It looks like you are attempting to register 'extensions' 6000 and 1000. You need to define these in sip.conf (infact, the sip.conf you provided appears to have no edits at all).
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Andrew Ladanowski > Sent: 21 January 2008 11:55 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Here is my sip.conf I am having > a problemconnecting my X-Litetomy Asterix box > > I keep getting Registration error 404-Not found When I look > at the log file I get. > [Jan 20 14:17:46] NOTICE[2637] chan_sip.c: Registration from > '<sip:[EMAIL PROTECTED]@192.168.3.125>' failed for > '192.168.3.116' - Device does not match ACL [Jan 20 14:21:25] > NOTICE[2637] chan_sip.c: Registration from > '<sip:[EMAIL PROTECTED]@192.168.3.125>' failed for > '192.168.3.116' - Device does not match ACL [Jan 20 14:22:47] > NOTICE[2637] chan_sip.c: Registration from > '"Csilla"<sip:[EMAIL PROTECTED]>' failed for '192.168.3.116' > - Device does not match ACL [Jan 20 14:25:09] NOTICE[2637] > chan_sip.c: Registration from > '<sip:[EMAIL PROTECTED]@192.168.3.125>' failed for > '192.168.3.116' - Device does not match ACL [Jan 20 14:28:47] > NOTICE[2637] chan_sip.c: Registration from > '<sip:[EMAIL PROTECTED]@192.168.3.125>' failed for > '192.168.3.116' - Device does not match ACL [] Here is my > sip.conf I am having a problemconnecting my X-Litetomy Asterix box > > > ; 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/[EMAIL PROTECTED] 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/[EMAIL PROTECTED] ; > 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 > ; > ; 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 > > ;domain=mydomain.tld ; Set default domain for this host > ; If configured, Asterisk will > only allow > ; INVITE and REFER to non-local domains > ; Use "sip show domains" to > list local domains > ;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/README.tos 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. > > ; > ; 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 ; Notify subscriptions on > RINGING state (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. > > ;----------------------------------------- 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[:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] > ; > ; This will pass incoming calls to the 's' extension > ; > ; > ;register => 2345:[EMAIL PROTECTED]/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. > > ;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'. > ;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. > ; 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 > ;allowexternalinvites=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:[EMAIL PROTECTED] > ; > ; 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 > ; username > ; template > ; fromdomain > ; regexten > ; fromuser > ; host > ; port > ; qualify > ; defaultip > ; rtptimeout > ; rtpholdtimeout > ; sendrpid > ; outboundproxy > ; rfc2833compensate > > ;[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. > ; > ;[EMAIL PROTECTED] ; 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 README.callingpres 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 > ;[EMAIL PROTECTED],[EMAIL PROTECTED] ; 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 > ;[EMAIL PROTECTED],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. > > > Andrew Ladanowski > AddInSolutions Inc. > www.addinsol.com > [EMAIL PROTECTED] > Phone: 954-815-2402 > Fax: 954-414-8432 > > > CONFIDENTIAL : The information in this email (including any > attachments) is confidential and may be privileged. If you > are not the intended recipient, you may not and must not > read, print, forward, use or disseminate the information > contained herein. Although this email (and any attachments) > are believed to be free of any virus or other defect that > might affect any computer system into which it is received > and opened, it is the responsibility of the recipient to > ensure that it is free of viruses or defects and no > responsibility is accepted by the sender for any loss or > damage arising or resulting in any way from its receipt or > use. If you are not the intended recipient of this message, > please reply to the sender and include this message and then > delete this message from your inbox and your archive and/or > discarded messages files. > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D > Sent: Sunday, January 20, 2008 9:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] I am having a problem > connecting my X-Litetomy Asterix box > > Basically, You will need to send the sip.conf file. It will > not work unless you have stuff set up in sip.conf. > > x-Lite works fine; I'm using it without a hitch. > > HTH, > Shane > > On 1/20/08, Erik Anderson <[EMAIL PROTECTED]> wrote: > > On Jan 20, 2008 8:06 PM, Andrew Ladanowski > > <[EMAIL PROTECTED]> > > wrote: > > > Windows XP. > > > > Andrew - you're going to need to get us your sip.conf before we can > > really assist you any further. > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > -Shane > Blog: http://blind-geek.com/blog/ > CoOwner: http://sjtechzone.com > AIM: inhaddict > Skype: chatter8712 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
