Asterisk-sip.conf

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Overview

Asterisk sip.conf 문서 설명

General section

SIP Configuration - general The [general] section of sip.conf includes the following variables:


  • allowsubscribe = yes|no
Allow or Ignore Subscribe requests
  • allow = <codec>
Allow codecs in order of preference (Use DISALLOW=ALL first, 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 yes. (New in v1.2.x).
  • allowguest = yes|no
Allow or reject guest calls. Default is yes. (this can also be set to 'osp' if asterisk was compiled with OSP support). (New in v1.2.x).
  • Asterisk sip allowoverlap = yes|no
Enable/disable overlap dialing support.
Default yes (Overlap dial provides for a longer time-out period between digits, also called the inter-digit timer.
With overlap dial set to off, 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, then the device waits up to about 2 seconds between digits).
  • autocreatepeer = yes|no
If set, 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 hostnames and local IP address to the domain list.
externip or externhost are also taken into the domain list. Default no. (New in v1.2.x).
  • bindaddr = IP_Address
IP Address to bind to (listen on).
Default 0.0.0.0 (all network interfaces).
  • bindport = Number
UDP Port to bind to (listen on).
Used to be port in Asterisk v1.0.x. Default 5060.
  • callerid = <string>
Caller ID information used when nothing else is available.
Defaults to asterisk. (The ability to override the default appears to available in Asterisk 1.0.9. Unsure about other versions.)
  • canreinvite = update|yes|no|nonat (global setting)
For some reason this defaults to yes, 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 should send compact (i.e. abbreviated) headers in the SIP messages.
Default no. (New in v1.2.x)
  • context = <contextname>
This is the default context and is used when a endpoint has no context property.
The context in section of an endpoint is used to route calls from that endpoint to the wanted destination.
The context body is located in extensions.conf.
  • defaultexpiry= Number
Default duration (in seconds) of incoming/outgoing registration. Default 120 seconds.
  • domain = domains
Comma separated list of domains which Asterisk is responsible for. (New in Asterisk 1.2.x)
  • dtmfmode = inband|info|rfc2833 (global setting)
Default rfc2833. Warning: inband very high CPU load.
  • dumphistory = yes|no
Enable support for dumping of SIP conversation's transaction history to LOG_DEBUG. Default no. (New in v1.2.x)
  • externip = IP_Address or a hostname
Address that we're going to put in SIP messages if we're behind a NAT.
If a hostname is used as the value, 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, 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 entered in 'externhost'. Default 10 seconds. (New in Asterisk 1.2.x).
  • ignoreregexpire = yes|no
Indicates whether to use Contact information about a peer even if the information is stale because it has reached its expiration time.
Default no. (New in v1.2.x)
  • jbenable = yes|no
Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4)
  • jbforce = yes|no
Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no". (Added in Version 1.4)
  • jbmaxsize = Number
Max length of the jitterbuffer in milliseconds. (Added in Version 1.4)
  • jbresyncthreshold = Number
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. (Added in Version 1.4)
  • jbimpl = fixed|adaptive
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. (Added in Version 1.4)
  • jblog = no|yes
Enables jitterbuffer frame logging. Defaults to "no". (Added in Version 1.4)
  • language = <string>
Default language used by any Playback()/Background().
  • limitonpeers = yes|no
If set to yes use only the peer call counter for both incoming and outgoing calls (ref. hints, subscriptions, BLF; added in 1.4)
  • localnet = NetAddress/Netmask
local network and mask.
  • fromdomain= <domain>
Sets default From: domain in SIP messages when acting as a SIP ua (client)
  • insecure = very|yes|no|invite|port
Specifies how to handle connections with peers.
Default no (authenticate all connections). invite and port added in v1.2.x, yes and very removed in v1.6.x, 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 = value
one of the classes specified in musiconhold.conf
  • musdiconhold = value
same as musicclass
  • nat = yes|no
Please note that as of Asterisk 1.0.x nat can now have the values: yes|no|never|route. Default no which really means to use rfc3581 techniques.
  • notifymimetype = mediatype/subtype
Allow overriding of mime type in MWI NOTIFY used in voicemail online messages. Valid MIME types can be found here. Default application/simple-message-summary. (New in v1.2.x).
  • notifyringing = yes|no
Notify subscription on RINGING state. Default yes. (New in v1.2.x).
  • outboundproxy = IP_address or DNS SRV name (excluding the _sip._udp prefix)
SRV name, hostname, 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, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 default yes).
  • port = <portno>
Default SIP port of peer. (this is not the port for Asterisk to listen. See bindport).
  • progressinband = never|no|yes
If we should generate in-band ringing always. Default never.
  • promiscredir= yes|no
Allows support for 302 Redirects; (Note: will redirect them all to the local extension returned in Contact rather than to that extension at the destination). Default no.
  • qualify = yes|no|milliseconds
Check if client is reachable. If yes, 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 SIP conversation's transaction history. Default no. (New in v1.2.x).
  • regcontext = context
Default context to use for SIP REGISTER replies from the SIP Registrar.
  • register => <username>:<password>:[authid]@<sip client/peer id in sip.conf>/<contact>
Register with a SIP provider
  • registerattempts = Number
Number of SIP REGISTER messages to send to a SIP Registrar before giving up. Default 0 (no limit). (New in v1.2.x).
  • registertimeout = Number
Number of seconds to wait for a response from a SIP Registrar before classifying the SIP REGISTER has timed out. Default 20 seconds. (New in v1.2.x).
  • relaxdtmf = yes|no
Default no.
  • rtautoclear = yes|no|number
Auto-Expire friends created on the fly. If yes the autoexpire will be in 120 seconds. Default yes. (New in v1.2.x). Buggy up to 1.4.19, 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.?)
  • rtpholdtimeout = Number
Max number of seconds of inactivity before terminating a call on hold. Default 0 (no limit). (New in v1.2.x).
  • rtpkeepalive = 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). (New in v1.2.x).
  • rtptimeout = Number
Number of seconds, 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 support. Default yes. (New in v1.2.x).
  • sendrpid = yes|no
If a Remote-Party-ID SIP header should be sent. Default no.
  • sipdebug = yes|no
Default setting for whether SIP debug is enabled upon loading of the sip.conf. Default no. (New in v1.2.x).
  • srvlookup = yes|no
Enable DNS SRV lookups on calls. Default yes. (Default is no prior to v1.4.14)
  • tos = <value>
Set IP QoS parameters for outgoing media streams (numeric values are also accepted, like tos=184 )
  • trustrpid = yes|no
If Remote-Party-ID SIP header should be trusted. Default no.
  • useclientcode = yes|no
If yes, 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 ";user=phone" to the URI. Default no. (New in v1.2.x)
  • useragent = <string>
Allow the SIP header "User-Agent" to be customized. Default asterisk.
  • videosupport = yes|no
Turn on support for SIP video (peer specific setting added in SVN Dec 21 2005, bug 5427. Default no.
  • vmexten = <string>
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.
  • disallowed_methods=
(1.8.x) 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 and 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. Its use may be expanded in the future.

preferred_codec_only= (1.8.x) 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.

engine= (1.8.x) RTP engine to use when communicating with the device


Peer section

SIP configurations - peers and clients These variables can be configured for each SIP peer definition:

(If not specified, the configuration variable can be used for both type=peer and type=user.)


  • accountcode = <string>
Users may be associated with an accountcode. See Asterisk billing
  • allow = <codec>
Allow codecs in order of preference (Use DISALLOW=ALL first, 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, 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, omit, billing, documentation. See Asterisk billing
  • astdb
Appears to insert a value in the Asterisk database. See example below.
  • auth = <authname>
Value assigned to the Digest username= SIP header.
  • callerid = <string>
Caller ID information used when nothing else is available. Defaults to asterisk.
  • busylevel = number
Number of simultaneous calls until user/peer is busy
  • call-limit = number
Number of simultaneous calls through this user/peer.
  • callgroup = num1,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, allowed_passed_screen, allowed_failed_screen, allowed, prohib_not_screened, prohib_passed_screen, prohib_failed_screen, prohib, and unavailable.
See SetCallerPres for more information.
Default allowed_not_screened.
  • canreinvite = update|yes|no|nonat
Client 가 SIP re-invite 를 할 수 있는지 없는지를 설정한다.
기본값은 yes 이다.
  • cid_number = <string>
On incoming (through this peer) calls sets the outbound $CALLERID(num) to <string>. (New in v.1.4.x)
  • context = <context_name>
If type=user, 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 entry matches an inbound call, then a type=peer or type=friend will match if the hostname or IP address defined in host= matches.
  • defaultip = Dotted.Quad.IP.Addr
host=dynamic 지정시 사용하게 될 기본 IP address 를 설정한다.
Client 가 아직 Register 하지 않았을 경우 사용된다.
Valid only for type=peer.
  • defaultuser
사용자 id를 설정한다. username 항목은 사용하지 않는다.
  • directrtpsetup = yes|no
canreinvite 옵션과 비슷하다. 하지만 SIP Proxy 와 비슷하게 별도의 미디어 처리과정 없이 바로 콜을 넘겨준다는 점이 다르다.
  • dtmfmode = inband|info|rfc2833
DTMF 시그널 처리 방식을 설정한다.
기본값은 rfc2833 이다.
inband 방식 지정시, 많은 량의 CPU 파워가 필요하다.
  • fromuser = <from_ID>
Specify user to put in "from" instead of $CALLERID(number) (overrides the callerid) when placing calls _to_ peer (another SIP proxy).
Valid only for type=peer.
"from" 에 $CALLERID(number) 대신에 여기에 설정된 ID를 입력한다. type=peer 일때만 유효하다.
  • fromdomain = <domain>
Sets default From: domain in SIP messages when placing calls _to_ peer.
Valid only when in [general] section or type=peer.
SIP 헤더에서의 Default from: domain 을 설정한다. type=peer 일때만 유효하다.
SIP URI contact for realtime peer. Valid only for realtime peers.
  • fullname = "Full Name"
Sets outbound $CALLERID(name). (New in v1.4.x)
  • host = dynamic|hostname|IPAddr
Client 접속 방식을 지정한다.
지정된 IP 주소 혹은 도메인 네임, 혹은 동적 IP 등록 가능 여부를 설정할 수 있다.
  • incominglimit and outgoinglimit = Number
Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer.
  • insecure = very|yes|no|invite|port
Specifies how to handle connections with peers. Default no (authenticate all connections).
Invite and port added in v1.2.x, yes and very removed in v1.6.x, 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 - defines language for prompts
  • mailbox = mailbox
Voicemail extension (for message waiting indications). Valid only for type=peer. Edit: also valid for type=friend (verified with 1.4.22.1).
  • md5secret
MD5-Hash of "<user>:==SIP_realm==:<secret>" (can be used instead of secret).
Default for authenticating to an Asterisk server when SIP realm is not explicitly declared is "<user>:asterisk:<secret>".
  • musicclass
one of the classes specified in musiconhold.conf
  • musiconhold
Set class of musiconhold on calls from SIP phone.
Calls to the phone require SetMusicOnHold cmd of higher priority (lower numerical value of priority) than Dial cmd in dialplan in order to set this class for the call.
Calls have the MusicOnHold class set on a per call basis, not on a per phone basis, 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 (added in v1.4)
  • name = <name>
Name of the realtime peer. If the peer is an actual phone then this will generally be the extension number of that phone.
On some softphones this field corresponds to the "username" field/option in the softphone's settings. Valid only for realtime peers.
  • nat = yes|no
This variable changes the behaviour of Asterisk for clients behind a firewall.
This does not solve the problem if Asterisk is behind the firewall and the client on the outside.
Please note that Asterisk 1.0.x nat can now have the values: yes|no|never|route; Asterisk 1.8 can have the values: yes|no|force_rport|comedia.
Default no which really means to use rfc3581 techniques.
  • outboundproxy = IP_address or DNS SRV name (excluding the _sip._udp prefix)
SRV name, hostname, or IP address of the outbound SIP Proxy. Valid only in [general] section and type=peer. (New in v1.2.x).
  • permit, deny,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
  • progressinband = never|no|yes
If we should generate in-band ringing always.
Default never.
  • promiscredir = yes|no
Allows support for 302 Redirects;
(Note: will redirect them all to the local extension returned in Contact rather than to that extension at the destination).
Default no.
  • qualify = yes|no|milliseconds
Client 의 연결 여부를 확인한다.
Yes 설정시, "sip show peers" 명령에서 Status 부분에서 연결 여부를 확인할 수 있다.
연결 확인 주기는 qualifyfreq 항목을 참조한다. 만약 숫자를 설정하게 되면, millisecond 단위로 확인 주기가 설정된다.
설정 전
 Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
 sip_test_01/sip_test_ 192.168.0.56                             D  Auto (No)  No             55500    Unmonitored                                  
 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 3 offline]
설정 후
 Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
 sip_test_01/sip_test_ 192.168.0.56                             D  Auto (No)  No             64129    OK (179 ms)                                  
 4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]
  • regexten =
None
  • regseconds = seconds
Number of seconds between SIP REGISTER. Valid only for realtime peer entries.
  • restrictcid
(yes/no) To have the callerid restricted -> sent as ANI; use this to hide the caller ID.
This does not seem to work. This variable has been deprecated as of v1.2.x.
  • rtpkeepalive = seconds
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 call if x seconds of no RTP activity when we're not on hold. Valid only in [general] section and type=peer.
  • rtpholdtimeout = seconds
Terminate call if x seconds of no RTP activity when we're on hold (must be larger than rtptimeout). Valid only in [general] section and type=peer.
  • secret
If Asterisk is acting as a SIP Server, 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 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-Party-ID SIP header should be sent. Default no.
  • setvar = variable=value
Channel variable to be set for all calls from this peer/user.
  • subscribecontext = <context_name>
Set a specific context for SIP SUBSCRIBE requests
  • trunkname
Indicates this peer definition is for a SIP trunk. As a result, 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 = user|peer|friend
Relationship to client - outbound provider or full client?
peer : Match incoming requests to a configuration entry using the source IP address and port number.
user : Match incoming requests to a configuration entry using the username in the From header of the SIP request. This name is matched to a section in sip.conf with the same name in square brackets.
friend : This enables matching rules for both peer and user. This is the setting most commonly used for SIP phones.
  • useclientcode = yes|no
If yes, 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 ";user=phone" to the URI. Default no. Valid only in [general] and type=peer.
  • username = <username[@realm]>
If Asterisk is accepting SIP INVITE requests from a remote SIP client, this field specifies the user name for authentication. (Contrast with fromuser.) Also, for peers that
register with Asterisk, this username is used in INVITEs until we have a registration.
  • vmexten = <string>
Dialplan extension to reach mailbox. Default asterisk. Valid only in [general] or type=peer.




Notes 

   Asterisk 1.6 and later support SIP over TCP. Before that it only supports SIP over UDP. Asterisk 1.8 comes with IPv6 support.
   For Grandstream phones: set dtmfmode=info
   Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled silence suppression. There is a solution for the silence suppression problem, see bug 5374 for details.

Example

outbound 설정

자체 트렁크를 가지고 있고, 트렁크를 사용하여 발신을 하고 싶다면 다음의 설정을 해주어야 한다.

  • outbound example 1
[mysipprovider-out]
type=peer
secret=password
username=2345
host=sipserver.mysipprovider.com
fromuser=2345
fromdomain=fwd.pulver.com
canreinvite=no
insecure=invite,port
qualify=yes
nat=yes
context=from-mysipprovider ; this section will be defined in extensions.conf
  • outbound example 2
[myprovider]
type = peer
host = your.provider.tld
defaultuser = username
secret = password
; Most providers won't authenticate when they send calls to you,
; so you need this line to just accept their calls.
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = ulaw

External link

References

<references />

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