SHDSL.bis

در این مقاله توضیحاتی جهت آشنایی بیشتر با مودم های سیپ ترانک SIP Trunk  و یا بهتر بگوییم خطوط 5 رقمی و یا خطوط ویژه مخابراتی و MPLS

که به آنها مودم های دات بیز SHDSL.bis گفته میشود.

SHDSL چیست ؟

Symmetrical High-Bit Rate DSL
 به مشترکین سرویس هاsubscriber

SHDSL یکی از تکنولوژی های xDsl یا همان G.991 است که برای ارتباطات WAN در شرایطی که امکان استفاده از زیرساخت های فیبر نوری موجود نباشد به عنوان ارتباطات پرسرعت بر روی سیم مسی استفاده میشود .

در ایران و حتی کشورهای توسعه یافته از شبکه ATM جهت مدیا برای ارتباطات Poni to Point , Point to MultiPoint , Leased Line , MPLS و … به جهت فراهم نبودن امکانات فیبرنوری و وجود بستر سیم مسی در اکثر نقاط و سرعت مناسب ( در حال حاضر تا 20mb/s ) به وفور استفاده میشود

 

      جدول مقایسه مسافت و پهنای باند در پروتکل EFM

 

         

 

Distance & Rate relationship
26 AWG_ Without Noise _EFM mode
Line rate (kbps)
1-Pair  
Longest reach
 (feet)
2-Pairs  Longest reach (feet)
3-Pairs Longest reach (feet)
4-Pairs Longest reach (feet)
192
18000
18000
18000
18000
256
18000
18000
18000
18000
384
18000
18000
18000
18000
768
16400
16400
16400
16400
2560
11500
11500
11500
11500
3072
10500
10500
10500
10500
3392
10000
10000
10000
10000
3584
9500
9500
9500
9500
3848
9400
9400
9400
9400

 

 

 

 

 

 

 

 

 

 

Specifications

Standard Compliance
 
Comply to ITU-T G.991.2
 
 
Transmission rate up to 5.69 Mbps on a 2-wire line
Transmission rate up to 11.38 Mbps on a 4-wire line
Transmission rate up to 22.76 Mbps on a 8-wire line
Support of Annex A, Annex B, Annex E, Annex F, and Annex G
Support point-to-point configuration which enables two CPEs to communicate end to end

 

 

 

 








.

SIP Value

SIP Configuration example 
;
; Note: Please read the security documentation in order to
; 	understand the risks of installing Asterisk with the sample
;	configuration. If your Asterisk is installed on a public
;	IP address connected to the Internet, you will want to learn
;	about the various security settings BEFORE you start
;	Asterisk. 
;
;	Especially note the following settings:
;		- allowguest (default enabled)
;		- permit/deny - IP address filters
;		- contactpermit/contactdeny - IP address filters for registrations
;		- context - Which set of services you offer various users
;
; SIP dial strings
;-----------------------------------------------------------
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
;        SIP/devicename
;        SIP/username@domain   (SIP uri)
;        SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
;        SIP/devicename/extension
;        SIP/devicename/extension/IPorHost
;        SIP/username@domain//IPorHost
;
;
; Devicename
;        devicename is defined as a peer in a section below.
;
; username@domain
;        Call any SIP user on the Internet
;        (Don't forget to enable DNS SRV records if you want to use this)
;
; devicename/extension
;        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
;        This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
;        This form allows you to specify password or md5secret and authname
;        without altering any authentication data in config.
;        Examples:
;
;        SIP/*98@mysipproxy
;        SIP/sales:topsecret::این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید:5062
;        SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید.0.1
;
; IPorHost
;        The next server for this call regardless of domain/peer
;
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
;
;         SIP/sales@mysipproxy!این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
;
; A new feature for 1.8 allows one to specify a host or IP address to use
; when routing the call. This is typically used in tandem with func_srv if
; multiple methods of reaching the same domain exist. The host or IP address
; is specified after the third slash in the dialstring. Examples:
;
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
;
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
;   sip show peers               Show all SIP peers (including friends)
;   sip show registry            Show status of hosts we register with
;
;   sip set debug on             Show all SIP messages
;
;   sip reload                   Reload configuration file
;   sip show settings            Show the current channel configuration
;
;------- Naming devices ------------------------------------------------------
;
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
;	1. Asterisk checks the SIP From: address username and matches against
;	   names of devices with type=user
;	   The name is the text between square brackets [name]
;	2. Asterisk checks the From: addres and matches the list of devices
;	   with a type=peer
;	3. Asterisk checks the IP address (and port number) that the INVITE
;	   was sent from and matches against any devices with type=peer
;
; Don't mix extensions with the names of the devices. Devices need a unique
; name. The device name is *not* used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
;
; When setting up trunks, make sure there's no risk that any From: username
; (caller ID) will match any of your device names, because then Asterisk
; might match the wrong device.
;
; Note: The parameter "username" is not the username and in most cases is
;       not needed at all. Check below. In later releases, it's renamed
;       to "defaultuser" which is a better name, since it is used in
;       combination with the "defaultip" setting.
;-----------------------------------------------------------------------------

; ** Old configuration options **
; The "call-limit" configuation option is considered old is replaced
; by new functionality. To enable callcounters, you use the new 
; "callcounter" setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
; You can still set limits per device in sip.conf or in a database by using
; "setvar" to set variables that can be used in the dialplan for various limits.

[general]
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).
;match_auth_username=yes        ; if available, match user entry using the
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowoverlap=yes               ; Enable RFC3578 overlap dialing support.
                                ; Can use the Incomplete application to collect the
                                ; needed digits from an ambiguous dialplan match.
;allowoverlap=dtmf              ; Enable overlap dialing support using DTMF delivery
                                ; methods (inband, RFC2833, SIP INFO) in the early
                                ; media phase.  Uses the Incomplete application to
                                ; collect the needed digits.
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
                                ; Default is enabled. The Dial() options 't' and 'T' are not
                                ; related as to whether SIP transfers are allowed or not.
;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
;domainsasrealm=no              ; Use domains list as realms
                                ; 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.

; With the current situation, you can do one of four things:
;  a) Listen on a specific IPv4 address.      Example: bindaddr=192.0.2.1
;  b) Listen on a specific IPv6 address.      Example: bindaddr=2001:db8::1
;  c) Listen on the IPv4 wildcard.            Example: bindaddr=0.0.0.0
;  d) Listen on the IPv4 and IPv6 wildcards.  Example: bindaddr=::
; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)
; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
;  IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
;
; Using bindaddr will only enable UDP support in order to be backwards compatible with those systems
; that were upgraded prior to TCP support. Use udpbindaddr and tcpbindaddr to bind to UDP and TCP
; independently.
;
; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
; for TLS).
;   IPv4 example: bindaddr=0.0.0.0:5062
;   IPv6 example: bindaddr=[::]:5062
;
; The address family of the bound UDP address is used to determine how Asterisk performs
; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
; however, that Asterisk ignores all records except the first one. In case d), when both A
; and AAAA records are available, either an A or AAAA record will be first, and which one
; depends on the operating system. On systems using glibc, AAAA records are given
; priority.

udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

; When a dialog is started with another SIP endpoint, the other endpoint
; should include an Allow header telling us what SIP methods the endpoint
; implements. However, some endpoints either do not include an Allow header
; or lie about what methods they implement. In the former case, Asterisk
; makes the assumption that the endpoint supports all known SIP methods.
; If you know that your SIP endpoint does not provide support for a specific
; method, then you may provide a comma-separated list of methods that your
; endpoint does not implement in the disallowed_methods option. Note that 
; if your endpoint is truthful with its Allow header, then there is no need 
; to set this option. This option may be set in the general section or may
; be set per endpoint. If this option is set both in the general section and
; in a peer section, then the peer setting completely overrides the general
; setting (i.e. the result is *not* the union of the two options).
;
; Note also that while Asterisk currently will parse an Allow header to learn
; what methods an endpoint supports, the only actual use for this currently
; is for determining if Asterisk may send connected line UPDATE requests and
; MESSAGE requests. Its use may be expanded in the future.
;
; disallowed_methods = UPDATE

;
; Note that the TCP and TLS support for chan_sip is currently considered
; experimental.  Since it is new, all of the related configuration options are
; subject to change in any release.  If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
;
tcpenable=no                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0            ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
                                ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
                                ; Remember that the IP address must match the common name (hostname) in the
                                ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
                                ; For details how to construct a certificate for SIP see 
                                ; http://tools.ietf.org/html/draft-ietf-sip-domain-certs

;tcpauthtimeout = 30            ; tcpauthtimeout specifies the maximum number
				; 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)

;tcpauthlimit = 100             ; tcpauthlimit specifies the maximum number of
				; 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.

;pedantic=yes                   ; Enable checking of tags in headers,
                                ; international character conversions in URIs
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "yes")

; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service 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.
;tos_text=af41                  ; Sets TOS for RTP text packets.

;cos_sip=3                      ; Sets 802.1p priority for SIP packets.
;cos_audio=5                    ; Sets 802.1p priority for RTP audio packets.
;cos_video=4                    ; Sets 802.1p priority for RTP video packets.
;cos_text=3                     ; Sets 802.1p priority for RTP text 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
;mwiexpiry=3600                 ; Expiry time for outgoing MWI subscriptions
;maxforwards=70			; Setting for the SIP Max-Forwards: header (loop prevention)
				; Default value is 70
;qualifyfreq=60                 ; Qualification: How often to check for the host to be up in seconds
				; and reported in milliseconds with sip show settings.
                                ; Set to low value if you use low timeout for NAT of UDP sessions
				; Default: 60
;qualifygap=100			; Number of milliseconds between each group of peers being qualified
				; Default: 100
;qualifypeers=1			; Number of peers in a group to be qualified at the same time
				; Default: 1
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;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.
;mwi_from=asterisk              ; When sending MWI NOTIFY requests, use this setting in
                                ; 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
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
                                ; Message-Account in the MWI notify message
                                ; defaults to "asterisk"

; Codec negotiation
;
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
;
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; *not* switch to whatever codec the callee is sending.
;
;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
                                ; 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=ulaw                     ; Allow codecs in order of preference
;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
				; for framing options
;autoframing=yes		; Set packetization based on the remote endpoint's (ptime)
				; preferences. Defaults to no.
;
; 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
;
;parkinglot=plaza               ; Sets the default parking lot for call parking
                                ; This may also be set for individual users/peers
                                ; Parkinglots are configured in features.conf
;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 (defaults to no)
;sendrpid = rpid                ; Use the "Remote-Party-ID" header
                                ; to send the identity of the remote party
                                ; This is identical to sendrpid=yes
;sendrpid = pai                 ; Use the "P-Asserted-Identity" header
                                ; to send the identity of the remote party
;rpid_update = no               ; In certain cases, the only method by which a connected line
                                ; 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.
;trust_id_outbound = no         ; Controls whether or not we trust this peer with private identity
                                ; information (when the remote party has callingpres=prohib or equivalent).
                                ; no - RPID/PAI headers will not be included for private peer information
                                ; yes - RPID/PAI headers will include the private peer information. Privacy
                                ;       requirements will be indicated in a Privacy header for sendrpid=pai
                                ; legacy - RPID/PAI will be included for private peer information. In the
                                ;       case of sendrpid=pai, private data that would be included in them
                                ;       will be anonymized. For sendrpid=rpid, private data may be included
                                ;       but the remote party's domain will be anonymized. The way legacy
                                ;       behaves may violate RFC-3325, but it follows historic behavior.
                                ; This option is set to 'legacy' by default
;prematuremedia=no              ; Some ISDN links send empty media frames before 
                                ; 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.

;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
                                ; The default user agent string also contains the Asterisk
                                ; version. If you don't want to expose this, change the
                                ; useragent 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 (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

;compactheaders = yes           ; send compact sip headers.
;
;videosupport=yes               ; Turn on support for SIP video. You need to turn this
                                ; 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]

;textsupport=no                 ; Support for ITU-T T.140 realtime text.
                                ; The default value is "no".

;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)
;authfailureevents=no           ; generate manager "peerstatus" events when peer can't
                                ; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to 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.

;auth_options_requests = yes    ; Enabling this option will authenticate OPTIONS requests just like
                                ; INVITE requests are.  By default this option is disabled.

;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 :-(
;outboundproxy=proxy.provider.domain            ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080       ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force      ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain      ; same as '=proxy.provider.domain' except we try to connect with tls
;outboundproxy=192.0.2.1                        ; IPv4 address literal (default port is 5060)
;outboundproxy=2001:db8::1                      ; IPv6 address literal (default port is 5060)
;outboundproxy=192.168.0.2.1:5062               ; IPv4 address literal with explicit port
;outboundproxy=[2001:db8::1]:5062               ; IPv6 address literal with explicit port
;                                               ; (could also be tcp,udp) - defining transports on the proxy line only
;                                               ; applies for the global proxy, otherwise use the transport= option
;matchexternaddrlocally = yes     ; Only substitute the externaddr 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.

;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from registering
                                ; 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.

;contactdeny=0.0.0.0/0.0.0.0           ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                       ; register their phones.

;rtp_engine=asterisk            ; RTP engine to use when communicating with the device

;
; 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
;regextenonqualify=yes          ; Default "no"
                                ; If you have qualify on and the peer becomes unreachable
                                ; this setting will enforce inactivation of the regexten
                                ; extension for the peer
;legacy_useroption_parsing=yes	; Default "no"      ; If you have this option enabled and there are semicolons
                                                    ; 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.

; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets.  For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled.  Disabling this option results in no modification
; of the caller id value, which is necessary when the caller id represents something
; that must be preserved.  This option can only be used in the [general] section.
; By default this option is on.
;
;shrinkcallerid=yes     ; on by default


;use_q850_reason = no ; Default "no"
                      ; Set to yes add Reason header and use Reason header if it is available.
;
;------------------------ TLS settings ------------------------------------------------------------
;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
                                        ; The certificates must be sorted starting with the subject's certificate
                                        ; and followed by intermediate CA certificates if applicable.
                                        ; Default is to look for "asterisk.pem" in current directory

;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
                                      ; If no tlsprivatekey is specified, tlscertfile is searched for
                                      ; for both public and private key.

;tlscafile=</path/to/certificate>
;        If the server your connecting to uses a self signed certificate
;        you should have their certificate installed here so the code can
;        verify the authenticity of their certificate.

;tlscapath=</path/to/ca/dir>
;        A directory full of CA certificates.  The files must be named with
;        the CA subject name hash value.
;        (see man SSL_CTX_load_verify_locations for more info)

;tlsdontverifyserver=[yes|no]
;        If set to yes, don't verify the servers certificate when acting as
;        a client.  If you don't have the server's CA certificate you can
;        set this and it will connect without requiring tlscafile to be set.
;        Default is no.

;tlscipher=<SSL cipher string>
;        A string specifying which SSL ciphers to use or not use
;        A list of valid SSL cipher strings can be found at:
;                http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1     ; values include tlsv1, sslv3, sslv2.
                           ; Specify protocol for outbound client connections.
                           ; If left unspecified, the default is sslv2.
;
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
                                ; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
                                ; Defaults to 500 ms or the measured round-trip
                                ; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional response is not received
                                ; in this amount of time, the call will autocongest
                                ; Defaults to 64*timert1

;--------------------------- 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 Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
;
; * session-timers    - Session-Timers feature operates in the following three modes:
;                            originate : Request and run session-timers always
;                            accept    : Run session-timers only when requested by other UA
;                            refuse    : Do not run session timers in any case
;                       The default mode of operation is 'accept'.
; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
;                            uac - Default to the caller initially refreshing when possible
;                            uas - Default to the callee initially refreshing when possible
;
; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
; endpoint's preference for who will handle refreshes. Asterisk will never override the
; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
; fighting over who sends the refreshes. This holds true for the initiation of session
; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
; whether Asterisk is currently the refresher or not.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uac
;
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes                 ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration.
                                ; NOTE: You cannot use the CLI to turn it off. You'll
                                ; need to edit this and reload the config.
;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 counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; 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 = no             ; Control whether subscriptions already INUSE get sent
                                ; RINGING when another call is sent (default: yes)
;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.
;notifycid = yes                ; Control whether caller ID information is sent along with
                                ; 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.
;callcounter = yes              ; Enable call counters on devices. This can be set per
                                ; device too.

;----------------------------------------- T.38 FAX SUPPORT ----------------------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
;
; t38pt_udptl = yes            ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,fec        ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
; t38pt_udptl = yes,none       ; Enables T.38 with no error correction.
;
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value (during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation, and results in failures
; because Asterisk does not believe it can send T.38 packets of a reasonable size to that
; endpoint (Cisco media gateways are one example of this situation). In these cases, during a
; T.38 call you will see warning messages on the console/in the logs from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX packets. If this occurs, you
; can set an override (globally, or on a per-device basis) to make Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a configured value instead.
; This can be done by appending 'maxdatagram=<value>' to the t38pt_udptl configuration option,
; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error correction and overrides
;                                       ; the other endpoint's provided value to assume we can
;                                       ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists)
; based one or more events being detected. The events that can be detected are an incoming
; CNG tone or an incoming T.38 re-INVITE request.
;
; faxdetect = yes		; Default 'no', 'yes' enables both CNG and T.38 detection
; faxdetect = cng		; Enables only CNG detection
; faxdetect = t38		; Enables only T.38 detection
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS  ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
;	- domain in DNS
; 	- host name in DNS
;	- the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; 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).
;
; A similar effect can be achieved by adding a "callbackextension" option in a peer section.
; this is equivalent to having the following line in the general section:
;
;        register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Note that a register= line doesn't mean that we will match the incoming call in any
; other way than described above. If you want to control where the call enters your
; dialplan, which context, you want to define a peer with the hostname of the provider's
; server. If the provider has multiple servers to place calls to your system, you need
; a peer for each server.
;
; Beginning with Asterisk version 1.6.2, the "user" portion of the register line may
; contain a port number. Since the logical separator between a host and port number is a
; ':' character, and this character is already used to separate between the optional "secret"
; and "authuser" portions of the line, there is a bit of a hoop to jump through if you wish
; to use a port here. That is, you must explicitly provide a "secret" and "authuser" even if
; they are blank. See the third example below for an illustration.
;
;
; Examples:
;
;register => 1234:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
;
;     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 inbound and outbound sections for SIP providers
;           (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
;    Note that in this example, the optional authuser and secret portions have
;    been left blank because we have specified a port in the user section
;
;register => tls://username:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
;
;    The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
;    Using 'udp://' explicitly is also useful in case the username part
;    contains a '/' ('user/name').

;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
;register_retry_403=yes         ; Treat 403 responses to registrations as if they were
                                ; 401 responses and continue retrying according to normal
                                ; retry rules.

;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones. At this time, you can only subscribe using UDP as the transport.
; Format for the mwi register statement is:
;       mwi => user[:secret[:authuser]]@host[:port]/mailbox
;
; Examples:
;mwi => 1234:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید/1234
;mwi => 1234:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید:6969/1234
;mwi => 1234:password:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید/1234
;mwi => 1234:password:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید:6969/1234
;
; MWI received will be stored in the 1234 mailbox of the SIP_Remote context. It can be used by other phones by following the below:
; mailbox=1234@SIP_Remote
;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the "local" address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone "inside" or "outside" of the NATted network.
;   This is configured by assigning the "localnet" parameter with a list
;   of network addresses that are considered "inside" of the NATted network.
;   IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
;   Multiple entries are allowed, e.g. a reasonable set is the following:
;
;      localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
;      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 "externally visible" address and port number to be used when talking
;   to a host outside the NAT. This information is derived by one of the
;   following (mutually exclusive) config file parameters:
;
;   a. "externaddr = hostname[:port]" specifies a static address[:port] to
;      be used in SIP and SDP messages.
;      The hostname is looked up only once, when [re]loading sip.conf .
;      If a port number is not present, use the port specified in the "udpbindaddr"
;      (which is not guaranteed to work correctly, because a NAT box might remap the
;      port number as well as the address).
;      This approach can be useful if you have a NAT device where you can
;      configure the mapping statically. Examples:
;
;        externaddr = 12.34.56.78          ; use this address.
;        externaddr = 12.34.56.78:9900     ; use this address and port.
;        externaddr = mynat.my.org:12600   ; Public address of my nat box.
;        externtcpport = 9900   ; The externally mapped tcp port, when Asterisk is behind a static NAT or PAT. 
;                               ; externtcpport will default to the externaddr or externhost port if either one is set. 
;        externtlsport = 12600  ; The externally mapped tls port, when Asterisk is behind a static NAT or PAT.
;                               ; externtlsport port will default to the RFC designated port of 5061.	
;
;   b. "externhost = hostname[:port]" is similar to "externaddr" except
;      that the hostname is looked up every "externrefresh" seconds
;      (default 10s). This can be useful when your NAT device lets you choose
;      the port mapping, but the IP address is dynamic.
;      Beware, you might suffer from service disruption when the name server
;      resolution fails. Examples:
;
;        externhost=foo.dyndns.net       ; refreshed periodically
;        externrefresh=180               ; change the refresh interval
;
;   Note that at the moment all these mechanism work only for the SIP socket.
;   The IP address discovered with externaddr/externhost is reused for
;   media sessions as well, but the port numbers are not remapped so you
;   may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externaddr" and
; "externhost" might not help you configure addresses properly.
;
; NOTE 2: when using "externaddr" or "externhost", the address part is
; also used as the external address for media sessions. Thus, the port
; information in the SDP may be wrong!
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ' settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
;        nat = no                ; Use rport if the remote side says to use it.
;        nat = force_rport       ; Force rport to always be on. (default)
;        nat = yes               ; Force rport to always be on and perform comedia RTP handling.
;        nat = comedia           ; Use rport if the remote side says to use it and perform comedia RTP handling.
;
; 'comedia RTP handling' refers to the technique of sending RTP to the port that the
; the other endpoint's RTP arrived from, and means 'connection-oriented media'. This is
; only partially related to RFC 4145 which was referred to as COMEDIA while it was in
; draft form. This method is used to accomodate endpoints that may be located behind
; NAT devices, and as such the port number they tell Asterisk to send RTP packets to
; for their media streams is not actual port number that will be used on the nearer
; side of the NAT.
;
; IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.
;
; In addition to these settings, Asterisk *always* uses 'symmetric RTP' mode as defined by
; RFC 4961; Asterisk will always send RTP packets from the same port number it expects
; to receive them on.
;
; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
;
; media_address = 172.16.42.1
;
; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
; perceived external network address has changed.  When the stun_monitor is installed and
; configured, chan_sip will renew all outbound registrations when the monitor detects any sort
; of network change has occurred. By default this option is enabled, but only takes effect once
; res_stun_monitor is configured.  If res_stun_monitor is enabled and you wish to not
; generate all outbound registrations on a network change, use the option below to disable
; this feature.
;
; subscribe_network_change_event = yes ; on by default

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams 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 well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
;
;directmedia=yes                ; Asterisk by default tries to redirect the
                                ; 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.

;directmedia=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).

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

;directmedia=outgoing           ; When sending directmedia reinvites, do not send an immediate
                                ; 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'.

;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 directmedia is enabled when
                                ; the device is actually behind NAT.

;directmediadeny=0.0.0.0/0      ; Use directmediapermit and directmediadeny to restrict 
;directmediapermit=172.16.0.0/16; which peers should be able to pass directmedia to each other
                                ; (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.

;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
                                ; 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.

;sdpsession=Asterisk PBX        ; Allows you to change the SDP session name string, (s=)
                                ; Like the useragent parameter, the default user agent string
                                ; also contains the Asterisk version.
;sdpowner=root                  ; Allows you to change the username field in the SDP owner string, (o=)
                                ; This field MUST NOT contain spaces
;encryption=no                  ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
                                ; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
                                ; the peer does not support SRTP. Defaults to no.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
;
;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'. Note: realtime peers will
                                ; probably not function across reloads in the way that you expect, if
                                ; you turn this option off.
;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.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; 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.

;------------------------------ Advice of Charge CONFIGURATION --------------------------
; snom_aoc_enabled = yes;     ; This options turns on and off support for sending AOC-D and
                              ; AOC-E to snom endpoints.  This option can be used both in the
                              ; peer and global scope.  The default for this option is off.


;------------------------------ 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.

; jbtargetextra = 40          ; This option only affects the jb when 'jbimpl = adaptive' is set.
                              ; 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.

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

;----------------------------- SIP_CAUSE reporting ---------------------------------
; storesipcause = no          ; This option causes chan_sip to set the
			      ; 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".

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

[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:این آدرس ایمیل توسط  spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm

;------------------------------------------------------------------------------
; DEVICE CONFIGURATION
;
; SIP entities have a 'type' which determines their roles within Asterisk.
; * For entities with 'type=peer':
;   Peers handle both inbound and outbound calls and are matched by ip/port, so for
;   The case of incoming calls from the peer, the IP address must match in order for
;   The invitation to work. This means calls made from either direction won't work if
;   The peer is unregistered while host=dynamic or if the host is otherise not set to
;   the correct IP of the sender.
; * For entities with 'type=user':
;   Asterisk users handle inbound calls only (meaning they call Asterisk, Asterisk can't
;   call them) and are matched by their authorization information (authname and secret).
;   Asterisk doesn't rely on their IP and will accept calls regardless of the host setting
;   as long as the incoming SIP invite authorizes successfully.
; * For entities with 'type=friend':
;   Asterisk will create the entity as both a friend and a peer. Asterisk will accept
;   calls from friends like it would for users, requiring only that the authorization
;   matches rather than the IP address. Since it is also a peer, a friend entity can
;   be called as long as its IP is known to Asterisk. In the case of host=dynamic,
;   this means it is necessary for the entity to register before Asterisk can call it.
; 
; Use remotesecret for outbound authentication, and secret for authenticating
; inbound requests. For historical reasons, if no remotesecret is supplied for an
; outbound registration or call, the secret will be used. 
;
; 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
;
; Configuration options available
; --------------------
; context
; callingpres
; permit
; deny
; secret
; md5secret
; remotesecret
; transport
; dtmfmode
; directmedia
; nat
; callgroup
; pickupgroup
; language
; allow
; disallow
; autoframing
; insecure
; trustrpid
; trust_id_outbound
; progressinband
; promiscredir
; useclientcode
; accountcode
; setvar
; callerid
; amaflags
; callcounter
; busylevel
; allowoverlap
; allowsubscribe
; allowtransfer
; ignoresdpversion
; subscribecontext
; template
; videosupport
; maxcallbitrate
; rfc2833compensate
; mailbox
; session-timers
; session-expires
; session-minse
; session-refresher
; t38pt_usertpsource
; regexten
; fromdomain
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; timert1
; timerb
; qualifyfreq
; t38pt_usertpsource
; contactpermit         ; Limit what a host may register as (a neat trick
; contactdeny           ; is to register at the same IP as a SIP provider,
;                       ; then call oneself, and get redirected to that
;                       ; same location).
; directmediapermit
; directmediadeny
; unsolicited_mailbox
; use_q850_reason
; maxforwards
; encryption

;[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
;remotesecret=guessit             ; Our password to their service
;defaultuser=yourusername         ; Authentication user for outbound proxies
;fromuser=yourusername            ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;transport=udp,tcp                ; This sets the default transport type to udp for outgoing, and will
;                                 ; accept both tcp and udp. The default transport type is only used for
;                                 ; outbound messages until a Registration takes place.  During the
;                                 ; peer Registration the transport type may change to another supported
;                                 ; type if the peer requests so.

;usereqphone=yes                  ; This provider requires ";user=phone" on URI
;callcounter=yes                  ; Enable call counter
;busylevel=2                      ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain  ; send outbound signaling to this proxy, not directly to the peer
;port=80                          ; The port number we want to connect to on the remote side
                                  ; Also used as "defaultport" in combination with "defaultip" settings

;--- sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
;fromuser=4015552299              ; how your provider knows you
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
;secret=gissadetdu                ; The password they use to contact us
;callbackextension=123            ; Register with this server and require calls coming back to this extension
;transport=udp,tcp                ; This sets the transport type to udp for outgoing, and will
;                                 ;   accept both tcp and udp. Default is udp. The first transport
;                                 ;   listed will always be used for outgoing connections.
;unsolicited_mailbox=4015552299   ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
;                                 ;   message count will be stored in the configured virtual mailbox. It can be used
;                                 ;   by any device supporting MWI by specifying <configured value>@SIP_Remote as the
;                                 ;   mailbox.

;
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend

[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic

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

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw

; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
;        secret = peekaboo
; [2134](natted-phone,ulaw-phone)
;        secret = not_very_secret
; [2136](public-phone,ulaw-phone)
;        secret = not_very_secret_either
; ...
;

; Standard configurations not using templates look like this:
;
;[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
;directmedia=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 (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.
                                 ;
;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 function CALLERPRES documentation for possible
                                 ; values.

;[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
;directmedia=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
;registertrying=yes              ; Send a 100 Trying when the device registers.

;[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
;defaultuser=polly               ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
                                 ; 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
;qualifyfreq=60                  ; Qualification: How often to check for the
                                 ; host to be up in seconds
                                 ; Set to low value if you use low timeout for
                                 ; NAT of UDP sessions
;
; 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
;permit=192.168.0.60/24          ; we can also use CIDR notation for subnet masks
;permit=2001:db8::/32            ; IPv6 ACLs can be specified if desired. IPv6 ACLs
                                 ; apply only to IPv6 addresses, and IPv4 ACLs apply
                                 ; only to IPv4 addresses.

;[cisco1]
;type=friend
;secret=blah
;qualify=200                     ; Qualify peer is no more than 200ms away
;host=dynamic                    ; This device registers with us
;directmedia=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
;defaultuser=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 or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep   ; This channel variable will
                                                ; cause the given audio file to
                                                ; be played upon completion of
                                                ; an attended transfer.

;[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.
;t38pt_usertpsource=yes         ; Use the source IP address of RTP as the destination IP address for UDPTL packets
                                ; 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

QSIG-ISDN Timers

Timeout/Timer Settings - Timer Definitions

T301

Started when an ISDN call enters the alerting state (i.e.receive an ALERT message) and if it expires clears the call. Prevents the call from ringing forever.

T302

Specifies the time between digits in overlapped dialing.  If it expires it is assumed all digits have been received.

T303

Specifies the time after sending a ISDN SETUP message to wait for a response. First expiration will resend the SETUP and second expiration will fail the call.

T305

Specifies the time to wait after sending an ISDN DISCONNECT message to wait for the RELEASE message response. If it expires we then send an ISDN RELEASE message.

T308

Specifies the time to wait after sending an ISDN RELEASE message to wait for the RELEASE COMPLETE message response. First expiration will resend the RELEASE message and second expiration will send a RELEASE COMPLETE message.

T309

Specifies the time to wait after the D channel goes down for B channels with calls in the active state to wait for the D channel to come back up. If it expires and the D channel has not come back up the call on the B channel is cleared.

T310

Specifies the time to wait after receiving a CALL PROCEEDING message to wait for some form of call progress, i.e. a PROGRESS, ALERT or CONNECT message. If it expires the call is cleared.

T313

Specifies the time for user side ISDN to wait after sending a CONNECT message to wait for the CONNECT ACKNOWLEDGE message. If it expires then the call is cleared.

T314

Specifies the time to wait between SEGMENT messages for segmented message support in QSIG. If it expires all previously received SEGMENT messages are discarded and ignored.

T316

Specifies the time to wait after sending a RESTART message to wait for the RESTART ACKNOWLEDGE. First expiration will resend the RESTART message and second expiration will put the channel in local maintenance state for protocols that support service messaging. For protocols that don't support service messaging channel will be returned to the idle state.

T322

Specifies the time to wait after sending a STATUS ENQUIRY message to get the STATUS message response. First expiration will resent the STATUS ENQUIRY message and second expiration will clear an active call. Only started if T309 doesn't expire and the D channel comes back up before T309 expires on B channels with active calls.

T3M1(T323)

Specifies the time to wait after sending a SERVICE message to get the SERVICE ACKNOWLEDGE response. If it expires the SERVICE message will be resent. For NI-2 this is a maximum of 3 attempts before leaving the B channel in the state it was before and not changing its state.

نحوه دریافت SIP Trunk

سرويس SIP TRUNK

 شرح سرویس SIP TRUNK

در نسل جدید شبکه های مخابراتی (Next Genaration Network) NGN (شبکه های نسل آینده) که کلیه ارتباطات بر مبنای IP می باشد، جهت ارتباط بین مراکز تلفنی و همچنین ارتباط مراکز تلفن خصوصی PBX از ترانکهای بسته ای ( PACKET TRUNK) استفاده می شود که نوعی از آن SIP TRUNK می باشد و جایگزین ارتباطات PRI و E۱ در شبکه های TDM است. مشترکانی که می خواهند از سرویس های متنوع VOIP برخوردار شوند می توانند از این سرویس بهره بگیرند.
از این لینک می توان برای انتقال Voice برروی IP استفاده کرد.
جهت اتصال مرکز تلفن بخش خصوصی با ورودی دیتا به شبکه مخابراتی به جای PRI  يا  E۱ از لینک های مبتنی بر دیتا نظیر ADSL یا G.SHDSL با پروتکل SIP Trunkاستفاده می شود.
 

  مزایای سرویس

- امکان واگذاری گستره متنوعی از کانال های همزمان و نامبرینگ متناظر با آن بنا به درخواست متقاضی. به طور مثال: تعداد ۱۰ ، ۲۰ ، ۱۵۰ و یا ۱۰۰۰ کانال
- قابل استفاده بر روی کابل مسی و فیبر نوری
- عدم نیاز به افزایش تعداد تجهیزات سمت متقاضی با افزایش تعدادشماره (مثلا ۱۲۰ شماره تلفن با فقط یک مودم قابل سرویس دهی است)
- عدم وجود محدودیت جغرافیایی برای ارائه سرویس
- کاهش هزینه تجهیزات مورد نیاز
- امکان مدیریت ترافیک صوتی ارسالی بر روی لینک به دلیل استفاده از بسته های IP
 

ویژگی های سرویس

- تجهیزات نهایی سمت متقاضی باید پروتکل SIP وIP را پشتیبانی نماید .
- متقاضی سرویس نیازمند خریداری مودم های G.ShDSL.Bis می باشد.
- استفاده از پهنای باند اینترانتی به میزان مورد نیاز جهت مکالمه همزمان و عدم دایری اینترنت بر روی پورت ADSL یا G.SHDSL
 

 مدارک مورد نیاز :

- اساسنامه شرکت
- روزنامه رسمی تاسیس و تغییرات شرکت که بیش از دوسال از تاریخ آن نگذشته باشد
- فرم کد اقتصادی
- معرفی نامه نماینده شرکت
- نامه درخواست ممهور به مهر و امضا مدیر عامل و دارندگان امضا شرکت
- اصل و کپی و کارت ملی و شناسنامه نماینده شرکت
- تکمیل فرم کارت اشتراک و فرم پرسشنامه
محل ثبت نام :


- متقاضیان حقوقی دریافت لینک فوق  به نزدیک ترین منطقه مخابراتی مربوطه خود مراجعه نمایند.

نام مرکز     نشانی مرکز                                        شماره تماس مرکز                        شماره تماس رئیس مرکز
شهید باهنر خ شریعتی-بالاتراز دوراهی قلهک جنب سینما فرهنگ 22623310 22601111
استقلال خ ولیعصر -بالاتر از صدا و سیما -نرسیده به بزرگراه چمران - خ استقلال 22662217-22662215 22041111
پاسداران خ پاسداران –بهستان هشتم 22778771-22778751 22777011
جماران پاسداران- سه راه اقدسیه - مرکز مخابرات جماران 26114080 26114051
شهید تندگویان بزرگراه رسالت - خ استادحسن بنا شمالی روبروی ندارایانه 26319978-26319973 26319990-26319991
شهید جعفری خ ولنجک نبش خ 17 22401111 22181509
شهید دستغیب میدان قدس - اول دربند - نبش کوچه نخشب 22756850-22756851 22756801-22756802
شهید قدوسی خ ظفر نبش خ نفت شمالی روبروی بانک ملی 22221330-22222864 22222103-22921440
شهید غریبی بزرگراه آیت ا... صدر - قیطریه شمالی بلوار صبا نبش کوچه کتابی 22696155 22696188-22696156
سلمان فارسی سعادت آباد بین میدان کاج و قیصر امین پور 22142170-22142178 22351111
شهید محلاتی مینی سیتی - بزرگراه ارتش شهرک شهید محلاتی خ توحید مرکز مخابرات شهید محلاتی 22469040 22441111
شهید مفتح اول خیابان پاسداران نرسیده به سه راه ضرابخانه ک کوکب مرکز مخابرات شهید مفتح 22893905 22893900
شهید لطیفی لویزان -خ شعبانلو -روبروی پارک سخندان -جنب کوچه نسترن 22980074-22980034 22980015-22980016
شهدای لواسان لواسان کوچک میدان امام خمینی (ره) 26549550 26542222
ایراء کم ظرفیت روستای ایراء 26543420 26542222
لواسان بزرگ کم ظرفیت روستای لواسان بزرگ 26543420 26542222
کلان کم ظرفیت روستای کلان 26543420 26542222
نیکنامده کم ظرفیت روستای نیکنامده 26543420 26542222
برگ جهان کم ظرفیت روستای برگ جهان 26543420 26542222
افجه کم ظرفیت روستای افجه 26543420 26542222
سینک کم ظرفیت روستای سینک 26543420 26542222
کند کم ظرفیت روستای کند 26543420 26542222
بوجان کم ظرفیت روستای بوجان 26543420 26542222
راحت آباد کم ظرفیت روستای راحت آباد 26543420 26542222
امام علی فشم رودبار قصران میدان فشم 26500372 26500000
شمشک کم ظرفیت روستای شمشک 26500372 26500000
دربندسر کم ظرفیت روستای در بندسر 26500372 26500000
میگون کم ظرفیت روستای میگون 26500372 26500000
گرمابدر کم ظرفیت روستای گرمابدر 26500372 26500000
زایگان کم ظرفیتروستای زایگان 26500372 26500000
لالان کم ظرفیت روستای لالان 26500372 26500000
روته کم ظرفیت روستای روته 26500372 26500000
امامه کم ظرفیت روستای امامه 26500372 26500000
آهار کم ظرفیت روستای آهار 26500372 26500000
ایگل کم ظرفیت روستای ایگل 26500372 26500000
اوشان کم ظرفیت روستای اوشان 26500372 26500000
کلوگان کم ظرفیت روستای کلوگان 26500372 26500000
رودک کم ظرفیت روستای رودک 26500372 26500000
زردبند کم ظرفیت روستای زردبند 26500372 26500000
ملت خ سعدي جنوبي خ اكباتان 9و33978407 33978403
شهيد ديالمه خ ري نبش كوچه اربابي جنب بيمارستان شهيد اندرزگو 33134743, 33542099 33545002
پيروزي خ پيروزي سه راه سليمانيه ، خ بيمارستان فجر ، پلاك 2 33253002, 33253007 33253010
شهید صالح طبری شهرک اکباتان – فاز1- سه راه مخابرات 44632080-44632060 44641111
شهید ثابت قدم کیلومتر 12 مخصوص کرج – شهرک دانشگاه 44909010 44901111
شهدای ورد اورد کیلومتر 19 مخصوص کرج- خیابان ازادی 44982727 44992020 44991111 44996666
شهید یزدان پناه انتهای اشرفی اصفهانی –بلوار سیمون بولیوار – خیابان مخابرات 44850014 44850015 44801111
شهدای چیتگر کیلومتر 14 مخصوص کرج – چهارراه ایران خودرو – چیتگر شمالی – خیابان جهاد- خیابان بدر 44190045 44191111
شهید کاظمیان کوهسار- جنب آتش نشانی 44350460 44301111
سولقان جاده امامزاده داود-روستای سولقان 44393400 44301111
کشار جاده امامزاده داود-روستای سولقان 44393400 44301111
رندان جاده امامزاده داود-روستای سولقان 44393400 44301111
سنگان جاده امامزاده داود-روستای سولقان 44393400 44301111
امامزاده داود جاده امامزاده داود-روستای سولقان 44393400 44301111
کیگا جاده امامزاده داود-روستای سولقان 44393400 44301111
وردیج وواریش جاده امامزاده داود-روستای سولقان 44393400 44301111
شهید توکلی تهرانسر – بلوار لاله – خیابان قباد شمالی 44569093-8 44569093-8
شهید سعادتمند خیابان مرزداران – خیابان شهید ابراهیمی –روبروی برج الوند 44211516 44211516
شهید حق شناس شهرک گلستان – بلوار امیرکبیر- بلوار گلها- بنفشه 7 44712014 44712014
شهید زارعی شهر زیبا – بلوار تعاون – بلوار آلاله – خ عدالت – روبروی مسجد امام رضا 44107004-44105050 44107004-44105050
شهید ایت ا... کاشانی بلوار فردوس- نبش خیابان سلیمی جهرمی 44083121-44963396 44083121-44963396
پیام نور جنت اباد- 35 متری گلستان – خیابان ایران زمین جنوبی – خیابان حیدری مقدم شرقی 44482022-3 44482022-3
شهدای شهر قدس هفت جوی زرنان شهرستان قدس – بلوار شهید قربانی 46822220-46892222-46838000 46822220-46892222-46838000
مهدیه خ ولیعصر جنب مهدیه تهران مرکز مخابرات مهدیه 55390019-55393370-55392241-55488300 55391111
شهیدقندی خیابان خیام شمالی بالاتر از میدان محمدیه 55589963 55811111
شهید بختیاری جاده قدیم قم بلوار الغدیر نرسیده به بهشت زهرا باقرشهر مرکز مخابرات شهید بختیاری 55201212 55204444
بعثت بزرگراه بعثت ، جنب ترمینال جنوب 55092222-55333030 55061111
شهید نواب صفوی میدان رازی خ قزوین روبروی شرکت دخانیات مرکز مخابرات 55425020-55425030-55425090-55425002-55425010 55411515-55402222
شهید خوشقدم بزرگراه آیت اله سعیدی م سروری بزرگراه چراغی (جوانه سابق) جنب خ ورزش 55883398 55821111
سیدالشهداء(ع) خیابان قزوین ، دوراهی قپان ، خیابا ن قاسم فرهنگ 55171701-55171702-55171703-55171704-55171705-55171706-55171708-55171709-55171710 55701111
سیزده آبان خ رجایی ـ انتهای خ ستاره ـ شهرک وصال ـ مرکز مخابرات سیزده آبان 55539980-7 55001313
آیت اله ایروانی ج ساوه ـ چهاردانگه ـ روبروی آتش نشانی ـ مرکز ایروانی 55281128 55241111-55242222
شهید منتظری شهرری جاده قم خیابان شهید غیبی پلاک 1 55931600 - 55931700 55951884 - 55961900 55901111
شهید سلیمانی(حسن آباد) حسن آباد -بلوارامام خمینی روبروی کلانتری مرکز مخابرات شهیدسلیمانی 56223117-56223112 56222222
شمس آباد حسن آباد-شهرک صنعتی شمس آباد بلوار گلستان گلشن 10 56223117-56223112 56222222
ابراهیم آباد حسن آباد-روستای ابراهیم آباد ابتدای روستا 56223117-56223112 56222222
کهریزک و قاسم آباد جاده قدیم قم ـ بعد از بهشت زهرا ـ قاسم آباد ـ 60بعد ازمتری جنب درمانگاه حضرت ابوالفضل (ع) 56541200ــــ56541000 56544444
شهید رمضانی میدان رازی - خ رباط کریم اصلی - بعد از چهارراه انبار نفت - مرکز مخابرات شهید رمضانی 55643377 "55651111-55645555 "
قندي اسلامشهر بلواربسيج مستضعفين خ قدس رضوي مركزمخابرات 56141269-56141568 56121111-56131111
بهشتي اسلامشهر بلواربسيج مستضعفين ابتداي خ امام محمدباقر(ع) روبروی پاسگاه 56379400-56379401 56351111
باهنر اسلامشهر شهرك قائميه بلوار خليج فارس مركزمخابرات 56492040-56492050 56461111-56471111
رجايي شهرك واوان خ امام خميني روبروي خ گلها مركزمخابرات 56162161-56162171 56171111-56161111
ميان آباد اسلامشهر شهرك ميان آباد ابتداي شهرك مركزمخابرات 56552636 56551111
احمدآباد مستوفي احمدآباد مستوفي ميدان مشعل جنب دفتر پست مركزمخابرات 56718300 56717070-56712222
حسن آباد اسلامشهر-روستاي حسن آباد(مركزاحمدآباد مستوفي) 56718300 56717070-56712222
فيروزبهرام اسلامشهرروستاي فيروزبهرام(مركزاحمدآباد مستوفي) 56718300 56717070-56712222
انبيا اسلامشهر شهرك انبيا خ جانبازان مركزمخابرات 56833060-56835001 56363010-56831111
ايرين اسلامشهر روستاي ايرين (مركزمخابرات انبيا) 56833060-56835001 56363010-56831111
چيچكلو اسلامشهر روستاي چيچكلو (مركزمخابرات انبيا) 56833060-56835001 56363010-56831111
علي آباد تپانچه اسلامشهر روستاي علي آباد (مركزمخابرات باهنر) 56492040-56492050 56461111-56471111
نسیم شهر نسیم شهر - انتهای خ امام خمینی نرسیده به میدان شهید رجایی مرکز مخابرات نسیم شهر 56753003 56754444
مهرچین ملارد مهرچین خ تعاون مرکز مخابرات 65581111  
صالح آباد صالح آباد- خ شهید باهنر - روبروی آبفا - مرکزمخابرات صالح آباد 56627777 56622222
قلعه میر قلعه میر - خ اصلی روبروی آتش نشانی-مرکز مخابرات قلعه میر 56862121 56862222
گلستان جاده ساوه- ابتدای گلستان - روبروی پمپ بنزین - مرکز مخابرات گلستان 56322700 56322222
اورین اتوبان ساوه - خروجی آدران -اورین - مرکز مخابرات اورین 56573000 56573333
خیرآباد نسیم شهر - خ خیرآباد - مرکز مخابرات خیرآباد 56646200 56646200
باهنر رباط کریم - بلوار امام خمینی نرسیده به پلیس راه - مرکز مخابرات باهنر 56422121 56423000
پرند پرند - میدان استقلال خ جمهوری نبش خ کوهستان مرکز مخابرات پرند 56720015 56720017
الارد رباط کریم - جاده شهریار - الارد - مرکز مخابرات الارد 56422121 56423000
منجیل آباد رباط کریم- جاده شهریار- روستای منجیل آباد/ مرک مخابرات منجیل آباد 56422121 56423000
انجم آباد رباط کریم - جاده شهریار - انجم آباد - مرکز مخابرات انجم آباد 56422121 56423000
محمدیه رباط کریم - محمدیه - خ شهید بهشتی -مرکز مخابرات محمدیه 56422121 56423000
وهن آباد رباط کریم -میدان امام خمینی - به طرف اتوبان قم - وهن آباد - مرکز مخابرات وهن آباد 56422121 56423000
شهر صنعتی پرند رباط کریم - به طرف ساوه - شهرصنعتی پرند - مرکز مخابرات شهرصنعتی پرند 56720015 56720017
شهرک خانه رباط کریم - شهرک خانه - خیابان سوسن -مخابرات شهرک خانه 56720015 56720017
منطقه مسکونی پرند (23) رباط کریم - پادگان لشگر 23- مرکز مخابرات 56720015 56720017
گروه 33 رباط کریم - گروه 33 توپخانه - مرکز مخابرات 56720015 56720017
نصیرآباد قاجار رباط کریم - به طرف گلستان - نصیرآباد - مرکز مخابرات نصیرآباد 56656000 56651000
آدران رباط کریم - جاده ساوه- جاده ده حسن - آدران - مرکز مخابرات آدران 56584200 56322222
شهر صنعتی نصیر آباد رباط کریم - جاده ساوه - بعد از پل راه آهن - سمت راست شهر صنعتی نصیرآباد 56656000 56322222
صفادشت صفادشت -خ اصلی قبل ازشهرداری 65433000 65435000
یوسف آباد ملارد یوسف آبادقوام چهارراه یوسف اباد مرکزمخابرات 65583333 _
خوشنام ملارد خوشنام خ اصلی روبروی مدرسه دخترانه کاشانی پور 65473333  
بیدگنه ملارد بیدگنه ابتدای بلوار امام 65721111  
امیرابادتوانبخشی ملارد امیرآباد میدان شهدا نبش بلوارمرکز مخابرات 65741111  
اختراباد چهارراه بی بی سکینه بعداز پلیس راه به سمت اخترآبادروستای اخترآبادخ پیام مرکز مخابرات 65943333  
ارغش آباد ملارد چهارراه بی بی سکینه ارغش آباد جنب بهداری 65990000  
توحید خ اسکندری شمالی - انتهای کوچه خازنی 66428010-66576313 66941111
ابوذر خیابان شمشیری بعدازسه راه بوتان 66694444-66634444-66624444 66691111
ولیعصر(عج)-تهران م سپاه -خ خواجه نصیرالدین طوسی - خ اجاره دار 77500751 77522222
مركز مخابرات وليعصر بومهن بومهن - خيابان اصلي امام خميني (ره) - روبروي بانك كشاورزي 76225113-76225120 76224000
شهيد اكبري تهرانپارس - خ جشنواره - خ زهدي - خ ملكي 77796200 77797001
شهید حکمت شعار خیابان فرجام شرقی خیابان 41 77181919-77222060-77443040-77228055 - 77228044 - 77444444
شهید آیت خ تهران نو - خ وحيديه -ميدان تسليحات 7282810-5 - 77827000 77282801-3
مخابرات امامت سی متری نیروی هوائی - نبش ک 6/33 77436060 77418001
پردیس پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510- 76243070 76245001
کمرد صنعتی پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510 -76243070 76245001
جاجرود پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510 -76243070 76245001
خرمدشت پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510- 76243070 76245001
باغ کمش پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510 -76243070 76245001
واصفجان پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510 - 76243070 76245001
مرکز شهدای فیروزکوه و توابع فیروزکوه خیابان پاسداران روبروی بانک رفاه مرکز تلفن شهدای فیروزکوه 76402080 - 76402090 - 76443531 76442222 - 76443000
مخابرات عاطف بالاتر از فلکه دوم تهران پارس - مرکزمخابرات شهید عاطف 77733324 - 77702024 - 77886666 77861111
آزادگان خ شهیدبهشتی میدان تختی ابتدای خ صابونچی ک شهید ادایی (دوم ) انتهای خیابان 88531041-88522930 88731111
مرکز مخابرات شهید آل اسحاق تهرانپارس - حکیمیه - خیابان بهار - بلوار معراج 77009708-77309966 77309999
شهدای گمنام فلکه 4 تهرانپارس خیابان توحید کوچه 4 شرقی 77365151-77081005 77361111
رودهن 76513525 76513525 76502222
شهید بابایی اتوبان بابایی جنب دانشگاه امام حسین مرکز مخابرات شهیدبابایی 77106026 -77101819- 77106027 77101111
نبوت خيابان آيت - چهار راه تلفنخانه 77948400 - 77948200 - 77948100 - 77948300 77931111
شهید نظری خ ش مدنی (نظام آباد)، ایستگاه کهن 77541515 77575711
فجر دماوند دماوند- میدان 17 شهریور- اداره مخابرات دماوند 76325300 76322222
شهید بهشتی گیلاوند گیلاوند- بلوار آیت ا... خامنه ای- مرکز مخابرات شهید بهشتی گیلاوند 76311001 76314444
شهدای هفتم تیر خ مفتح جنوبی خ طهمورث نبش خ تور پ 8 88310551 88311111
شهید مدنی بلوار کشاورز خ کبکانیان نبش خ غفاری 88990021 88961111
شهید رجایی خ کریم خان انتهای ضلع غربی پارک بهجت آباد خ شهید گلابی 88906114 88901111
قدس خ کارگرشمالی روبروی خیابان 14 (شهید عزیزی ) پ 1458 88014950-8801033 88015010-88010166 88015016 88001111
شهید بهشتی خ قدوسی - نبش خ امجدی منش 88408020 88401111
شهید مطهری خ ولیعصر نرسیده به خ شهید مطهری جنب بیمارستان هاجر 88706913-88717711 88701111
شهید نصرالهی بزرگراه جلال آل احمد جنب آتش نشانی خ شهید ناظریان قمی 88272744-88257121 88271111
مرحوم نهری شهرک قدس (غرب) فاز 2 خ هرمزان بین خ 5 و7 88372737-88090066 88081111
شیخ فضل اله خ ملاصدرا - خ شیخ بهایی جنوبی جنب آتش نشانی 88219100-88617050 88031111
الغدير ميدان خراسان خ خاوران خ شهيد مينائي 33696930-1 33696920
هجرت خ افسريه خ هفدهم 42و41و40و33646339 33645700
شهيد اشرفي‌اصفهاني خ پيروزي انتهاي خ نبرد جنب تاكسيراني 33079702, 33079703 33079700, 33062040
شهيد درزي كيانشهر ميدان امام رضا (ع) 33608040, 33608050 33888815
شهيد عربسرخي فلكه اول دولت آباد خ ش احمد علي نواز نرسيده به صفائيه 33776250-1 33776177
شهيد مشهدي مهدي خ خاوران بالاتر از ميدان آقانور 33482081, 33482085 33482018, 33491111
شهيد نبوي جاده ورامين ، سه راه تقي‌آباد سمت چپ جاده 33411050, 33411090 33407040
سعدي خ سعدي شمالي چهارراه سعدي خ ش مرادي‌پور   33979348, 77682867
شهيد باكري بزرگراه بسيج (اتوبان افسريه) سه راه تختي بلوار هجرت بالاتر از ميدان ش باكري 39-33200038 33200035
قيامدشت جاده خاوران شهرك قيامدشت چهاراه مخابرات مركز ش كريم‌لو 33581033, 33582066 33582033
خاورشهر جاده خاوران ، خاورشهر فاز 1 خ شهيد عدالت پژوه كوي مخابرات 33857223, 33857600 33855200
مرصاد   36016096 36016030
پيشوا پيشوا جنب سه راه شهرداري 36723668, 36733668 36722222
دام گستر پلیس راه حسن آباد کیلومتر10به سمت اتوبان تهران قم مجتمع دام گستر 56223117-56223112 56222222
قلعه محمدعلیخان حسن آباد جاده قدیم تهران قم کیلومتر 30روستای قلعه محمدعلیخان 56223117-56223112 56222222
سلمان آباد حسن آباد روستای سلمان آباد مرکز مخابرات 56223117-56223112 56222222
بیجین پلیس راه حسن آباد-کیلومتر 10جاده چرمشهر ناحیه صنعتی بیجین مرکز مخابرات 56223117-56223112 56222222
خانلق حسن آباد-خ شیخ کلینی روستای خانلق ابتدای روستا 56223117-56223112 56222222
ورآباد چهارراه بی بی سکینه بعداز پلیس راه قطعه 4بعدازامیراباد قشلاق بحر خ اصلی 65945000  
امیرابادقشلاق چهارراه بی بی سکینه بعداز پلیس راه قطعه 4بعدازفدک امیراباد قشلاق بحر خ اصلی 65947000  
قره ترپاق چهارراه بی بی سکینه بعداز پلیس راه قطعه 4بعدازورآبادبحر خ اصلی 65948000  
محمودآباد ملارد بیدگنه به سمت قطعه چهارراه روستای محمودآباد 65943333  
اله آباد ملارد بیدگنه به سمت قطعه چهارراه بعداز محمودآباد روستای اله آباد 65943333  
مركز مخابرات پرديس2 پرديس -فاز2-ميدان امام -جنب درمانگاه كوثر 76243070-76247510 76245001
کمرد روستا پردیس- فاز 2- میدان امام خمینی (ره)- بلوار صیاد شیرازی- مرکز مخابرات پردیس 76247510- 76243070 76245001
شهید کلانتری خ گاندی نبش خ 17 88799030 88881111
ورامين ورامين ميدان امام حسين (ع) خ شهيد بهشتي خ معلم خ دولت 36245173, 36245178 36242222
شهيد مومن ورامين ميدان امام حسين (ع) خ شهيد بهشتي خ معلم خ دولت 36273007 36273007
خيرآباد ورامين ميدان امام حسين (ع) خ شهيد بهشتي خ معلم خ دولت 36235178 36232222
قرچك قرچك خ اصلي نبش خ آزادگان 36126050 36144444
باقرآباد (15 خرداد) قرچك خ اصلي نبش خ آزادگان 2-361660051 36132222
شهيد بهشتي قرچك خ اصلي نبش خ آزادگان   36125555
شهيد باهنر قرچك خ اصلي نبش خ آزادگان   36171111
شهید کرمی شهرک ولیعصر - خیابان شهید آقائی جنوبی - 18 متری یاسر - 12 متری شهید کاکلی 66203333-66213333-66223550-66223660 66201111-66291111
شهید بخششی و شهدای دانش جاده قدیم کرج خیابان خلیج خ شهید بهشتی 66254747-66277600 66251111
مركز مخابرات حافظ خ- جمهوري- تقاطع حافظ - روبروي ساختمان بورس كوجه سيمي 66759504 66759511
شهریار شهریار خ ولی عصر (عج) خ ش بهشتی اداره مخابرات شهدای شهریار 65224141 65240000
اندیشه1 اندیشه 1 - ابتدای خ هشتم شرقی 65522424 65524444
اندیشه 2 اندیشه 2 - بلوار نیلوفر - بلوار ارکیده 65542424 65541111
اندیشه3 اندیشه 3 - روبروی شهرداری 65553333 65551111
اندیشه4و5 اندیشه 4 - حد فاصل میدان شهدای گمنام و میدان خلیج فارس 65348141 65345555
وایین وایین - بلوار امام خمینی (ره) 65328111 65327700
امیریه برد آباد - میدان امام خمینی (ره) 65640011 65642222
وحیدیه وحیدیه - بلوار شهدا 65633222 65633001
خادم آباد جاده شهریار -تهران - ورودی خادم آباد 65235050 65234444
سعیدآباد سعید آباد - خ ش رجایی مرکز مخابرات 65602222 65605555
شاهدشهر شاهد شهر - میدان امام خمینی (ره)-خ مخابرات 65442111 65445000
فردوسیه فردوسیه -خ امام خمینی (ره) 65462222 65754444
اصیل آباد اصیل آباد - بلوار آزادگان خ طالقانی - مرکز مخابرات 65754444 65754444
بکه بکه خ حسن بیگی 65750333 65754444
حصارساتی حصارساتی - خ سعدی 65343333 65754444
کردزار کردزار خ اصلی به سمت ابراهیم آباد 65731111 65754444
یوسف آباد صیرفی یوسف آباد صیرفی - خ ش بهشتی 65772002 65754444
یبارک یبارک - خ گلها مرکز مخابرات 65452333 65754444
صباشهر صباشهر خ ش تاجیک 65625000 65256111
ویره جاده آدران 65763000 65256111
اسدآباد اسدآباد -خ اصلی 65682222 65256111
نصیرآباد بدر نصیرآباد - خ ش بهشتی 65954444 65233400
باباسلمان باباسلمان -خ ش کلهر 65733333 65233400
گلگون گلگون انتهای بلوار میلاد 65611111 65233400
دهشاد دهشاد خ مخابرات - بلوار اصلی 65962333 65233400
رزکان رزکان - خ اصلی 65933333 65233400
اسکمان اسکمان -خ اصلی 65960333 65233400
رضی آباد رضی آباد - خ مخابرات 65260222 65256111
حسن آباد خالصه   56718300 56722222

زیر مجموعه ها

ارتباطی پایدار، صدایی ماندگار، آکراتل

فراموش نکنیم تلفن اولین و بهترین راه کار ارتباطی و تجاری ست

با گذشت چند دهه از اختراع تلفن، هنوز این اختراع اولین و مهمترین و همیشگی ترین، ابزار ارتباطی کسب و کارها می باشد.