Asterisk PBX Integration Zimlet

Interested in talking about Mash-up's? This is the place.
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Thu Aug 19, 2010 9:07 am

hi typiquement
pls check the file








10.48.3.93

5038

zimbra

sachin12345



8000



true



intranet



SIP



03185858







s|.|-|,|((0)|(|))





00



SMS_MESSAGE



sms-send



CAPI/g1/0622100000








true

160



http://tel.local.ch/q/>



ext=1



phone





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


watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 7:54 am

[quote user="chlauber"]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.[/QUOTE]
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
typiquement
Posts: 29
Joined: Sat Sep 13, 2014 12:52 am

Asterisk PBX Integration Zimlet

Postby typiquement » Mon Aug 23, 2010 9:01 am

you have definied "intranet"
can you show your sip.conf and manager.conf with the [intranet] part ?
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 12:20 pm

[quote user="typiquement"]you have definied "intranet"
can you show your sip.conf and manager.conf with the [intranet] part ?[/QUOTE]
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 " 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
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 12:29 pm

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 ------------------------------------------
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 12:59 pm

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= ; 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[: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:password@mysipprovider.com

;

; This will pass incoming calls to the 's' extension

;

;

;register => 2345:password@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 ------------------------
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 1:01 pm

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|)

; 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=[,]

; 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 = :@

; auth = #@

; 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
;------------------------------------------------------------------------------
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 1:02 pm

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 ; 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"

;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
typiquement
Posts: 29
Joined: Sat Sep 13, 2014 12:52 am

Asterisk PBX Integration Zimlet

Postby typiquement » Mon Aug 23, 2010 1:04 pm

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 by in your config.xml for the zimlet. And change 03185858 by . But maybe you need this line.
watanesachin
Posts: 18
Joined: Sat Sep 13, 2014 1:24 am

Asterisk PBX Integration Zimlet

Postby watanesachin » Mon Aug 23, 2010 1:08 pm

[quote user="watanesachin"]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 ; 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"

;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[/QUOTE]



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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