sip.conf configuration Details

[2001]

Type=friend

context=localsets

host=dynamic

Nat=yes

Canreinvite=no

secret=123456

dfmfmode=rfc2833

Disallow=all

Allow=ulaw

Allow=alaw

allow=h263

Description

(1) the type of TYPE:SIP. Format: Type =user|peer|friend.

peer is used to authenticate outbound calls, and if you want to have multiple phones in a user (extension), define a extension that can call two SIP peers. User is used to authenticate incoming calls and users reach the server through contextual authentication. Friend is used to authenticate inbound and outbound calls, equivalent to (Peer+user).

(2) Username: Format: username =<username[@realm]>. If Asterisk accepts a client from a SIP invite request from a remote SIP, this field specifies the authenticated user name.

(3) AccountCode: Format: AccountCode =<string>. This field is used to populate the CDR (call detail record) for the "AccountCode" realm.

(4) Secret: The password used for authentication. If Asterisk is acting as a SIP proxy, the SIP client must log on with this password (a shared password). If Asterisk is a SIP client that requires authentication as a SIP invite server for a remote SIP, then this field is used to verify that the asterisk SIP protocol is drawn to the remote SIP server.

(5) Context: Format: Context = <context_name> defines the location of the instruction, controls the phone's permissions, and how to handle incoming calls to this number. If the type is user, the context defines the incoming call to use. If the type is a node, the context defines the outgoing call to use. If the type is friend, define the context used for incoming and outgoing calls through the SIP entity.

(6) Host: Format: Host =dynamic|hostname| IPAddr. The host parameter specifies the user's hostname or SIP endpoint IP address. Configuring Host=dynamic will require a number to register, allowing Asterisk to know how to find the phone.

(7) NAT: Format: Nat =yes|no. This variable changes the behavior of the client behind the asterisk firewall. Configure Nat=yes, forcing Asterisk to ignore the contact information of the number, using the address information of the received package.

(8) Qualify: Format: Qualify =yes|no|milliseconds. To check if the client is reachable, we can monitor the delay between the asterisk server and the phone, and use Qualify=yes to verify that the remote device is reachable. The Qualify=yes can be used to monitor any remote device, including other asterisk servers. By default, Asterisk considers the delay to be up to 2,000 MS (2 seconds). You can configure asterisk to determine whether the peer can reach the time, by replacing yes with milliseconds.

(9) Canreinvite: Format: Canreinvite =update|yes|no|nonat. In the SIP protocol, the invitation is used to initiate the call, redirecting the media. Any invitations that are initiated in the same conversation after the initial invitation are considered re-invited (Reinvite). Configuring Canreinvite=no allows the asterisk media channel to pass through itself without allowing RTP information to be transmitted directly between endpoints. Asterisk does not initiate a re-invitation under any of the following circumstances: If either side of the client is configured as Canreinvite=no, Asterisk needs to perform a speech encoding conversion if the client cannot negotiate the encoding, if either side of the client is configured to Nat=yes If the asterisk needs to listen for dual tone multi-frequency (DTMF) tones (for call forward or other functions) in the call. Configure Canreinvite=yes "Allow RTP Media direct". Canreinvite = Nonat "Allow reinvite when local, deny Reinvite when Nat". Configure Canreinvite=update "Use update instead of invite". Configure Canreinvite = Nonat "Where the update is used, deny when Nat".
(Ten) Callgroup: Format: Callgroup =num1,num2-num3. A phone group that defines this extension.

(one) Pickupgroup: The same group can answer the phone, press *8 application.
(SETVAR): Format: Setvar = Variable=value. The channel variable is set to all calls from that node/user

Call-limit: Format call-limit = number. Number of simultaneous calls

2. In addition to the auto-generated configuration there are other configurations

(1) Allow: Format: Allow =<codec>. The codec is allowed in order of precedence.

(2) Disallow: Format: Disallow =all. This peer or any user-defined codec is not allowed.

(3) Allowguest: Format: allowguest = Yes|no. Reject or allow incoming calls.

(4) Amaflags: The CDR record of the classification. Select Yes by default, omit, billing, file.

(5) Astdb: Inserts a value into the asterisk database.

(6) Auth: Format: Auth =<authname>.

(7) CallerID: Format: CallerID = <string>. Use Call ID information when no information is available

(8) Busylevel: Format: busylevel= number. Simultaneous number of calls until the user/peer is busy

(9) Callingpres: Format: Callingpres =number|descriptive_text. Set the call display for the phone, the value that is valid to describe is allowed_not_screened, Allowed_passed_screen, allowed_failed_screen,allowed,prohib_not_screened, Prohib_passed_screen, Prohib_failed_screen, Prohib, and unavailable.

(10). Cid_number: Format: cid_number = <string>. Set a string to display externally

(one) Defaultip: Format Defaultip =dotted.quad.ip.addr. The default IP address if the client specifies host=dynamic. This entry is used if the client does not register with any other IP address. Only available for Type=peer

(directrtpsetup): Format Directrtpsetup =yes|no. Similar to Canreinvite, the media can be passed to the other side like the SIP agent immediately.

dtmfmode: Format: Dtmfmode =inband|info|rfc2833. How the customer handles DTMF signaling. Default rfc2833. Configure Dtmfmode = rfc2833 to allow dual-tone multi-frequency (DTMF) tones to be listened to in calls (for call forward or other functions)

Fromuser: Format: Fromuser =<from_id>. Specifies that the user enters "from" in place of $callerid (number).

(15) ... Fromdomain: Format: Fromdomain = <domain>.

(+) FULLCONTAC: Format: fullcontact = <sip:uri_contact>. SIP URI Contact, real-time peers. Only available in real-time peers

FullName: Format: FullName = "FullName". Set up outgoing Caller ID (name).

Incominglimitand outgoinglimit: Format: Incominglimitand outgoinglimit = number. Limit the number of simultaneous calls to SIP clients, only for Type=peer.

(Insecure:very|yes|no|invite|port). Specifies how the connection to the peers is handled.

Language: This is used for a specified language setting in the asterisk SIP account configuration option for this client. By using this setting, you may get localized sounds in different language prompts for different users.

(+) Mailbox: Format Mailbox =mailbox. Voice mail.

(Musicclass): class specified in musiconhold.conf

(musiconhold): Keep music.

SUBSCRIBEMWI: Instructs Asterisk not to send noitfy information while waiting for information. Determine how asterisk notifies the SIP client about voice mail messages

(+) Permit, deny, mask format: Permit=<ipaddress>/<network mask> deny=<ipaddress>/<n Etwork mask>

IP address and network restrictions. Allow or restrict access to certain specific networks.

(+) Client for Port:sip Port

Progressinband: Format: Progressinband =never|no|yes.

() Promiscredir: Format: Promiscredir = Yes|no. Whether to allow support for 302 redirects.

Regseconds: Format: regseconds = seconds. The number of seconds the SIP is registered.

(sendrpid): Format: Sendrpid =yes|no. Determine if the REMOTE-PARTY-IDSIP header is sent

(+) Subscribecontext: Format: Subscribecontext =<context_name>. Set a special context for SIP subscribe

(+) Trunkname: Define a name for the relay

(trustrpid): Format: Trustrpid =yes|no. Sets whether the REMOTE-PARTY-IDSIP header is trusted.

Vmexten: Format vmexten = <string> dialing rules extended to mailbox