -- Executing [h@default:1] Answer("SIP/VPS_IP-00000000", "") in new stack [Aug 10 12:56:50] VERBOSE pbx.c: == Spawn extension (default, h, 1) exited non-zero on 'SIP/VPS_IP-00000000' [Aug 10 12:56:50] VERBOSE netsock.c: == Using SIP RTP CoS mark 5 [Aug 10 12:56:50] VERBOSE pbx.c: -- Executing [000972592113512@default:1] Answer("SIP/VPS_IP-00000001", "") in new stack [Aug 10 12:56:51] VERBOSE pbx.c: == Spawn extension (default, 000972592113512, 1) exited non-zero on 'SIP/VPS_IP-00000001' [Aug 10 12:56:51] VERBOSE pbx.c: -- Executing [h@default:1] Answer("SIP/VPS_IP-00000001", "") in new stack [Aug 10 12:56:51] VERBOSE pbx.c: == Spawn extension (default, h, 1) exited non-zero on 'SIP/VPS_IP-00000001' [Aug 10 12:56:51] VERBOSE netsock.c: == Using SIP RTP CoS mark 5 [Aug 10 12:56:51] VERBOSE pbx.c: -- Executing [900972592113512@default:1] Answer("SIP/VPS_IP-00000002", "") in new stack [Aug 10 12:56:52] VERBOSE pbx.c: == Spawn extension (default, 900972592113512, 1) exited non-zero on 'SIP/VPS_IP-00000002' [Aug 10 12:56:52] VERBOSE pbx.c: -- Executing [h@default:1] Answer("SIP/VPS_IP-00000002", "") in new stack [Aug 10 12:56:52] VERBOSE pbx.c: == Spawn extension (default, h, 1) exited non-zero on 'SIP/VPS_IP-00000002' [Aug 10 12:56:53] VERBOSE netsock.c: == Using SIP RTP CoS mark 5 [Aug 10 12:56:53] VERBOSE pbx.c: -- Executing [700972592113512@default:1] Answer("SIP/VPS_IP-00000003", "") in new stack [Aug 10 12:56:53] VERBOSE pbx.c: == Spawn extension (default, 700972592113512, 1) exited non-zero on 'SIP/VPS_IP-00000003' [Aug 10 12:56:53] VERBOSE pbx.c: -- Executing [h@default:1] Answer("SIP/VPS_IP-00000003", "") in new stack [Aug 10 12:56:53] VERBOSE pbx.c: == Spawn extension (default, h, 1) exited non-zero on 'SIP/VPS_IP-00000003'
I have no peers or friends in my sip.conf. I don’t know how but someone make call through my asterisk.
Asterisk version is :
vps*CLI> core show version Asterisk 126.96.36.199-2+squeeze10 built by pbuilder @ sweetmorn on a x86_64 running Linux on 2013-01-14 18:32:21 UTC
Also i don’t understand why the source channel is SIP/VPS_IP when VPS_IP is the public ip address of the server
[general] context=default ; Default context for incoming calls ; 'username' field from the authentication line ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ; Default is enabled. The Dial() options 't' and 'T' are not ; related as to whether SIP transfers are allowed or not. ; defaults to "asterisk". If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061) ; Remember that the IP address must match the common name (hostname) in the ; certificate, so you don't want to bind a TLS socket to multiple IP addresses. ; For details how to construct a certificate for SIP see ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs ; default is to look for "asterisk.pem" in current directory ; of seconds a client has to authenticate. If ; the client does not authenticate beofre this ; timeout expires, the client will be ; disconnected. (default: 30 seconds) ; unauthenticated sessions that will be allowed ; to connect at any given time. (default: 100) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. ; international character conversions in URIs ; and multiline formatted headers for strict ; SIP compatibility (defaults to "no") ; and subscriptions (seconds) ; host to be up in seconds ; Set to low value if you use low timeout for ; NAT of UDP sessions ; fully. Enable this option to not get error messages ; when sending MWI to phones with this bug. ; Message-Account in the MWI notify message ; defaults to "asterisk" ; This may also be set for individual users/peers ; Parkinglots are configured in features.conf ; This may also be set for individual users/peers ; the call is in ringing or progress state. The SIP ; channel will then send 183 indicating early media ; which will be empty - thus users get no ring signal. ; Setting this to "no" will stop any media before we have ; call progress. Default is "yes". ; 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 ; The default user agent string also contains the Asterisk ; version. If you don't want to expose this, change the ; useragent string. ; Like the useragent parameter, the default user agent string ; also contains the Asterisk version. ; This field MUST NOT contain spaces ; Note that promiscredir when redirects are made to the ; local system will cause loops since Asterisk is incapable ; of performing a "hairpin" call. ; a valid phone number ; Other options: ; info : SIP INFO messages (application/dtmf-relay) ; shortinfo : SIP INFO messages (application/dtmf) ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) ; auto : Use rfc2833 if offered, inband otherwise ; on in this section to get any video support at all. ; You can turn it off on a per peer basis if the general ; video support is enabled, but you can't enable it for ; one peer only without enabling in the general section. ; If you set videosupport to "always", then RTP ports will ; always be set up for video, even on clients that don't ; support it. This assists callfile-derived calls and ; certain transferred calls to use always use video when ; available. [yes|NO|always] ; Videosupport and maxcallbitrate is settable ; for peers and users as well ; performs events (e.g. hold) ; authenticate with Asterisk. Peerstatus will be "rejected". ; for any reason, always reject with an identical response ; equivalent to valid username and invalid password/hash ; instead of letting the requester know whether there was ; a matching user or peer for their request. This reduces ; the ability of an attacker to scan for valid SIP usernames. ; 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 :-( ; your localnet setting. Unless you have some sort of strange network ; setup you will not need to enable this. ; 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. ; register their phones. ; If you have qualify on and the peer becomes unreachable ; this setting will enforce inactivation of the regexten ; extension for the peer ; Defaults to 100 ms ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 ; 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. ; on the audio channel ; when we're on hold (must be > rtptimeout) ; (default is off - zero) ; the moment the channel loads this configuration ; (see sip history / sip no history) ; SIP history is output to the DEBUG logging channel ; Useful to limit subscriptions to local extensions ; Settable per peer/user also ; RINGING when another call is sent (default: yes) ; Turning on notifyringing and notifyhold will add a lot ; more database transactions if you are using realtime. ; dialog-info+xml notifications (supported by snom phones). ; Note that this feature will only work properly when the ; incoming call is using the same extension and context that ; is being used as the hint for the called extension. This means ; that it won't work when using subscribecontext for your sip ; user or peer (if subscribecontext is different than context). ; This is also limited to a single caller, meaning that if an ; extension is ringing because multiple calls are incoming, ; only one will be used as the source of caller ID. Specify ; 'ignore-context' to ignore the called context when looking ; for the caller's channel. The default value is 'no.' Setting ; notifycid to 'ignore-context' also causes call-pickups attempted ; via SNOM's NOTIFY mechanism to set the context for the call pickup ; to PICKUPMARK. ; device too. ; 0 = continue forever, hammering the other server ; until it accepts the registration ; Default is 0 tries, continue forever ; RTP media stream to go directly from ; the caller to the callee. Some devices do not ; support this (especially if one of them is behind a NAT). ; The default setting is YES. If you have all clients ; behind a NAT, or for some other reason want Asterisk to ; stay in the audio path, you may want to turn this off. ; This setting also affect direct RTP ; at call setup (a new feature in 1.4 - setting up the ; call directly between the endpoints instead of sending ; a re-INVITE). ; the call directly with media peer-2-peer without re-invites. ; Will not work for video and cases where the callee sends ; RTP payloads and fmtp headers in the 200 OK that does not match the ; callers INVITE. This will also fail if directmedia is enabled when ; the device is actually behind NAT. ; Additionally this option does not disable all reINVITE operations. ; It only controls Asterisk generating reINVITEs for the specific ; purpose of setting up a direct media path. If a reINVITE is ; needed to switch a media stream to inactive (when placed on ; hold) or to T.38, it will still be done, regardless of this ; setting. Note that direct T.38 is not supported. ; (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). ; instead of INVITE. This can be combined with 'nonat', as ; 'directmedia=update,nonat'. It implies 'yes'. ; number in SDP packets and will only modify the SDP ; session if the version number changes. This option will ; force asterisk to ignore the SDP session version number ; and treat all SDP data as new data. This is required ; for devices that send us non standard SDP packets ; (observed with Microsoft OCS). By default this option is ; off. ; just like friends added from the config file only on a ; as-needed basis? (yes|no) ; Default= 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. ; 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. ; ; 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 ; Add domain and configure incoming context ; for external calls to this domain ; You can have several "domain" settings ; Default is yes ; name and local IP to domain list. ; 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. ; 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. ; channel. Defaults to "no". ; 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. ; 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. ; The option represents the number of milliseconds by which the new jitter buffer ; will pad its size. the default is 40, so without modification, the new ; jitter buffer will set its size to the jitter value plus 40 milliseconds. ; increasing this value may help if your network normally has low jitter, ; but occasionally has spikes. [authentication] ; Also used as "defaultport" in combination with "defaultip" settings [basic-options](!) ; a template dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options nat=no directmedia=yes [my-codecs](!) ; a template for my preferred codecs disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw [ulaw-phone](!) ; and another one for ulaw-only disallow=all allow=ulaw ; on incoming calls to Asterisk ; No registration allowed ; from the phone to asterisk (deprecated) ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; There is no combined call counter for a "friend" ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ; listed with allow= does NOT matter! ; See README.callingpres for more information ; subscribes for mailbox notification ; sets the Message-Account in the MWI notify message ; defaults to global vmexten which defaults to "asterisk" ; Normally you do NOT need to set this parameter ; matching port number ; Helps with NAT session ; qualify=yes uses default value ; host to be up in seconds ; Set to low value if you use low timeout for ; NAT of UDP sessions ; Send SIP and RTP to the IP address that packet is ; received from instead of trusting SIP headers ; 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). ; Normally you do NOT need to set this parameter ; cause the given audio file to ; be played upon completion of ; an attended transfer. ; You must have this turned on or DTMF reception will work improperly. ; 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.
[general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = yes hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1