Change HostName From Command Line Linux
To change the hostname, follow the steps shown below. 1. Use hostname command to Change Hostname In this example, we’ll
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SIP.conf : Asterisk
eduguru 0 Comments actually the new jb of IAX2). Defaults to fixed. (Added in Version 1.4) • jblog = no|yes: Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4) • language = : Default language u, allowed, allowed_failed_screen, allowed_passed_screen, also called the inter-digit timer. With overlap dial set to off, and making a call through any extension specifying SetMusicOnHold will override this value for the call. • subscribemwi: Instructs Asterisk to not send NOTIFY messages for message waiting indication (, and unavailable. See SetCallerPres for more information. Default allowed_not_screened. • canreinvite = update|yes|no|nonat : If the client is able to support SIP re-invites. Default yes. • cid_number , anyone will be able to log in as a peer (with no check of credentials; useful for operation with SER). Default no. • Asterisk sip autodomain = yes|no : Enable/disable Asterisk's ability to add local h, 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, asterisk sip.conf, before allowing other codecs) • disallow = all : Disallow all codecs (global configuration) • Asterisk sip allowexternaldomains = yes|no : Enable/Disable INVITE and REFER to non-local domains. Default, before allowing other codecs) • disallow = all : Disallow all codecs for this peer or user definition. • allowguest = yes|no : Allow or reject guest calls (default is yes, billing, BLF; added in 1.4) • localnet = NetAddress/Netmask : local network and mask. • fromdomain= : Sets default From: domain in SIP messages when acting as a SIP ua (client) • insecure = very|yes|no|invite|, bug 5427. Default no. • vmexten = : Dialplan extension to reach mailbox. Default asterisk. (New in v1.2.x) • callevents = yes|no: Set to yes to receive events on AMI when a call is put on/off hold. • , but right away passes media to the other party like a SIP proxy • dtmfmode = inband|info|rfc2833 : How the client handles DTMF signalling. Default rfc2833. Warning: inband very high CPU load. • fromus, deny, documentation. See Asterisk billing • astdb : Appears to insert a value in the Asterisk database. See example below. • auth = : Value assigned to the Digest username= SIP header. • callerid = : Caller, for peers that register with Asterisk, hostname, like tos=184 ) • trustrpid = yes|no : If Remote-Party-ID SIP header should be trusted. Default no. • useclientcode = yes|no : If yes, mask : IP address and network restriction • pickupgroup : Group that can pickup fellow workers' calls using *8 and the Pickup() application on the *8 extension • port : SIP port of the client • progre, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 defaultyes). • port = : Default SIP port of peer. (this is not the port for Asterisk to listen. See bindport). • progressinba, not on a per phone basis, num2-num3 : Defines call groups for calls to this device. • callingpres = number|descriptive_text : Set Caller-ID presentation on a call. Valid descriptive values are: allowed_not_screened, omit, or IP address of the outbound SIP Proxy. (New in v1.2.x). • outboundproxyport = Number : UDP port number for the Outbound SIP Proxy. (New in v1.2.x). • pedantic = yes|no : Enable slow, or IP address of the outbound SIP Proxy. Valid only in [general] section and type=peer. (New in v1.2.x). • permit, pedantic checking of Call-ID:s, possible to use multiple options separated by commas from v1.4.x • ipaddr : Dotted Quad IP address of the peer. Valid only for realtime peers. • language : A language code defined in indications.conf , possible to use multiple options separated by commas from v1.4.x • maxexpiry = Number : Max duration (in seconds) of incoming registration we allow. Default 3600 seconds. • musicclass = one of the cla, prohib, prohib_failed_screen, prohib_not_screened, prohib_passed_screen, see bug 12707 • rtcachefriends = yes|no : Cache realtime friends by adding them to the internal list just like friends added from the config file. Default no. (New in v1.2.x). Buggy up to 1.4.19, see bug 12707 • rtsavesysname = yes|no : If set will write the value of asterisk.conf (options) systemname to the sip peer table in the field "regserver". Useful for multi-server systems. (New in v1.?, see bug 5374 for details., SIP Configuration - general The [general] section of sip.conf includes the following variables: • allowsubscribe = yes|no : Allow or Ignore Subscribe requests • allow = : Allow codecs in order of pref, so beware... • Asterisk sip checkmwi = Number : Global interval (in seconds) between mailbox checks. Default 10 seconds. (New in v1.2.x) • Asterisk sip compactheaders = yes|no : Indicates Asterisk sho, some endpoints either do not include an Allow header or lie about what methods they implement. In the former case, subscriptions, the $CALLERID(name) will start off blank and requires the dialplan to set the$CALLERID(name). (New in v1.6.x) • trustrpid = yes|no : If Remote-Party-ID SIP header should be trusted. Default no. • type, the checks occur every 60 seconds. Default no. • realm = my realm (Change authentication realm from asterisk (default) to your own. Requires Asterisk v1.x) • recordhistory = yes|no. Enable logging of , the checks occur every 60 seconds. Valid only in [general] section and type=peer. • regexten = • regseconds = seconds : Number of seconds between SIP REGISTER. Valid only for realtime peer entries. • , the configuration variable can be used for both type=peer and type=user.) • accountcode = : Users may be associated with an accountcode. See Asterisk billing • allow = : Allow codecs in order of prefe, the Context for the inbound call from this SIP user definition. If type=peer, the Context in the dialplan for outbound calls from this SIP peer definition. If type=friend the context used for both inbound and outbound calls through the SIP entities definition. If no type=user e, the gateway expects to receive the digits one right after the other coming in to this line with very little delay between digits. With overlap dial set to on, the only actual use for this currently is for determining if Asterisk may send connected line UPDATE requests. Its use may be expanded in the future. • preferred_codec_only= (1.8.x) Respond to a SIP i, the other endpoint should include an Allow header telling us what SIP methods the endpoint implements. However, then a type=peer or type=friend will match if the hostname or IP address defined in host= matches. • defaultip = Dotted.Quad.IP.Addr : Default IP address of client if host=dynamic is specified. Used i, then the Call Originator as stated in the CDR will be changed to whatever is specified in a X-ClientCode SIP Header. Default no. (New in v1.2.x) • usereqphone = yes|no : Indicates whether to add a ";u, then the device waits up to about 2 seconds between digits). • autocreatepeer = yes|no : If set, then the IP address associated with the hostname is looked up only once during the reading of the sip.conf. If you want support for a hostname associated with a dynamic IP address, 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, 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 this field is used to authenticate SIP INVITEs that Asterisk sends to the remote SIP server. Asterisk 1.6.2.x: Changed the secret parameter to remotesecret. • sendrpid = yes|no : If a Remote-Part, then this SIP client must login with this Password (A shared secret). If Asterisk is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, 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, this can also be set to 'osp' if asterisk was compiled with OSP support). (New in v1.2.x). • amaflags : Categorization for CDR records. Choices are default, this field specifies the user name for authentication. (Contrast with fromuser.) Also, this username is used in INVITEs until we have a registration. • vmexten = : Dialplan extension to reach mailbox. Default asterisk. Valid only in [general] or type=peer. Notes • Asterisk 1.6 and later, to wait for RTP traffic before classify the connection as discontinued. Default 0 (no RTP timeout). (New in v1.2.x). • rtupdate = yes|no : Send registry updates to the database when using Realtime sup, use Asterisk sip externhost. • externhost = hostname.tld : (New in Asterisk 1.2.x) • externrefresh = Number : Specify how often (in seconds) a hostname DNS lookup should be performed for the value ent, use the keyword dynamicinstead of Host IP. • incominglimit and outgoinglimit = Number : Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer. • insecure : very|yes, 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, usually sent from exotic devices and programs. Defaults to 1000. (Added in Version 1.4) • jbimpl = fixed|adaptive: Jitterbuffer implementation, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive). (New in v1.2.x). • rtptimeout = Number : Number of seconds, when a RTP Keepalive packet will be sent if no other RTP traffic on that connection. Default 0 (no RTP Keepalive). Valid only in [general] section and type=peer. • rtptimeout = seconds : Terminate cal, with big jumps in/broken timestamps, yes and very removed in v1.6.x, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppression problem
SIP Configuration – general The [general] section of sip.conf includes the following variables: allowsubscribe = yes|no : Allow or Ignore Subscribe
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