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Thread: Asterisk PBX Integration Zimlet (new)

  1. #161
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    hi typiquement

    pls check the file


    <zimletConfig name="ch_bnc_asterisk" version="0.65">
    <!--
    This settings be global or local, which means per host. If properties are in global scope host settings will be overwritten.
    -->
    <host name="mail.company.com">
    <property name="astManagerIp">10.48.3.93</property>
    <property name="astManagerPort">5038</property>
    <property name="astManagerUser">zimbra</property>
    <property name="astManagerSecret">sachin12345</property>
    <!--
    Timeout for asterisk actions in [ms]
    -->
    <property name="astActionTimeout">8000</property>
    <!--
    Don't do Action ExtensionState before doing other actions. This may be useful because
    the ExtensionState action does need hints in dialplan to work correctly. But you will have less accurate
    warning messages.
    -->
    <property name="astNoExtenCheck">true</property>
    <!--
    Dialplan context used for dialing
    -->
    <property name="astDialContext">intranet</property>
    <!--
    Asterisk channel type to use. E.g. 'SIP', 'ZAP' etc.
    Action will use ${astChannelType}/${srcPhone}
    srcPhone is part of user config
    -->
    <property name="astDialChannelType">SIP</property>
    <!--
    srcPhonePrefix is used to compose the caller like
    ${srcPhonePrefix}${srcPhone} where ${srcPhone} is
    user configurable local number
    -->
    <property name="srcPhonePrefix">03185858</property>
    <!--
    Prefix prepended to every dialed number
    -->
    <property name="calleePrefix"></property>
    <!--
    Regexp applied to any user inputed number. Matches will be removed.
    -->
    <property name="numberCleanRegExp">\s|\.|-|\,|(\(0\)|\(|\))</property>

    <!--
    When dialing international number '+' gets
    replaced with the iddPrefix
    -->
    <property name="iddPrefix">00</property>
    <!--
    Variable name used in SMS() application for SMS message body
    -->
    <property name="astSMSVariable">SMS_MESSAGE</property>
    <!--
    Dialplan context used for sending sms
    -->
    <property name="astSMSSendContext">sms-send</property>
    <!--
    Full asterisk channel for sms center
    -->
    <property name="astSMSCenterChannel">CAPI/g1/0622100000</property>
    </host>

    <!--
    These settings have to be global because of a limitation when getting properties in zimlet's js.
    -->
    <global>
    <!--
    Enable/Disable SMS sending through asterisk originate function
    Needs a correctly configured SMS() application in dialplan that
    gets the SMS message from dialplan variable "astSMSVariable"
    example for astSMSVariable = 'SMS_MESSASGE':

    [sms-send]
    exten => _X.,n,SMS(${CALLERID(num)},${EXTEN},${SMS_MESSAGE} )
    exten => _X.,n,SMS(${CALLERID(num)})
    exten => _X.,n,Hangup

    -->
    <property name="enableSMS">true</property>
    <property name="maxSMSLength">160</property>
    <!--
    URL to search a Phonebook entry in web
    -->
    <property name="phonebookBaseUrl">http://tel.local.ch/q/</property>
    <!--
    Url parameters used for search in form param1=val1&param2=val2..

    -->
    <property name="phonebookUrlCommonParams">ext=1</property>
    <!--
    Url param which value will be filled with the phone number to search
    -->
    <property name="phonebookUrlNumberParam">phone</property>
    </global>
    </zimletConfig>



    i am useing the version for zcs is zcs-6.0.7_GA_2473.RHEL5.20100616214455.tgz

    pls reply

    i want to call only INTERNAL not EXTERNAL in zimbra
    Last edited by watanesachin; 08-19-2010 at 08:09 AM.

  2. #162
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    Quote Originally Posted by chlauber View Post
    Hi pmx

    I've created a 0.63 beta with a user configurable Context. You must let the astDialContext empty in global config, otherwise usersetting is not working.
    Let me know if it works.
    when i call from Zimbra it show call failed in yellow mark i dont have trunk line i want to call on internal network pls suggest

  3. #163
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    you have definied "<property name="astDialContext">intranet</property>"

    can you show your sip.conf and manager.conf with the [intranet] part ?

  4. #164
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    Quote Originally Posted by typiquement View Post
    you have definied "<property name="astDialContext">intranet</property>"

    can you show your sip.conf and manager.conf with the [intranet] part ?
    this is manager.conf

    ;
    ; AMI - The Asterisk Manager Interface
    ;
    ; Third party application call management support and PBX event supervision
    ;
    ; This configuration file is read every time someone logs in
    ;
    ; Use the "manager list commands" at the CLI to list available manager commands
    ; and their authorization levels.
    ;
    ; "manager show command <command>" will show a help text.
    ;
    ; ---------------------------- SECURITY NOTE -------------------------------
    ; Note that you should not enable the AMI on a public IP address. If needed,
    ; block this TCP port with iptables (or another FW software) and reach it
    ; with IPsec, SSH, or SSL vpn tunnel. You can also make the manager
    ; interface available over http if Asterisk's http server is enabled in
    ; http.conf and if both "enabled" and "webenabled" are set to yes in
    ; this file. Both default to no. httptimeout provides the maximum
    ; timeout in seconds before a web based session is discarded. The
    ; default is 60 seconds.
    ;
    [general]
    displaysystemname = yes
    enabled = yes
    ;webenabled = yes
    port = 5038


    ;httptimeout = 60
    ; a) httptimeout sets the Max-Age of the http cookie
    ; b) httptimeout is the amount of time the webserver waits
    ; on a action=waitevent request (actually its httptimeout-10)
    ; c) httptimeout is also the amount of time the webserver keeps
    ; a http session alive after completing a successful action

    bindaddr = 0.0.0.0
    ;displayconnects = yes
    ;
    ; Add a Unix epoch timestamp to events (not action responses)
    ;
    ;timestampevents = yes

    ;[admin]
    ;secret = sachin1987
    ;deny = 0.0.0.0/0.0.0.0
    ;permit = 127.0.0.1/255.255.255.0
    ;read = call,command
    ;write = call,command




    [zimbra]
    secret = sachin12345
    deny = 0.0.0.0/0.0.0.0
    permit = 10.48.3.174/255.255.0.0
    read = system,call,log,verbose,command,agent,user
    write = system,call,log,verbose,command,agent,user



    ;[mark]
    ;secret = mysecret
    ;deny=0.0.0.0/0.0.0.0
    ;permit=209.16.236.73/255.255.255.0
    ;permit=127.0.0.1/255.255.255.0
    ;
    ; If the device connected via this user accepts input slowly,
    ; the timeout for writes to it can be increased to keep it
    ; from being disconnected (value is in milliseconds)
    ;
    ; writetimeout = 100
    ;
    ; Authorization for various classes
    ;read = system,call,log,verbose,command,agent,user,config
    ;write = system,call,log,verbose,command,agent,user,config

  5. #165
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    this all line i add in sip.conf
    i am sending the sip file in 4 part ...

    1 Part


    ;
    ; SIP Configuration example for Asterisk
    ;
    ; Syntax for specifying a SIP device in extensions.conf is
    ; SIP/devicename where devicename is defined in a section below.
    ;
    ; You may also use
    ; SIP/username@domain to call any SIP user on the Internet
    ; (Don't forget to enable DNS SRV records if you want to use this)
    ;
    ; If you define a SIP proxy as a peer below, you may call
    ; SIP/proxyhostname/user or SIP/user@proxyhostname
    ; where the proxyhostname is defined in a section below
    ;
    ; Useful CLI commands to check peers/users:
    ; sip show peers Show all SIP peers (including friends)
    ; sip show users Show all SIP users (including friends)
    ; sip show registry Show status of hosts we register with
    ;
    ; sip debug Show all SIP messages
    ;
    ; reload chan_sip.so Reload configuration file
    ; Active SIP peers will not be reconfigured
    ;

    [general]
    videosupport=yes ; Enable Asterix video support
    context=default ; Default context for incoming calls
    allowsubscribe = yes
    notifyringing = yes
    notifyhold = yes
    limitonpeers = yes
    ;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/ip-tos.txt for a description of these parameters.
    ;tos_sip=cs3 ; Sets TOS for SIP packets.
    ;tos_audio=ef ; Sets TOS for RTP audio packets.
    ;tos_video=af41 ; Sets TOS for RTP video packets.

    ;maxexpiry=3600 ; Maximum allowed time of incoming registrations
    ; and subscriptions (seconds)
    ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
    ;defaultexpiry=120 ; Default length of incoming/outgoing registration
    ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
    ; Defaults to 100 ms
    ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
    ;checkmwi=10 ; Default time between mailbox checks for peers
    ;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
    ; fully. Enable this option to not get error messages
    ; when sending MWI to phones with this bug.
    ;vmexten=voicemail ; dialplan extension to reach mailbox sets the
    ; Message-Account in the MWI notify message
    ; defaults to "asterisk"
    ;disallow=all ; First disallow all codecs
    ;allow=ulaw ; Allow codecs in order of preference
    ;allow=ilbc ; see doc/rtp-packetization for framing options
    ;
    ; This option specifies a preference for which music on hold class this channel
    ; should listen to when put on hold if the music class has not been set on the
    ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
    ; channel putting this one on hold did not suggest a music class.
    ;
    ; This option may be specified globally, or on a per-user or per-peer basis.
    ;
    ;mohinterpret=default
    ;
    ; This option specifies which music on hold class to suggest to the peer channel
    ; when this channel places the peer on hold. It may be specified globally or on
    ; a per-user or per-peer basis.
    ;
    ;mohsuggest=default
    ;
    ;language=en ; Default language setting for all users/peers
    ; This may also be set for individual users/peers
    ;relaxdtmf=yes ; Relax dtmf handling
    ;trustrpid = no ; If Remote-Party-ID should be trusted
    ;sendrpid = yes ; If Remote-Party-ID should be sent
    ;progressinband=never ; If we should generate in-band ringing always
    ; use 'never' to never use in-band signalling, even in cases
    ; where some buggy devices might not render it
    ; Valid values: yes, no, never Default: never
    ;useragent=Asterisk PBX ; Allows you to change the user agent string
    ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
    ; Note that promiscredir when redirects are made to the
    ; local system will cause loops since Asterisk is incapable
    ; of performing a "hairpin" call.
    ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
    ; a valid phone number
    ;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
    ; Other options:
    ; info : SIP INFO messages
    ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
    ; auto : Use rfc2833 if offered, inband otherwise

    ;compactheaders = yes ; send compact sip headers.
    ;
    ;videosupport=yes ; Turn on support for SIP video. You need to turn this on
    ; in the this section to get any video support at all.
    ; You can turn it off on a per peer basis if the general
    ; video support is enabled, but you can't enable it for
    ; one peer only without enabling in the general section.
    ;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
    ; Videosupport and maxcallbitrate is settable
    ; for peers and users as well
    ;callevents=no ; generate manager events when sip ua
    ; performs events (e.g. hold)
    ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
    ; for any reason, always reject with '401 Unauthorized'
    ; instead of letting the requester know whether there was
    ; a matching user or peer for their request

    ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
    ; order instead of RFC3551 packing order (this is required
    ; for Sipura and Grandstream ATAs, among others). This is
    ; contrary to the RFC3551 specification, the peer _should_
    ; be negotiating AAL2-G726-32 instead :-(

    ;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
    ; your localnet setting. Unless you have some sort of strange network
    ; setup you will not need to enable this.

    ;
    ; 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 ------------------------------------------
    Last edited by watanesachin; 08-23-2010 at 10:58 AM.

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    2] part


    ;--------------------------- 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.
    ; "sip show inuse" will only show active calls on
    ; the peer side of a "type=friend" object if this
    ; setting is turned on.

    ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
    ;
    ; This setting is available in the [general] section as well as in device configurations.
    ; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
    ; both parties have T38 support enabled in their Asterisk configuration
    ; This has to be enabled in the general section for all devices to work. You can then
    ; disable it on a per device basis.
    ;
    ; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
    ;
    ; t38pt_udptl = yes ; Default false
    ;
    ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
    ; Asterisk can register as a SIP user agent to a SIP proxy (provider)
    ; Format for the register statement is:
    ; register => user[:secret[:authuser]]@host[ort][/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 => 1234assword@mysipprovider.com
    ;
    ; This will pass incoming calls to the 's' extension
    ;
    ;
    ;register => 2345assword@sip_proxy/1234
    ;
    ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
    ; connect to local extension 1234 in extensions.conf, default context,
    ; unless you configure a [sip_proxy] section below, and configure a
    ; context.
    ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
    ; Tip 2: Use separate type=peer and type=user sections for SIP providers
    ; (instead of type=friend) if you have calls in both directions

    ;registertimeout=20 ; retry registration calls every 20 seconds (default)
    ;registerattempts=10 ; Number of registration attempts before we give up
    ; 0 = continue forever, hammering the other server
    ; until it accepts the registration
    ; Default is 0 tries, continue forever

    ;----------------------------------------- NAT SUPPORT ------------------------
    ; The externip, externhost and localnet settings are used if you use Asterisk
    ; behind a NAT device to communicate with services on the outside.

    ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP
    ; messages if we're behind a NAT

    ; The externip and localnet is used
    ; when registering and communicating with other proxies
    ; that we're registered with
    ;externhost=foo.dyndns.net ; Alternatively you can specify an
    ; external host, and Asterisk will
    ; perform DNS queries periodically. Not
    ; recommended for production
    ; environments! Use externip instead
    ;externrefresh=10 ; How often to refresh externhost if
    ; used
    ; You may add multiple local networks. A reasonable
    ; set of defaults are:
    ;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
    ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
    ;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
    ;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

    ; The nat= setting is used when Asterisk is on a public IP, communicating with
    ; devices hidden behind a NAT device (broadband router). If you have one-way
    ; audio problems, you usually have problems with your NAT configuration or your
    ; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
    ; ports for incoming audio in rtp.conf
    ;
    ;nat=no ; Global NAT settings (Affects all peers and users)
    ; yes = Always ignore info and assume NAT
    ; no = Use NAT mode only according to RFC3581 (;rport)
    ; never = Never attempt NAT mode or RFC3581 support
    ; route = Assume NAT, don't send rport
    ; (work around more UNIDEN bugs)

    ;----------------------------------- MEDIA HANDLING --------------------------------
    ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
    ; no reason for Asterisk to stay in the media path, the media will be redirected.
    ; This does not really work with in the case where Asterisk is outside and have
    ; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
    ;
    ;canreinvite=yes ; Asterisk by default tries to redirect the
    ; RTP media stream (audio) to go directly from
    ; the caller to the callee. Some devices do not
    ; support this (especially if one of them is behind a NAT).
    ; The default setting is YES. If you have all clients
    ; behind a NAT, or for some other reason wants Asterisk to
    ; stay in the audio path, you may want to turn this off.

    ; In Asterisk 1.4 this setting also affect direct RTP
    ; at call setup (a new feature in 1.4 - setting up the
    ; call directly between the endpoints instead of sending
    ; a re-INVITE).

    ;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
    ; the call directly with media peer-2-peer without re-invites.
    ; Will not work for video and cases where the callee sends
    ; RTP payloads and fmtp headers in the 200 OK that does not match the
    ; callers INVITE. This will also fail if canreinvite is enabled when
    ; the device is actually behind NAT.

    ;canreinvite=nonat ; An additional option is to allow media path redirection
    ; (reinvite) but only when the peer where the media is being
    ; sent is known to not be behind a NAT (as the RTP core can
    ; determine it based on the apparent IP address the media
    ; arrives from).

    ;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
    ; instead of INVITE. This can be combined with 'nonat', as
    ; 'canreinvite=update,nonat'. It implies 'yes'.

    ;----------------------------------------- REALTIME SUPPORT ------------------------

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    3] Part
    ; 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
    ;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
    ; Default is yes
    ;autodomain=yes ; Turn this on to have Asterisk add local host
    ; name and local IP to domain list.

    ; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
    ; non-peers, use your primary domain "identity"
    ; for From: headers instead of just your IP
    ; address. This is to be polite and
    ; it may be a mandatory requirement for some
    ; destinations which do not have a prior
    ; account relationship with your server.

    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
    ; SIP channel. Defaults to "no". An enabled jitterbuffer will
    ; be used only if the sending side can create and the receiving
    ; side can not accept jitter. The SIP channel can accept jitter,
    ; thus a jitterbuffer on the receive SIP side will be used only
    ; if it is forced and enabled.

    ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
    ; channel. Defaults to "no".

    ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

    ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
    ; resynchronized. Useful to improve the quality of the voice, with
    ; big jumps in/broken timestamps, usually sent from exotic devices
    ; and programs. Defaults to 1000.

    ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
    ; channel. Two implementations are currently available - "fixed"
    ; (with size always equals to jbmaxsize) and "adaptive" (with
    ; variable size, actually the new jb of IAX2). Defaults to fixed.

    ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
    ;-----------------------------------------------------------------------------------

    [authentication]
    ; Global credentials for outbound calls, i.e. when a proxy challenges your
    ; Asterisk server for authentication. These credentials override
    ; any credentials in peer/register definition if realm is matched.
    ;
    ; This way, Asterisk can authenticate for outbound calls to other
    ; realms. We match realm on the proxy challenge and pick an set of
    ; credentials from this list
    ; Syntax:
    ; auth = <user>:<secret>@<realm>
    ; auth = <user>#<md5secret>@<realm>
    ; Example:
    ;auth=mark:topsecret@digium.com
    ;
    ; You may also add auth= statements to [peer] definitions
    ; Peer auth= override all other authentication settings if we match on realm

    ;------------------------------------------------------------------------------
    ; Users and peers have different settings available. Friends have all settings,
    ; since a friend is both a peer and a user
    ;
    ; User config options: Peer configuration:
    ; -------------------- -------------------
    ; context context
    ; callingpres callingpres
    ; permit permit
    ; deny deny
    ; secret secret
    ; md5secret md5secret
    ; dtmfmode dtmfmode
    ; canreinvite canreinvite
    ; nat nat
    ; callgroup callgroup
    ; pickupgroup pickupgroup
    ; language language
    ; allow allow
    ; disallow disallow
    ; insecure insecure
    ; trustrpid trustrpid
    ; progressinband progressinband
    ; promiscredir promiscredir
    ; useclientcode useclientcode
    ; accountcode accountcode
    ; setvar setvar
    ; callerid callerid
    ; amaflags amaflags
    ; call-limit call-limit
    ; allowoverlap allowoverlap
    ; allowsubscribe allowsubscribe
    ; allowtransfer allowtransfer
    ; subscribecontext subscribecontext
    ; videosupport videosupport
    ; maxcallbitrate maxcallbitrate
    ; rfc2833compensate mailbox
    ; 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

    ;------------------------------------------------------------------------------

  8. #168
    Join Date
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    Posts
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    Rep Power
    5

    Default

    4] Part

    ; Definitions of locally connected SIP devices
    ;
    ; type = user a device that authenticates to us by "from" field to place calls
    ; type = peer a device we place calls to or that calls us and we match by host
    ; type = friend two configurations (peer+user) in one
    ;
    ; For device names, we recommend using only a-z, numerics (0-9) and underscore
    ;
    ; For local phones, type=friend works most of the time
    ;
    ; If you have one-way audio, you probably have NAT problems.
    ; If Asterisk is on a public IP, and the phone is inside of a NAT device
    ; you will need to configure nat option for those phones.
    ; Also, turn on qualify=yes to keep the nat session open

    ;[grandstream1]
    ;type=friend
    ;context=from-sip ; Where to start in the dialplan when this phone calls
    ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
    ; on incoming calls to Asterisk
    ;host=192.168.0.23 ; we have a static but private IP address
    ; No registration allowed
    ;nat=no ; there is not NAT between phone and Asterisk
    ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
    ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
    ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
    ; from the phone to asterisk
    ; 1 for the explicit peer, 1 for the explicit user,
    ; remember that a friend equals 1 peer and 1 user in
    ; memory
    ; This will affect your subscriptions as well.
    ; There is no combined call counter for a "friend"
    ; so there's currently no way in sip.conf to limit
    ; to one inbound or outbound call per phone. Use
    ; the group counters in the dial plan for that.
    ;
    ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
    ;disallow=all ; need to disallow=all before we can use allow=
    ;allow=ulaw ; Note: In user sections the order of codecs
    ; listed with allow= does NOT matter!
    ;allow=alaw
    ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
    ;allow=g729 ; Pass-thru only unless g729 license obtained
    ;callingpres=allowed_passed_screen ; Set caller ID presentation
    ; See doc/callingpres.txt for more information


    ;[xlite1]
    ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
    ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
    ;type=friend
    ;regexten=1234 ; When they register, create extension 1234
    ;callerid="Jane Smith" <5678>
    ;host=dynamic ; This device needs to register
    ;nat=yes ; X-Lite is behind a NAT router
    ;canreinvite=no ; Typically set to NO if behind NAT
    ;disallow=all
    ;allow=gsm ; GSM consumes far less bandwidth than ulaw
    ;allow=ulaw
    ;allow=alaw
    ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes


    ;[snom]
    ;type=friend ; Friends place calls and receive calls
    ;context=from-sip ; Context for incoming calls from this user
    ;secret=blah
    ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
    ;language=de ; Use German prompts for this user
    ;host=dynamic ; This peer register with us
    ;dtmfmode=inband ; Choices are inband, rfc2833, or info
    ;defaultip=192.168.0.59 ; IP used until peer registers
    ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
    ;subscribemwi=yes ; Only send notifications if this phone
    ; subscribes for mailbox notification
    ;vmexten=voicemail ; dialplan extension to reach mailbox
    ; sets the Message-Account in the MWI notify message
    ; defaults to global vmexten which defaults to "asterisk"
    ;disallow=all
    ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!


    ;[polycom]
    ;type=friend ; Friends place calls and receive calls
    ;context=from-sip ; Context for incoming calls from this user
    ;secret=blahpoly
    ;host=dynamic ; This peer register with us
    ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
    ;username=polly ; Username to use in INVITE until peer registers
    ; Normally you do NOT need to set this parameter
    ;disallow=all
    ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
    ;progressinband=no ; Polycom phones don't work properly with "never"


    ;[pingtel]
    ;type=friend
    ;secret=blah
    ;host=dynamic
    ;insecure=port ; Allow matching of peer by IP address without
    ; matching port number
    ;insecure=invite ; Do not require authentication of incoming INVITEs
    ;insecure=port,invite ; (both)
    ;qualify=1000 ; Consider it down if it's 1 second to reply
    ; Helps with NAT session
    ; qualify=yes uses default value
    ;
    ; Call group and Pickup group should be in the range from 0 to 63
    ;
    ;callgroup=1,3-4 ; We are in caller groups 1,3,4
    ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
    ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
    ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
    ;permit=192.168.0.60/255.255.255.0

    ;[cisco1]
    ;type=friend
    ;secret=blah
    ;qualify=200 ; Qualify peer is no more than 200ms away
    ;nat=yes ; This phone may be natted
    ; Send SIP and RTP to the IP address that packet is
    ; received from instead of trusting SIP headers
    ;host=dynamic ; This device registers with us
    ;canreinvite=no ; Asterisk by default tries to redirect the
    ; RTP media stream (audio) to go directly from
    ; the caller to the callee. Some devices do not
    ; support this (especially if one of them is
    ; behind a NAT).
    ;defaultip=192.168.0.4 ; IP address to use until registration
    ;username=goran ; Username to use when calling this device before registration
    ; Normally you do NOT need to set this parameter
    ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device

    ;[pre14-asterisk]
    ;type=friend
    ;secret=digium
    ;host=dynamic
    ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
    ; You must have this turned on or DTMF reception will work improperly.






    [intranet](!) ; this is template.
    type=friend
    context=intranet
    host=dynamic
    disallow=all
    allow=ulaw
    allow=alaw
    allow=g723
    allow=g729
    dtmfmode=rfc2833




    [1000]
    type=friend
    nat=yes
    canreinvite=no
    insecure=very
    host=dynamic
    secret=1000
    username=1000
    context=intranet




    [1001]
    type=friend
    nat=yes
    canreinvite=no
    insecure=very
    host=dynamic
    secret=1001
    username=1001
    context=intranet

  9. #169
    Join Date
    Nov 2009
    Location
    Nantes, France
    Posts
    29
    Rep Power
    5

    Default

    in your manager.conf, try this :

    bindaddr = 10.48.3.174

    [zimbra]
    secret = sachin12345
    permit = 10.48.0.0/255.255.0.0
    read = system,call,log,verbose,command,agent,user
    write = system,call,log,verbose,command,agent,user

    with a sip phone, you can establish a communication between 1000 and 1001 ?


    if it doesn't work, try to replace <host></host> by <global></global> in your config.xml for the zimlet. And change <property name="srcPhonePrefix">03185858</property> by <property name="srcPhonePrefix"></property>. But maybe you need this line.
    Last edited by typiquement; 08-23-2010 at 11:13 AM.

  10. #170
    Join Date
    Aug 2010
    Posts
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    Rep Power
    5

    Default

    Quote Originally Posted by watanesachin View Post
    4] Part

    ; Definitions of locally connected SIP devices
    ;
    ; type = user a device that authenticates to us by "from" field to place calls
    ; type = peer a device we place calls to or that calls us and we match by host
    ; type = friend two configurations (peer+user) in one
    ;
    ; For device names, we recommend using only a-z, numerics (0-9) and underscore
    ;
    ; For local phones, type=friend works most of the time
    ;
    ; If you have one-way audio, you probably have NAT problems.
    ; If Asterisk is on a public IP, and the phone is inside of a NAT device
    ; you will need to configure nat option for those phones.
    ; Also, turn on qualify=yes to keep the nat session open

    ;[grandstream1]
    ;type=friend
    ;context=from-sip ; Where to start in the dialplan when this phone calls
    ;callerid=John Doe <1234> ; Full caller ID, to override the phones config
    ; on incoming calls to Asterisk
    ;host=192.168.0.23 ; we have a static but private IP address
    ; No registration allowed
    ;nat=no ; there is not NAT between phone and Asterisk
    ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
    ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
    ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
    ; from the phone to asterisk
    ; 1 for the explicit peer, 1 for the explicit user,
    ; remember that a friend equals 1 peer and 1 user in
    ; memory
    ; This will affect your subscriptions as well.
    ; There is no combined call counter for a "friend"
    ; so there's currently no way in sip.conf to limit
    ; to one inbound or outbound call per phone. Use
    ; the group counters in the dial plan for that.
    ;
    ;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
    ;disallow=all ; need to disallow=all before we can use allow=
    ;allow=ulaw ; Note: In user sections the order of codecs
    ; listed with allow= does NOT matter!
    ;allow=alaw
    ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
    ;allow=g729 ; Pass-thru only unless g729 license obtained
    ;callingpres=allowed_passed_screen ; Set caller ID presentation
    ; See doc/callingpres.txt for more information


    ;[xlite1]
    ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
    ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
    ;type=friend
    ;regexten=1234 ; When they register, create extension 1234
    ;callerid="Jane Smith" <5678>
    ;host=dynamic ; This device needs to register
    ;nat=yes ; X-Lite is behind a NAT router
    ;canreinvite=no ; Typically set to NO if behind NAT
    ;disallow=all
    ;allow=gsm ; GSM consumes far less bandwidth than ulaw
    ;allow=ulaw
    ;allow=alaw
    ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes


    ;[snom]
    ;type=friend ; Friends place calls and receive calls
    ;context=from-sip ; Context for incoming calls from this user
    ;secret=blah
    ;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
    ;language=de ; Use German prompts for this user
    ;host=dynamic ; This peer register with us
    ;dtmfmode=inband ; Choices are inband, rfc2833, or info
    ;defaultip=192.168.0.59 ; IP used until peer registers
    ;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
    ;subscribemwi=yes ; Only send notifications if this phone
    ; subscribes for mailbox notification
    ;vmexten=voicemail ; dialplan extension to reach mailbox
    ; sets the Message-Account in the MWI notify message
    ; defaults to global vmexten which defaults to "asterisk"
    ;disallow=all
    ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!


    ;[polycom]
    ;type=friend ; Friends place calls and receive calls
    ;context=from-sip ; Context for incoming calls from this user
    ;secret=blahpoly
    ;host=dynamic ; This peer register with us
    ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
    ;username=polly ; Username to use in INVITE until peer registers
    ; Normally you do NOT need to set this parameter
    ;disallow=all
    ;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
    ;progressinband=no ; Polycom phones don't work properly with "never"


    ;[pingtel]
    ;type=friend
    ;secret=blah
    ;host=dynamic
    ;insecure=port ; Allow matching of peer by IP address without
    ; matching port number
    ;insecure=invite ; Do not require authentication of incoming INVITEs
    ;insecure=port,invite ; (both)
    ;qualify=1000 ; Consider it down if it's 1 second to reply
    ; Helps with NAT session
    ; qualify=yes uses default value
    ;
    ; Call group and Pickup group should be in the range from 0 to 63
    ;
    ;callgroup=1,3-4 ; We are in caller groups 1,3,4
    ;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
    ;defaultip=192.168.0.60 ; IP address to use if peer has not registered
    ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
    ;permit=192.168.0.60/255.255.255.0

    ;[cisco1]
    ;type=friend
    ;secret=blah
    ;qualify=200 ; Qualify peer is no more than 200ms away
    ;nat=yes ; This phone may be natted
    ; Send SIP and RTP to the IP address that packet is
    ; received from instead of trusting SIP headers
    ;host=dynamic ; This device registers with us
    ;canreinvite=no ; Asterisk by default tries to redirect the
    ; RTP media stream (audio) to go directly from
    ; the caller to the callee. Some devices do not
    ; support this (especially if one of them is
    ; behind a NAT).
    ;defaultip=192.168.0.4 ; IP address to use until registration
    ;username=goran ; Username to use when calling this device before registration
    ; Normally you do NOT need to set this parameter
    ;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device

    ;[pre14-asterisk]
    ;type=friend
    ;secret=digium
    ;host=dynamic
    ;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
    ; You must have this turned on or DTMF reception will work improperly.






    [intranet](!) ; this is template.
    type=friend
    context=intranet
    host=dynamic
    disallow=all
    allow=ulaw
    allow=alaw
    allow=g723
    allow=g729
    dtmfmode=rfc2833




    [1000]
    type=friend
    nat=yes
    canreinvite=no
    insecure=very
    host=dynamic
    secret=1000
    username=1000
    context=intranet




    [1001]
    type=friend
    nat=yes
    canreinvite=no
    insecure=very
    host=dynamic
    secret=1001
    username=1001
    context=intranet






    this is mailbox log error

    2010-08-23 23:39:39,681 INFO [Asterisk-Java ManagerConnection-14-Reader-0] [] ManagerReaderImpl - Terminating reader thread: No more lines available: Scanner closed

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