ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

Asterisk Opensource IP Pbx

ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

โพสต์โดย voip4share » 16 ธ.ค. 2009 00:02

ไฟล์ sip.conf เป็นไฟล์คอนฟิกสำหรับโปรโตคอล SIP ของ Asterisk ครับ ในไฟล์นี้มีข้อมูลเกี่ยวกับ SIP มากมายทั้งของ Asterisk เอง, ของเบอร์ Extension ที่จะมารีจิสเตอร์กับ Asterisk แบบ SIP และรายละเอียดของ Trunk ที่เป็น SIP ที่ Asterisk จะเชื่อมต่อด้วย

เนื่องจากข้อมูลในไฟล์นี้มีหลายส่วน ผมขอแบ่งออกเป็นส่วนๆดังต่อไปนี้นะครับ

1. ข้อมูลทั่วไป ซึ่ง Asterisk จะใช้สำหรับตัวมันเอง
2. ข้อมูลของเบอร์ SIP Extension ซึ่งใช้กำหนดค่า SIP พารามิเตอร์ของเบอร์ Extension
3. ข้อมูลของ SIP Trunk ซึ่งใช้กำหนดค่า SIP พารามิเตอร์ของ Trunk
4. ข้อมูล Username/Password เมื่อต้องเอา Asterisk ไปรีจิสเตอร์กับ SIP Server อื่น

ถ้าต้องการใส่หมายเหตุกำกับ หรือต้องการปิดไม่ใช่บรรทัดใดบรรทัดหนึ่ง ให้ใส่เครื่องหมายเซมิโคล่อน ; ไว้หน้าบรรทัดนะครับ

และถ้าต้องการแยกไฟล์ sip.conf ออกเป็นหลายๆไฟล์ เพื่อความสะดวกในการคอนฟิกและตรวจสอบ ก็สามารถทำได้ครับ แต่ต้องระบุชื่อไฟล์เหล่านั้นเข้าไปในไฟล์ sip.conf ด้วย เพื่อให้ Asterisk จะได้ตามข้อมูลพบ ยกตัวอย่างเช่น แยกเป็นไฟล์ sip_general.conf, sip_clients.conf, sip_trunk.conf ให้เพิ่มบรรทัดเหล่านี้เข้าไปในไฟล์ sip.conf

#include sip_general.conf
#include sip_clients.conf
#include sip_trunk.conf

เซฟไฟล์ sip.conf แล้วเข้า Asterisk Console จากนั้นพิมพ์คำสั่ง sip reload เพื่อให้มีผลครับ

มาดูรายละเอียดของแต่ละส่วนกันครับ
voip4share
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ลงทะเบียนเมื่อ: 18 พ.ย. 2009 11:26
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Re: ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

โพสต์โดย voip4share » 17 ธ.ค. 2009 12:36

1. ข้อมูลทั่วไป

จะอยู่ภายใต้ [general] ดังต่อไปนี้

[general]
; Default context for incoming calls
context=default
; Allow or reject guest calls (default is yes)
;allowguest=no
; Disable overlap dialing support. (Default is yes)
allowoverlap=no
; Disable all transfers (unless enabled in peers or users) Default is enabled
;allowtransfer=no
; 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
;realm=mydomain.tld
; UDP Port to bind to (SIP standard port is 5060) bindport is the local UDP port that Asterisk will listen on
bindport=5060
; IP address to bind to (0.0.0.0 binds to all)
bindaddr=0.0.0.0
; 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
srvlookup=yes
; Enable checking of tags in headers, international character conversions in URIs and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
;pedantic=yes

; See doc/ip-tos.txt for a description of these parameters.
; Sets TOS for SIP packets
;tos_sip=cs3
; Sets TOS for RTP audio packets.
;tos_audio=ef
; Sets TOS for RTP video packets
;tos_video=af41
; Maximum allowed time of incoming registrations and subscriptions (seconds)
;maxexpiry=3600
; Minimum length of registrations/subscriptions (default 60)
;minexpiry=60
; Default length of incoming/outgoing registration
;defaultexpiry=120
; Minimum roundtrip time for messages to monitored hosts. Defaults to 100 ms
;t1min=100
; Allow overriding of mime type in MWI NOTIFY
;notifymimetype=text/plain
; Default time between mailbox checks for peers
;checkmwi=10
; 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.
;buggymwi=no
; dialplan extension to reach mailbox sets the Message-Account in the MWI notify message defaults to "asterisk"
;vmexten=voicemail
; First disallow all codecs
;disallow=all
; Allow codecs in order of preference
;allow=ulaw
; see doc/rtp-packetization for framing options
;allow=ilbc

; 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
;
; Default language setting for all users/peers. This may also be set for individual users/peers
;language=en
; Relax dtmf handling
;relaxdtmf=yes
; If Remote-Party-ID should be trusted
;trustrpid = no
; If Remote-Party-ID should be sent
;sendrpid = yes
; 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
;progressinband=never
; Allows you to change the user agent string
;useragent=Asterisk PBX
; 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.
;promiscredir = no
; If yes, ";user=phone" is added to uri that contains a valid phone number
;usereqphone = no
; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;dtmfmode = rfc2833
; send compact sip headers.
;compactheaders = yes
;
; Turn on support for SIP video. You need to turn this on in the this section to get any video support at all. You can turn it
; off on a per peer basis if the general video support is enabled, but you can't enable it for one peer only without enabling
; in the general section.
;videosupport=yes
; Maximum bitrate for video calls (default 384 kb/s) Videosupport and maxcallbitrate is settable
;maxcallbitrate=384
; For peers and users as well generate manager events when sip ua performs events (e.g. hold)
;callevents=no
; 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
;alwaysauthreject = 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 :-(
;g726nonstandard = yes

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

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

;
; If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who
; registers or unregisters with us and have a "regexten=" configuration item. Multiple contexts may be specified by
; separating them with '&'. The actual extension is the 'regexten' parameter of the registering peer or its name if 'regexten'
; is not provided. If more than one context is provided, the context must be specified within regexten by appending the
; desired context after '@'. More than one regexten may be supplied if they are separated by '&'. Patterns may be used in
; regexten.
;
;regcontext=sipregistrations
;
;RTP timers
; These timers are currently used for both audio and video streams. The RTP timeouts are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
; 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.
;rtptimeout=60
; Terminate call if 300 seconds of no RTP or RTCP activity on the audio channel when we're on hold (must be > rtptimeout)
;rtpholdtimeout=300
; Send keepalives in the RTP stream to keep NAT open (default is off - zero)
;rtpkeepalive=<secs>

;SIP DEBUGGING
; Turn on SIP debugging by default, from the moment the channel loads this configuration
;sipdebug = yes
; Record SIP history by default (see sip history / sip no history)
;recordhistory=yes
; Dump SIP history at end of SIP dialogue SIP history is output to the DEBUG logging channel
;dumphistory=yes

;STATUS NOTIFICATIONS (SUBSCRIPTIONS)
; You can subscribe to the status of extensions with a "hint" priority (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call limit set for a device. When the call limit is filled, we
; will indicate busy. Note that you need at least 2 in order to be able to do attended transfers.
;
; For queues, you will need this level of detail in status reporting, regardless if you use SIP subscriptions. Queues
; and manager use the same internal interface for reading status information.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the realtime switch.
;
; Disable support for subscriptions. (Default is yes)
;allowsubscribe=no
; Set a specific context for SUBSCRIBE requests Useful to limit subscriptions to local extensions Settable per peer/user also
;subscribecontext = default
; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: no)
;notifyringing = 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.
;notifyhold = yes
; Apply call limits on peers only. This will improve status notification when you are using type=friend. Inbound calls, that
; really apply to the user part of a friend will now be added to and compared with the peer limit instead of applying two call
; limits, one for the peer and one for the user. "sip show inuse" will only show active calls on the
; peer side of a "type=friend" object if this setting is turned on.
;limitonpeers = yes

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

; Retry registration calls every 20 seconds (default)
;registertimeout=20
; 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
;registerattempts=10

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

; Address that we're going to put in outbound SIP messages if we're behind a NAT. The externip and localnet is used when
; registering and communicating with other proxies that we're registered with
;externip = 200.201.202.203
; Alternatively you can specify an external host, and Asterisk will perform DNS queries periodically. Not recommended for
; production environments! Use externip instead
;externhost=foo.dyndns.net
; How often to refresh externhost if used
;externrefresh=10
; You may add multiple local networks. A reasonable set of defaults are
; All RFC 1918 addresses are local networks
;localnet=192.168.0.0/255.255.0.0
; Also RFC1918
;localnet=10.0.0.0/255.0.0.0
; Another RFC1918 with CIDR notation
;localnet=172.16.0.0/12
;Zero conf local network
;localnet=169.254.0.0/255.255.0.0

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

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

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

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

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

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

;REALTIME SUPPORT
; For additional information on ARA, the Asterisk Realtime Architecture, please read realtime.txt and extconfig.txt in
; the /doc directory of the source code.
;
; 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)
;rtcachefriends=yes

; Save systemname in realtime database at registration. Default= no
;rtsavesysname=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
;rtupdate=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.
;rtautoclear=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
;ignoreregexpire=yes


; 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

;Add domain and configure incoming context for external calls to this domain
;domain=mydomain.tld,mydomain-incoming
; Add IP address as local domain. You can have several "domain" settings
;domain=1.2.3.4
; Disable INVITE and REFER to non-local domains. Default is yes
;allowexternaldomains=no
; Turn this on to have Asterisk add local host name and local IP to domain list
;autodomain=yes

; 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
; fromdomain=mydomain.tld

; ควบคุม Jitter Buffer เพื่อให้เสียงราบรื่นหู
;JITTER BUFFER CONFIGURATION
; 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
; jbenable = yes

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

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

; 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
; jbresyncthreshold = 1000

; 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.
; jbimpl = fixed
; Enables jitterbuffer frame logging. Defaults to "no".
; jblog = no
voip4share
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ลงทะเบียนเมื่อ: 18 พ.ย. 2009 11:26
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Re: ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

โพสต์โดย voip4share » 17 ธ.ค. 2009 12:42

2. ข้อมูลของ SIP Extension

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

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


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

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


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


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

;[cisco1]
;type=friend
;secret=blah
; Qualify peer is no more than 200ms away
;qualify=200
; This phone may be natted. Send SIP and RTP to the IP address that packet is received from instead of trusting SIP headers
;nat=yes
; This device registers with us
;host=dynamic
; 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)
;canreinvite=no
; IP address to use until registration
;defaultip=192.168.0.4
; Username to use when calling this device before registration Normally you do NOT need to set this parameter
;username=goran
; Channel variable to be set for all calls from this device
;setvar=CUSTID=5678

;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.You must have this turned on or DTMF reception will work improperly.
;rfc2833compensate=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.
;t38pt_usertpsource=yes
voip4share
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Re: ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

โพสต์โดย voip4share » 17 ธ.ค. 2009 12:44

3. ข้อมูลของ SIP Trunk

;[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]
; we only want to call out, not be called
;type=peer
;secret=guessit
; Authentication user for outbound proxies
;username=yourusername
; Many SIP providers require this!
;fromuser=yourusername
;fromdomain=provider.sip.domain
;host=box.provider.com
; This provider requires ";user=phone" on URI
;usereqphone=yes
; Permit only 5 simultaneous outgoing calls to this peer
;call-limit=5
; Send outbound signaling to this proxy, not directly to the peer Call-limits will not be enforced on real-time peers, since they are not stored in-memory
;outboundproxy=proxy.provider.domain
; The port number we want to connect to on the remote side. Also used as "defaultport" in combination with "defaultip" settings
;port=80
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Re: ไฟล์ sip.conf มีไว้ทำอะไร มีรายละเอียดอย่างไร

โพสต์โดย voip4share » 17 ธ.ค. 2009 12:47

4. เมื่อ Asterisk ของเราต้องไปรีจิสเตอร์กับ SIP Server อื่น

เช่นเมื่อต้องเอา Asterisk ไปรีจิสเตอร์กับผู้ให้บริการ SIP อื่นครับ เช่น TOT NetCall, True NetTalk, TT&T CallCafe, ThaiSIP เป็นต้น

ข้อมูลเหล่านี้ต้องเขียนให้อยู่ภายใต้ [general] นะครับ ที่ผมแยกออกมาแบบนี้ก็เพื่อให้ดูง่ายๆ

;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions

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