Софтфон не коннектится астериску

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Софтфон не коннектится астериску

Непрочитанное сообщение Talay » 2013-06-28 7:59:31


настроил Астериск на фрибсд

дальше иду коннектится с софтфона. нету коннекта. а с сервера выходит такая ошибка:

Код: Выделить всё

[Jun 28 16:53:29] NOTICE[15458]: chan_sip.c:25673 handle_request_register: Registration from '"user"<sip:333@>' failed for '' - Wrong password

вот конфиг:

Код: Выделить всё

 grep -v "^;" /usr/local/etc/asterisk/sip.conf

context=default                 ; Default context for incoming calls
allowguest=no                  ; Allow or reject guest calls (default is yes)
                                ; If your Asterisk is connected to the Internet
                                ; and you have allowguest=yes
                                ; you want to check which services you offer everyone
                                ; out there, by enabling them in the default context (see below).
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
                                ; Can use the Incomplete application to collect the
                                ; needed digits from an ambiguous dialplan match.
                                ; methods (inband, RFC2833, SIP INFO) in the early
                                ; media phase.  Uses the Incomplete application to
                                ; collect the needed digits.
                                ; 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
                                ; You can serve multiple Realms specifying several
                                ; 'domain=...' directives (see below).
                                ; In this case Realm will be based on request 'From'/'To' header
                                ; and should match one of domain names.
                                ; Otherwise default 'realm=...' will be used.

udpbindaddr=             ; IP address to bind UDP listen socket to ( binds to all)
                                ; Optionally add a port number, (default is port 5060)

tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=             ; IP address for TCP server to bind to ( binds to all interfaces)
                                ; Optionally add a port number, (default is port 5060)

                                ; Optionally add a port number, (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

                                ; 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)

transport=udp                   ; Set the default transports.  The order determines the primary default transport.
                                ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.

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 "yes")

                                ; and subscriptions (seconds)
                                ; Default value is 70
                                ; and reported in milliseconds with sip show settings.
                                ; Set to low value if you use low timeout for NAT of UDP sessions
                                ; Default: 60
                                ; Default: 100
                                ; Default: 1
                                ; fully. Enable this option to not get error messages
                                ; when sending MWI to phones with this bug.
                                ; the From: header as the "name" portion. Also fill the
                                ; "user" portion of the URI in the From: header with this
                                ; value if no fromuser is set
                                ; Default: empty
                                ; Message-Account in the MWI notify message
                                ; defaults to "asterisk"

                                ; rather than advertising all joint codec capabilities. This
                                ; limits the other side's codec choice to exactly what we prefer.

disallow=all                   ; First disallow all codecs
allow=alaw                     ; Allow codecs in order of preference
                                ; for framing options
                                ; This may also be set for individual users/peers
                                ; Parkinglots are configured in features.conf
language=ru                    ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers
                                ; to send the identity of the remote party
                                ; This is identical to sendrpid=yes
                                ; to send the identity of the remote party
                                ; change may be immediately transmitted is with a SIP UPDATE request.
                                ; If communicating with another Asterisk server, and you wish to be able
                                ; transmit such UPDATE messages to it, then you must enable this option.
                                ; Otherwise, we will have to wait until we can send a reinvite to
                                ; transmit the information.
                                ; 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 "yes" will stop any media before we have
                                ; call progress (meaning the SIP channel will not send 183 Session
                                ; Progress for early media). Default is "yes". Also make sure that
                                ; the SIP peer is configured with progressinband=never.
                                ; In order for "noanswer" applications to work, you need to run
                                ; the progress() application in the priority before the app.

                                ; 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.
                                ; 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.
                                ; This option is set to "yes" by default.

                                ; INVITE requests are.  By default this option is disabled.

                                ; 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
                                                    ; in the user field of a sip URI, the field be truncated
                                                    ; at the first semicolon seen. This effectively makes
                                                    ; semicolon a non-usable character for peer names, extensions,
                                                    ; and maybe other, less tested things.  This can be useful
                                                    ; for improving compatability with devices that like to use
                                                    ; user options for whatever reason.  The behavior is similar to
                                                    ; how SIP URI's were typically handled in 1.6.2, hence the name.

                      ; Set to yes add Reason header and use Reason header if it is available.
                                        ; default is to look for "asterisk.pem" in current directory

                                      ; If no tlsprivatekey is specified, tlscertfile is searched for
                                      ; for both public and private key.

                           ; Specify protocol for outbound client connections.
                           ; If left unspecified, the default is sslv2.
                                ; 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).

                                ; 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'.

                                ; reinvite on an incoming call leg. This option is useful when
                                ; peered with another SIP user agent that is known to send
                                ; immediate direct media reinvites upon call establishment. Setting
                                ; the option in this situation helps to prevent potential glares.
                                ; Setting this option implies 'yes'.

                                ; 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.

                                ; (There is no default setting, this is just an example)
                                ; Use this if some of your phones are on IP addresses that
                                ; can not reach each other directly. This way you can force
                                ; RTP to always flow through asterisk in such cases.

                                ; 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.

                                ; Like the useragent parameter, the default user agent string
                                ; also contains the Asterisk version.
                                ; This field MUST NOT contain spaces
                                ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
                                ; the peer does not support SRTP. Defaults to no.

                                ; 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.

                              ; AOC-E to snom endpoints.  This option can be used both in the
                              ; peer and global scope.  The default for this option is off.

                              ; 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.

                              ; HASH(SIP_CAUSE,<channel name>) channel variable
                              ; to the value of the last sip response.
                              ; WARNING: enabling this option carries a
                              ; significant performance burden. It should only
                              ; be used in low call volume situations. This
                              ; option defaults to "no".


                                  ; Also used as "defaultport" in combination with "defaultip" settings

[basic-options](!)                ; a template

[natted-phone](!,basic-options)   ; another template inheriting basic-options

[public-phone](!,basic-options)   ; another template inheriting basic-options

[my-codecs](!)                    ; a template for my preferred codecs

[ulaw-phone](!)                   ; and another one for ulaw-only

                                 ; 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
                                 ; apply only to IPv6 addresses, and IPv4 ACLs apply
                                 ; only to IPv4 addresses.

                                 ; 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.


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Сообщения: 84
Зарегистрирован: 2012-11-21 18:33:03

Re: Софтфон не коннектится астериску

Непрочитанное сообщение mikie » 2013-06-28 23:39:24

пишед же вам

Код: Выделить всё

Wrong password
неверный пароль