sip extension

7/23/2019 Sip Extension

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Asterisk Real-time Architecture Development

Sr NO Column Description Default value12 Deny deny=192.168.40.38/255.255.255.255

Denies traffic from this IP address

permit=192.168.40.0/255.255.255.0

!!o"s traffic from this net"or#

deny=0.0.0.0/0.0.0.0

permit=216.20$.245.4$/255.255.255.255

Deny e%ery address e&cept for the on!y one

a!!o"ed

3 Permint deny=192.168.40.38/255.255.255.255Denies traffic from this IP address

permit=192.168.40.0/255.255.255.0

!!o"s traffic from this net"or#

deny=0.0.0.0/0.0.0.0

permit=216.20$.245.4$/255.255.255.255

Deny e%ery address e&cept for the on!y onea!!o"ed

'(a!ify=&&&)no)yes If yo( t(rn on'(a!ifyin the confi*(ration of a+IP de%ice insip.conf, steris# "i!! send a

+IP-PI-+command re*(!ar!y to chec# that

the de%ice is sti!! on!ine. If the de%ice does

not ans"er "ithin the confi*(red or defa(!t

period in ms steris# considers the de%ice

off!ine for f(t(re ca!!s. his stat(s can e

chec#ed y the+IPP f(nction, and

in%erse!y this f(nction "i!! on!y pro%ide stat(s

information for peers "hich ha%e'(a!ify=yes.

his feat(re may a!so e (sed to #eep a DP

session open to a de%ice that is !ocated

ehind a net"or# address trans!ator .

7y sendin* the -PI-+ re'(est, the DP

port indin* in the on the o(tside

address of the /fire"a!! de%ice is

maintained y sendin* traffic thro(*h it. If

the indin* "ere to e&pire, there "o(!d e

no "ay for steris# to initiate a ca!! to the

+IP de%ice. his can e (sed in con(nction

"ith thenat=yessettin*.

7y defa(!t chansip.c sends the '(a!ify e%ery

60 seconds. t !east in 1.6.0 yo( can chan*e

this %a!(e "ith'(a!ifyfre'. he %a!(e in

'(a!fiy = represents the timeo(t after a

pac#et is sent efore "e consider the peer to

e (nreacha!e. If the pac#et is not

responded "ithin 1 second, asteris# "i!! #eep

tryin* (nti! $ pac#ets ha%e fai!ed. t this

point, asteris# "on:t try a*ain (nti! the ne&t

60 cyc!e period comp!etes. If a pac#et is !ost,

"hich can easi!y happen "ith DP, there are

$ more pac#ets "hich are transmitted.

'(a!ify=yes;

is i*nored if the peer is

rea!time and cachin* is not

t(rned on.

!nteractcrm"com
http://www.voip-info.org/wiki/view/Asterisk+config+sip.confhttp://www.voip-info.org/wiki/view/SIP+method+optionshttp://www.voip-info.org/wiki/view/Asterisk+func+sippeerhttp://www.voip-info.org/wiki/view/SIP+method+optionshttp://www.voip-info.org/wiki/view/Asterisk+func+sippeerhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf


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dditiona!!y asteris# "i!! #eep tryin* e%ery 60

seconds. +o e%en if a!! $ pac#ets are !ost,

asteris# tries a*ain at the ne&t 60 second

cyc!e. he n(mer of retransmits and time

et"een each '(a!ify is defined in

chansip.c,

# registerattempts

$ !nsecure insecure= %ery)yes)no)in%ite)port < +pecifies ho" to hand!e

connections "ith peers.

Defa(!t noa(thenticate a!!

connections. inviteand portadde

d in %1.2.&, yesand veryremo%ed

in %1.6.&, possi!e to (se m(!tip!e

options separated y commas from

%1.4.&

!nsecure%no

& NA' if nat=force_rportinone section and

nat=no in the; other, then validpeers with settingsdiffering from thosein the generalsection will

; be discoverable.

'force_rport,comedi

a'( directmedia

) insecure port:i*nore the port n(mer "here re'(estcame from

invite:don:t re'(ire a(thentication of

incomin* IIs

port,invite:don:t re'(ire initia! II to

a(thenticate and i*nore the port "here the

re'(est came from

insec(re=port > !!o"

matchin* of peer y IP

address "itho(t matchin* port

n(mer

insec(re=in%ite > Do not

re'(ire a(thentication of

incomin* IIs

insecure=port,invite ;

oth

ypica!!y (sed to a!!o"

incomin* ca!!s e.*. from

?@D "hi!e ha%in* a

type=friend entry defined "ith

(sername and pass"ord.1* transport=udp transport=udp transport=udp

11 +tmfmo+e Dtmfmo+e%all auto12 +irectme+ia ,es yes

13 Call roup

!nsi+e .feature"conf

-N/00

!nteractcrm"com
http://www.voip-info.org/wiki/view/Asterisk+sip+insecurehttp://www.voip-info.org/wiki/view/Asterisk+sip+insecure


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Name of roup

Domestic!nternation

AentSupervisore1 Pickuproup A named callgroup and pickupgroup can be

set to a comma separated list of case

sensitive name strings. The number of

named groups is unlimited. The number of

named groups you can specify at once is

limited by the line length supported.

-Null

1# 0anuae or locali4ation --Null1$ +isallo5 Au+io Co+ec to +isallo5 all1& allo5 Co+ec to allo5 au+io Ala5ula51( trustpi+1) proresan+ pro*ressinand=yes

@hen ;IA; e%ent is re'(ested, a!"ays

send 180 Ringingif it hasn:t een sent yet

fo!!o"ed y 183 Session Progressand in

and a(dio

pro*ressinand=no

+end 180 Ringingif 183 has not yet een

sent esta!ishin* a(dio path. If a(dio

path isesta!ished a!ready "ith 183 then

send inand rin*in* this is the "ay asteris#

historica!!y eha%ed eca(se of (**y

phones !i#e po!ycom

pro*ressinand=ne%er

@hene%er rin*in* occ(rs, send ;180 rin*in*;

as !on* as ;200 -B; has not yet een sent.

his is the defa(!t eha%io(r of steris#.

progressinband=yes

2* promiscredir progressinband (both)

You can

set progressinbandto yes, no

, or never, to configure whether or

not to generate in-band ringing.

Normally, Asterisk will send the

progress of a call via a few methods,

such as !" #ession $rogress, !%&inging, '! usy, and so on. *f you

set progressinband=yes,

Asterisk will indicate the call

progress in band by generating

tones.progressinband=yes

Default%No

!nteractcrm"com


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promiscredir (both)

You can

set promiscredirto yesor no.

Normally, when you perform call

forwarding on a phone, Asterisk will

use the +ocal channel (for eample,

ocal!%%//0peer). *f you

setpromiscredir=yes, Asterisk

will use the #*$ channel instead,

which enables you to forward the

calls to remote boes.

21 useclientco+e useclientcode= yes)no < If yes,then the Ca!! -ri*inator as stated in

the CD "i!! e chan*ed to

"hate%er is specified in a

C!ientCode +IP Eeader. Defa(!t no.

e" in %1.2.&

no

22 accountco+e accountcode< @hat acco(nt n(mer to (seaccept>"

N/00

3* session-e6pires


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secon+s" Defualts to )* secs"

32 session-

refresher uac>>uas>? DA/0' N/00

session-refresher - 'he sessionrefresher =uacuas?" Defaults to

>uas>"

8 uac - Default

to the caller initially refreshin

5hen possi7le

8 uas - Default

to the callee initially refreshin

5hen possi7le

N/00

3333#3$3&3(3)*

!nteractcrm"com


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http.BB+o6yen"asterisk"orBtrunkBConsip"html

CRA' 'AE0


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>info>>shortinfo>>in7an+>>auto>? DA/0' N/00

yes>>no>>nonat>>up+ate>? DA/0' N/00

yes>>no>>never>>route>? DA/0' N/00

>no>? DA/0' N/00

yes>>no>>never>? DA/0' N/00

yes>>no>? DA/0' N/00


>no>? DA/0' N/00

>no>? DA/0' N/00G



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>uas>? DA/0' N/00

>no>? DA/0' N/00

>no>? DA/0' N/00




!nteractcrm"com


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>allo5e+passe+screen>>allo5e+faile+screen>>allo5

e+>>prohi7notscreene+>>prohi7passe+screen>>prohi7faile+screen>>prohi7>?

DA/0' N/00

>no>? DA/0' N/00

>no>? DA/0' N/00




>peer>>frien+>? NO' N/00 DA/0' >frien+>

PR!9AR, H, =


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H,


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; if you attempt to place a call to the peer, theexisting information

; will be used in spite of it having expired

;

; 0or realtime peers, when the peer is retrieved fromrealtime storage,

; the registration information will be usedregardless of whether

; it has expired or not; if it expires while therealtime peer

; is still in memory $due to caching or otherreasons', the

; information will not be removed from realtimestorage

SIP configuration

sip.conf

;

; SIP 4onfiguration example for steris%

;

; 9ote1 Please read the security documentation for steris% in order to

; understand the ris%s of installing steris% with the sample

; configuration. If your steris% is installed on a public

; IP address connected to the Internet, you will want to learn

; about the various security settings +0/*+ you start; steris%.

;

; +specially note the following settings1

; ) allowguest $default enabled'

; ) permit&deny&acl ) IP address filters

; ) contactpermit&contactdeny&contactacl ) IP address filters forregistrations

; ) context ) 1password>1md?secret>1authname>1transport@@@@host>1port@

; SIP&devicename&extension

; SIP&devicename&extension&IPorAost

; SIP&usernamedomain&&IPorAost

;

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;

; 3evicename

; devicename is defined as a peer in a section below.

;

; usernamedomain

; 4all any SIP user on the Internet

; $3on8t forget to enable 39S S*( 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&userproxyhostname

; where the proxyhostname is defined in a section below

; This syntax also wor%s with T8s with 0B/ ports

;

; SIP&username>1password>1md?secret>1authname@@@host>1port@

; This form allows you to specify password or md?secret and authname

; without altering any authentication data in config.

; +xamples1

;

; SIP&CD!mysipproxy

; SIP&sales1topsecret11accountEFdomain.com1?EGF

; SIP&HF"?G!11bc?fEba!cebHdedFbEeE?cfDHde"f1mynameHDF.HG!.E.H

;

; IPorAost

; The next server for this call regardless of domain&peer

;

; ll of these dial strings specify the SIP reJuest *I.

; In addition, you can specify a specific To1 header by adding an

; exclamation mar% after the dial string, li%e

;; SIP&salesmysipproxyKsalesedvina.net

;

; new feature for H.! 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. +xamples1

;

; SIP&devicename&extension&IPorAost

; SIP&usernamedomain&&IPorAost

;

; 4-I 4ommands

; ))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))); seful 4-I commands to chec% peers&users1

; 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 *eload configuration file

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; sip show settings Show the current channel configuration

;

;))))))) 9aming devices ))))))))))))))))))))))))))))))))))))))))))))))))))))))

;

; name@

; F. steris% chec%s the 0rom1 addres and matches the list of devices

; with a type=peer

; . steris% chec%s the IP address $and port number' that the I9(IT+

; was sent from and matches against any devices with type=peer

;

; 3on8t mix extensions with the names of the devices. 3evices need a uniJue

; name. The device name is CnotC used as phone numbers. Phone numbers are

; anything you declare as an extension in the dialplan $extensions.conf'.

;

; general@

context=public ; 3efault context for incoming calls. 3efaults to8default8

;allowguest=no ; llow or re5ect guest calls $default is yes'

; If your steris% is connected to the Internet; and you have allowguest=yes

; you want to chec% which services you offereveryone

; out there, by enabling them in the defaultcontext $see below'.

;match:auth:username=yes ; if available, match user entry using the

; 8username8 field from the authentication line

; instead of the 0rom1 field.

!nteractcrm"com


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allowoverlap=no ; 3isable overlap dialing support. $3efault is yes'

;allowoverlap=yes ; +nable *04?! overlap dialing support.

; 4an use the Incomplete application to collect the

; needed digits from an ambiguous dialplan match.

;allowoverlap=dtmf ; +nable overlap dialing support using 3T#0 delivery

; methods $inband, *04F!, SIP I90/' in the early

; media phase. ses the Incomplete application to

; collect the needed digits.

;allowtransfer=no ; 3isable all transfers $unless enabled in peers orusers'

; 3efault is enabled. The 3ial$' options 8t8 and 8T8are not

; related as to whether SIP transfers are allowed ornot.

;realm=mydomain.tld ; *ealm for digest authentication

; defaults to Lasteris%L. If you set a system name in

; asteris%.conf, it defaults to that system name

; *ealms #ST be globally uniJue according to *04

FGH ; Set this to your host name or domain name

;domainsasrealm=no ; se domains list as realms

; Mou can serve multiple *ealms specifying several

; 8domain=...8 directives $see below'.

; In this case *ealm will be based on reJuest80rom8&8To8 header

; and should match one of domain names.

; /therwise default 8realm=...8 will be used.

;recordonfeature=automixmon ; 3efault feature to use when receiving 8*ecord1on8 header

; from an I90/ message. 3efaults to 8automon8.


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; $Mou can choose independently for 3P, T4P, and T-S, by specifying different valuesfor

; LudpbindaddrL, LtcpbindaddrL, and LtlsbindaddrL.'

; $9ote that using bindaddr=11 will show only a single IPvG soc%et in netstat.

; IPv" is supported at the same time using IPv")mapped IPvG addresses.'

;

; Mou may optionally add a port number. $The default is port ?EGE for 3P and T4P,?EGH

; for T-S'.

; IPv" example1 bindaddr=E.E.E.E1?EGF

; IPvG example1 bindaddr=>11@1?EGF

;

; The address family of the bound 3P address is used to determine how steris%performs

; 39S loo%ups. In cases a' and c' above, only records are considered. In case b',only

; records are considered. In case d', both and records are considered.9ote,

; however, that steris% ignores all records except the first one. In case d', whenboth

; and records are available, either an or record will be first, and whichone

; depends on the operating system. /n systems using glibc, records are given

; priority.

udpbindaddr=E.E.E.E ; IP address to bind 3P listen soc%et to $E.E.E.Ebinds to all'

; /ptionally add a port number, HDF.HG!.H.H1?EGF$default is port ?EGE'

;


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;

; 9ote that the T4P and T-S support for chan:sip is currently considered

; experimental. Since it is new, all of the related configuration options are

; sub5ect to change in any release. If they are changed, the changes will

; be reflected in this sample configuration file, as well as in the PN*3+.txt file.

;

tcpenable=no ; +nable server for incoming T4P connections $defaultis no'

tcpbindaddr=E.E.E.E ; IP address for T4P server to bind to $E.E.E.E bindsto all interfaces'

; /ptionally add a port number, HDF.HG!.H.H1?EGF$default is port ?EGE'

;tlsenable=no ; +nable server for incoming T-S $secure' connections$default is no'

;tlsbindaddr=E.E.E.E ; IP address for T-S server to bind to $E.E.E.E'binds to all interfaces'

; /ptionally add a port number, HDF.HG!.H.H1?EG$default is port ?EGH'

; *emember that the IP address must match the commonname $hostname' in the

; certificate, so you don8t want to bind a T-S soc%etto multiple IP addresses.

; 0or details how to construct a certificate for SIPsee

; http1&&tools.ietf.org&html&draft)ietf)sip)domain)certs

;tcpauthtimeout = E ; 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. $default1 E seconds'

;tcpauthlimit = HEE ; tcpauthlimit specifies the maximum number of

; unauthenticated sessions that will be allowed

; to connect at any given time. $default1 HEE'

transport=udp ; Set the default transports. The order determinesthe primary default transport.

; If tcpenable=no and the transport set is tcp, wewill fallbac% to 3P.

srvloo%up=yes ; +nable 39S S*( loo%ups on outbound calls

; 9ote1 steris% only uses the first host

; in S*( records

; 3isabling 39S S*( loo%ups 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 S*(

; loo%ups for that peer or call.

!nteractcrm"com


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;pedantic=yes ; +nable chec%ing of tags in headers,

; international character conversions in *Is

; and multiline formatted headers for strict

; SIP compatibility $defaults to LyesL'

; See https1&&wi%i.asteris%.org&wi%i&display&ST&IP2Ouality2of2Service for adescription of these parameters.

;tos:sip=cs ; Sets T/S for SIP pac%ets.

;tos:audio=ef ; Sets T/S for *TP audio pac%ets.

;tos:video=af"H ; Sets T/S for *TP video pac%ets.

;tos:text=af"H ; Sets T/S for *TP text pac%ets.

;cos:sip= ; Sets !EF.Hp priority for SIP pac%ets.

;cos:audio=? ; Sets !EF.Hp priority for *TP audio pac%ets.

;cos:video=" ; Sets !EF.Hp priority for *TP video pac%ets.

;cos:text= ; Sets !EF.Hp priority for *TP text pac%ets.

;maxexpiry=GEE ; #aximum allowed time of incoming registrations$seconds'

;minexpiry=GE ; #inimum length of registrations $default GE'

;defaultexpiry=HFE ; 3efault length of incoming&outgoing registration

;submaxexpiry=GEE ; #aximum allowed time of incoming subscriptions$seconds', default1 maxexpiry

;subminexpiry=GE ; #inimum length of subscriptions, default1 minexpiry

;mwiexpiry=GEE ; +xpiry time for outgoing #


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;mwi:from=asteris% ;


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;

;par%inglot=plaa ; Sets the default par%ing lot for call par%ing

; This may also be set for individual users&peers

; Par%inglots are configured in features.conf

;language=en ; 3efault language setting for all users&peers


;toneone=se ; 3efault toneone for all users&peers


;relaxdtmf=yes ; *elax dtmf handling

;trustrpid = no ; If *emote)Party)I3 should be trusted

;sendrpid = yes ; If *emote)Party)I3 should be sent $defaults to no'

;sendrpid = rpid ; se the L*emote)Party)I3L header

; to send the identity of the remote party

; This is identical to sendrpid=yes

;sendrpid = pai ; se the LP)sserted)IdentityL header

; to send the identity of the remote party

;rpid:update = no ; In certain cases, the only method by which aconnected line

; change may be immediately transmitted is with a SIPP3T+ reJuest.

; If communicating with another steris% server, andyou wish to be able

; transmit such P3T+ messages to it, then you mustenable this option.

; /therwise, we will have to wait until we can send areinvite to

; transmit the information.

;prematuremedia=no ; Some IS39 lin%s send empty media frames before

; the call is in ringing or progress state. The SIP

; channel will then send H! indicating early media ; which will be empty ) thus users get no ringsignal.

; Setting this to LyesL will stop any media before wehave

; call progress $meaning the SIP channel will notsend H! Session

; Progress for early media'. 3efault is LyesL. lsoma%e sure that

; the SIP peer is configured withprogressinband=never.

;

; In order for LnoanswerL applications to wor%, youneed to run

; the progress$' application in the priority beforethe app.

;progressinband=never ; If we should generate in)band ringing always

; use 8never8 to never use in)band signalling, evenin cases

; where some buggy devices might not render it

; (alid values1 yes, no, never 3efault1 never

!nteractcrm"com


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;useragent=steris% PB ; llows you to change the user agent string

; The default user agent string also contains thesteris%

; version. If you don8t want to expose this, changethe

; useragent string.

;promiscredir = no ; If yes, allows EF or *+3I* to non)local SIPaddress

; 9ote that promiscredir when redirects are made tothe

; local system will cause loops since steris% isincapable

; of performing a LhairpinL call.

;usereJphone = no ; If yes, L;user=phoneL is added to uri that contains

; a valid phone number

;dtmfmode = rfcF! ; Set default dtmfmode for sending 3T#0. 3efault1rfcF!

; /ther options1

; info 1 SIP I90/ messages $application&dtmf)relay'

; shortinfo 1 SIP I90/ messages $application&dtmf'

; inband 1 Inband audio $reJuires G" %bit codec)alaw, ulaw'

; auto 1 se rfcF! if offered, inband otherwise

;compactheaders = yes ; send compact sip headers.

;

;videosupport=yes ; Turn on support for SIP video. Mou need to turnthis

; on in this section to get any video support at all.

; Mou can turn it off on a per peer basis if thegeneral

; video support is enabled, but you can8t enable itfor

; one peer only without enabling in the generalsection.

; If you set videosupport to LalwaysL, then *TP portswill

; always be set up for video, even on clients thatdon8t

; support it. This assists callfile)derived callsand

; certain transferred calls to use always use videowhen

; available. >yes79/7always@

;maxcallbitrate=!" ; #aximum bitrate for video calls $default !" %b&s'

; (ideosupport 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 LpeerstatusL events when peercan8t

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; authenticate with steris%. Peerstatus will beLre5ectedL.

;alwaysauthre5ect = yes ;


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;outboundproxy=proxy.provider.domain ; send outbound signaling to thisproxy, not directly to the devices

;outboundproxy=proxy.provider.domain1!E!E ; send outbound signaling to thisproxy, not directly to the devices

;outboundproxy=proxy.provider.domain,force ; Send -- outbound signalling toproxy, ignoring route1 headers

;outboundproxy=tls1&&proxy.provider.domain ; same as 8=proxy.provider.domain8except we try to connect with tls

;outboundproxy=HDF.E.F.H ; IPv" address literal $default portis ?EGE'

;outboundproxy=FEEH1db!11H ; IPvG address literal $default portis ?EGE'

;outboundproxy=HDF.HG!.E.F.H1?EGF ; IPv" address literal with explicitport

;outboundproxy=>FEEH1db!11H@1?EGF ; IPvG address literal with explicitport

; ; $could also be tcp,udp' ) definingtransports on the proxy line only

; ; applies for the global proxy,otherwise use the transport= option

;matchexternaddrlocally = yes ; /nly substitute the externaddr or externhostsetting if it matches

; your localnet setting. nless you have some sort ofstrange networ%

; setup you will not need to enable this.

;dynamic:exclude:static = yes ; 3isallow 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=E.E.E.E&E.E.E.E ; se contactpermit and contactdeny to

;contactpermit=HF.HG.E.E&F??.F??.E.E ; restrict at what IPs your users may

; register their phones.

;contactacl=named:acl:example ; se named 4-s defined in acl.conf

;engine=asteris% ; *TP engine to use when communicating with thedevice

;

; If regcontext is specified, steris% will dynamically create and destroy a

; 9o/p priority H extension for a given peer who registers or unregisters with

; us and have a Lregexten=L configuration item.

; #ultiple contexts may be specified by separating them with 8Q8. The

; actual extension is the 8regexten8 parameter of the registering peer or its

; name if 8regexten8 is not provided. If more than one context is provided,

; the context must be specified within regexten by appending the desired

; context after 88. #ore than one regexten may be supplied if they are

; separated by 8Q8. Patterns may be used in regexten.

;

;regcontext=sipregistrations

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;regextenonJualify=yes ; 3efault LnoL

; If you have Jualify on and the peer becomesunreachable

; this setting will enforce inactivation of theregexten

; extension for the peer

;legacy:useroption:parsing=yes ; 3efault LnoL ; If you have this optionenabled and there are semicolons

; in the user field of a sip *I,the field be truncated

; at the first semicolon seen.This effectively ma%es

; semicolon a non)usablecharacter for peer names, extensions,

; and maybe other, less testedthings. This can be useful

; for improving compatabilitywith devices that li%e to use

; user options for whatever

reason. The behavior is similar to ; how SIP *I8s were typicallyhandled in H.G.F, hence the name.

;send:diversion=no ; 3efault LyesL ; steris% normally sends3iversion headers with certain SIP

; invites to relay data aboutforwarded calls. If this option

; is disabled, steris% won8tsend 3iversion headers unless

; they are added manually.

; The shrin%callerid function removes 8$8, 8 8, 8'8, non)trailing 8.8, and 8)8 not

; in sJuare brac%ets. 0or example, the caller id value ???.???? becomes ???????; when this option is enabled. 3isabling 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.

; y default this option is on.

;

;shrin%callerid=yes ; on by default

;use:J!?E:reason = no ; 3efault LnoL

; Set to yes add *eason header and use *eason header if it isavailable.

;


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; below is for transitional compatibility only.

;

;refer:addheaders=yes ; on by default

;autocreatepeers=no ; llow any not exsplicitly defined here 4 toregister

;


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; http1&&www.openssl.org&docs&apps&ciphers.html4IPA+*:ST*I9NS

;

;tlsclientmethod=tlsvH ; values include tlsvH, sslv, sslvF.

; Specify protocol for outbound client connections.

; If left unspecified, the default is sslvF.

;

;))))))))))))))))))))))))))) SIP timers))))))))))))))))))))))))))))))))))))))))))))))))))))

; These timers are used primarily in I9(IT+ transactions.

; The default for Timer TH is ?EE ms or the measured run)trip time between

; steris% and the device if you have Jualify=yes for the device.

;

;tHmin=HEE ; #inimum roundtrip time for messages to monitoredhosts

; 3efaults to HEE ms

;timertH=?EE ; 3efault TH timer

; 3efaults to ?EE ms or the measured round)trip

; time to a peer $Jualify=yes'.

;timerb=FEEE ; 4all setup timer. If a provisional response is notreceived

; in this amount of time, the call will autocongest

; 3efaults to G"CtimertH

;))))))))))))))))))))))))))) *TP timers))))))))))))))))))))))))))))))))))))))))))))))))))))

; These timers are currently used for both audio and video streams. The *TP timeouts

; are only applied to the audio channel.

; The settings are settable in the global section as well as per device

;

;rtptimeout=GE ; Terminate call if GE seconds of no *TP or *T4P

activity ; on the audio channel

; when we8re not on hold. This is to be able tohangup

; a call in the case of a phone disappearing from thenet,

; li%e a powerloss or grandma tripping over a cable.

;rtpholdtimeout=EE ; Terminate call if EE seconds of no *TP or *T4Pactivity

; on the audio channel

; when we8re on hold $must be rtptimeout'

;rtp%eepalive=Rsecs ; Send %eepalives in the *TP stream to %eep 9T open

; $default is off ) ero'

;))))))))))))))))))))))))))) SIP Session)Timers $*04"EF!'))))))))))))))))))))))))))))))))))))

; SIP Session)Timers provide an end)to)end %eep)alive mechanism for active SIPsessions.

; This mechanism can detect and reclaim SIP channels that do not terminate throughnormal

; signaling procedures. Session)Timers can be configured globally or at a user&peerlevel.

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; The operation of Session)Timers is driven by the following configurationparameters1

;

; C session)timers ) Session)Timers feature operates in the following three modes1

; originate 1 *eJuest and run session)timers always

; accept 1 *un session)timers only when reJuested by

other ; refuse 1 3o not run session timers in any case

; The default mode of operation is 8accept8.

; C session)expires ) #aximum session refresh interval in seconds. 3efaults to H!EEsecs.

; C session)minse ) #inimum session refresh interval in seconds. 3efualts to DEsecs.

; C session)refresher ) The session refresher $uac7uas'. 3efaults to 8uas8.

; uac ) 3efault to the caller initially refreshing whenpossible

; uas ) 3efault to the callee initially refreshing whenpossible

;

; 9ote that, due to recommendations in *04 "EF!, steris% will always honor the other

; endpoint8s preference for who will handle refreshes. steris% will never overridethe

; preferences of the other endpoint. 3oing so could result in steris% and theendpoint

; fighting over who sends the refreshes. This holds true for the initiation ofsession

; timers and subseJuent re)I9(IT+ reJuests whether steris% is the caller or callee,or

; whether steris% is currently the refresher or not.

;

;session)timers=originate

;session)expires=GEE;session)minse=DE

;session)refresher=uac

;

;))))))))))))))))))))))))))) SIP 3+NNI9N)))))))))))))))))))))))))))))))))))))))))))))))))))

;sipdebug = yes ; Turn on SIP debugging by default, from

; the moment the channel loads this configuration

;recordhistory=yes ; *ecord SIP history by default

; $see sip history & sip no history'

;dumphistory=yes ; 3ump SIP history at end of SIP dialogue

; SIP history is output to the 3+N logging channel

;))))))))))))))))))))))))))) STTS 9/TI0I4TI/9S $SS4*IPTI/9S'))))))))))))))))))))))))))))

; Mou can subscribe to the status of extensions with a LhintL priority

; $See extensions.conf.sample for examples'

; chan:sip support two ma5or formats for notifications1 dialog)info and SI#P-+

;

; Mou will get more detailed reports $busy etc' if you have a call counter enabled

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; for a device.

;

; If you set the busylevel, we will indicate busy when we have a number of calls that

; matches the busylevel treshold.

;

; 0or Jueues, you will need this level of detail in status reporting, regardless

; if you use SIP subscriptions. Oueues and manager use the same internal interface

; for reading status information.

;

; 9ote1 Subscriptions does not wor% if you have a realtime dialplan and use the

; realtime switch.

;

;allowsubscribe=no ; 3isable support for subscriptions. $3efault is yes'

;subscribecontext = default ; Set a specific context for SS4*I+ reJuests

; seful to limit subscriptions to local extensions

; Settable per peer&user also

;notifyringing = no ; 4ontrol whether subscriptions already I9S+ getsent

; *I9NI9N when another call is sent $default1 yes'

;notifyhold = yes ; 9otify subscriptions on A/-3 state $default1 no'

; Turning on notifyringing and notifyhold will add alot

; more database transactions if you are usingrealtime.

;notifycid = yes ; 4ontrol whether caller I3 information is sent alongwith

; dialog)info2xml notifications $supported by snomphones'.

; 9ote that this feature will only wor% properly whenthe

; incoming call is using the same extension andcontext that

; is being used as the hint for the called extension.This means

; that it won8t wor% when using subscribecontext foryour sip

; user or peer $if subscribecontext is different thancontext'.

; This is also limited to a single caller, meaningthat if an

; extension is ringing because multiple calls areincoming,

; only one will be used as the source of caller I3.Specify

; 8ignore)context8 to ignore the called context whenloo%ing

; for the caller8s channel. The default value is8no.8 Setting

; notifycid to 8ignore)context8 also causes call)pic%ups attempted

; via S9/#8s 9/TI0M mechanism to set the context forthe call pic%up

; to PI4UP#*U.

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;callcounter = yes ; +nable call counters on devices. This can be setper

; device too.

;))))))))))))))))))))))))))))))))))))))))) T.! 0B SPP/*T))))))))))))))))))))))))))))))))))

;; This setting is available in the >general@ section as well as in deviceconfigurations.

; Setting this to yes enables T.! 0B $3PT-' on SIP calls; it defaults to off.

;

; t!pt:udptl = yes ; +nables T.! with 0+4 error correction.

; t!pt:udptl = yes,fec ; +nables T.! with 0+4 error correction.

; t!pt:udptl = yes,redundancy ; +nables T.! with redundancy error correction.

; t!pt:udptl = yes,none ; +nables T.! with no error correction.

;

; In some cases, T.! endpoints will provide a T!0ax#ax3atagram value $during T.!setup' that

; is based on an incorrect interpretation of the T.! recommendation, and results infailures

; because steris% does not believe it can send T.! pac%ets of a reasonable sie tothat

; endpoint $4isco media gateways are one example of this situation'. In these cases,during a

; T.! call you will see warning messages on the console&in the logs from thesteris% 3PT-

; stac% complaining about lac% of buffer space to send T.! 0B pac%ets. If thisoccurs, you

; can set an override $globally, or on a per)device basis' to ma%e steris% ignorethe

; T!0ax#ax3atagram value specified by the other endpoint, and use a configured valueinstead.

; This can be done by appending 8maxdatagram=Rvalue8 to the t!pt:udptlconfiguration option,

; li%e this1

;

; t!pt:udptl = yes,fec,maxdatagram="EE ; +nables T.! with 0+4 error correction andoverrides

; ; the other endpoint8s provided value toassume we can

; ; send "EE byte T.! 0B pac%ets to it.

;

; 0B detection will cause the SIP channel to 5ump to the 8fax8 extension $if itexists'

; based one or more events being detected. The events that can be detected are anincoming

; 49N tone or an incoming T.! re)I9(IT+ reJuest.

;

; faxdetect = yes ; 3efault 8no8, 8yes8 enables both 49N and T.!detection

; faxdetect = cng ; +nables only 49N detection

; faxdetect = t! ; +nables only T.! detection

;

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;))))))))))))))))))))))))))))))))))))))))) /T/93 SIP *+NIST*TI/9S))))))))))))))))))))))))

; steris% can register as a SIP user agent to a SIP proxy $provider'

; 0ormat for the register statement is1

; register = >peer6@>transport1&&@user>domain@>1secret>1authuser@@host>1port@>&extension@>Vexpiry@

;;

;

; domain is either

; ) domain in 39S

; ) host name in 39S

; ) the name of a peer defined below or in realtime

; The domain is where you register your username, so your SIP uri you are registeringto

; is usernamedomain

;

; If no extension is given, the 8s8 extension is used. The extension needs to

; be defined in extensions.conf to be able to accept calls from this SIP proxy; $provider'.

;

; similar effect can be achieved by adding a Lcallbac%extensionL option in a peersection.

; this is eJuivalent to having the following line in the general section1

;

; register = username1secrethost&callbac%extension

;

; and more readable because you don8t have to write the parameters in two places

; $note that the LportL is ignored ) this is a bug that should be fixed'.

;

; 9ote that a register= line doesn8t 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 theprovider8s

; server. If the provider has multiple servers to place calls to your system, youneed

; a peer for each server.

;

; eginning with steris% version H.G.F, the LuserL portion of the register line may

; contain a port number. Since the logical separator between a host and port numberis a

; 818 character, and this character is already used to separate between the optionalLsecretL

; and LauthuserL portions of the line, there is a bit of a hoop to 5ump through ifyou wish

; to use a port here. That is, you must explicitly provide a LsecretL and LauthuserLeven if

; they are blan%. See the third example below for an illustration.

;

;

; +xamples1

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;

;register = HF"1passwordmysipprovider.com

;

; This will pass incoming calls to the 8s8 extension

;

;

;register = F"?1passwordsip:proxy&HF"

;

; *egister F"? at sip provider 8sip:proxy8. 4alls from this provider

; connect to local extension HF" in extensions.conf, default context,

; unless you configure a >sip:proxy@ section below, and configure a

; context.

; Tip H1 void assigning hostname to a sip.conf section li%e >provider.com@

; Tip F1 se separate inbound and outbound sections for SIP providers

; $instead of type=friend' if you have calls in both directions

;

;register = "?Gmydomain1?E!F11mysipprovider.com

;

; 9ote that in this example, the optional authuser and secret portions have

; been left blan% because we have specified a port in the user section

;

;register = tls1&&username1xxxxxxsip)tls)proxy.example.org

;

; The 8transport8 part defaults to 8udp8 but may also be 8tcp8, 8tls8, 8ws8, or8wss8.

; sing 8udp1&&8 explicitly is also useful in case the username part

; contains a 8&8 $8user&name8'.

;registertimeout=FE ; retry registration calls every FE seconds $default'

;registerattempts=HE ; 9umber of registration attempts before we give up

; E = continue forever, hammering the other server

; until it accepts the registration

; 3efault is E tries, continue forever

;))))))))))))))))))))))))))))))))))))))))) /T/93 #1authuser@@host>1port@&mailbox

;

; +xamples1

;mwi = HF"1passwordmysipprovider.com&HF"

;mwi = HF"1passwordmyportprovider.com1GDGD&HF"

;mwi = HF"1password1authusermyauthprovider.com&HF"

;mwi = HF"1password1authusermyauthportprovider.com1GDGD&HF"

;

; #


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; mailbox=HF"SIP:*emote

;))))))))))))))))))))))))))))))))))))))))) 9T SPP/*T ))))))))))))))))))))))))

;

;


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; $default HEs'. This can be useful when your 9T device lets you choose

; the port mapping, but the IP address is dynamic.

; eware, you might suffer from service disruption when the name server

; resolution fails. +xamples1

;

; externhost=foo.dyndns.net ; refreshed periodically

; externrefresh=H!E ; change the refresh interval

;

; 9ote that at the moment all these mechanism wor% only for the SIP soc%et.

; 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.

;

; 9/T+ H1 in some cases, 9T boxes will use different port numbers in

; the internalR)external mapping. In these cases, the LexternaddrL and

; LexternhostL might not help you configure addresses properly.

;

; 9/T+ F1 when using LexternaddrL or LexternhostL, the address part is

; also used as the external address for media sessions. Thus, the port

; information in the S3P may be wrongK

;

; In addition to the above, steris% has an additional LnatL parameter to

; address 9T)related issues in incoming SIP or media sessions.

; In particular, depending on the 8nat= 8 settings described below, steris%

; may override the address&port information specified in the SIP&S3P messages,

; and use the information $sender address' supplied by the networ% stac% instead.

; Aowever, this is only useful if the external traffic can reach us.

; The following settings are allowed $both globally and in individual sections'1

;

; nat = no ; 3o no special 9T handling other than *04?!H; nat = force:rport ; Pretend there was an rport parameter even if therewasn8t

; nat = comedia ; Send media to the port steris% received it fromregardless

; ; of where the S3P says to send it.

; nat = auto:force:rport ; Set the force:rport option if steris% detects 9T$default'

; nat = auto:comedia ; Set the comedia option if steris% detects 9T

;

; The nat settings can be combined. 0or example, to set both force:rport and comedia

; one would set nat=force:rport,comedia. If any of the comma)separated options is8no8,

; steris% will ignore any other settings and set nat=no. If one of the LautoLsettings

; is used in con5unction with its non)auto counterpart $nat=comedia,auto:comedia',then

; the non)auto option will be ignored.

;

; The *04 ?!H)defined 8rport8 parameter allows a client to reJuest that steris%send

; SIP responses to it via the source IP and port from which the reJuest originated

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; instead of the address&port listed in the top)most (ia header. This is useful if a

; client %nows that it is behind a 9T and therefore cannot guess from whataddress&port

; its reJuest will be sent. steris% will always honor the 8rport8 parameter if it is

; sent. The force:rport setting causes steris% to always send responses bac% to the

; address&port from which it received reJuests; even if the other side doesn8t

support; adding the 8rport8 parameter.

;

; 8comedia *TP handling8 refers to the techniJue of sending *TP to the port that the

; the other endpoint8s *TP arrived from, and means 8connection)oriented media8. Thisis

; only partially related to *04 "H"? which was referred to as 4/#+3I while it was in

; draft form. This method is used to accomodate endpoints that may be located behind

; 9T devices, and as such the address&port they tell steris% to send *TP pac%ets to

; for their media streams is not the actual address&port that will be used on thenearer

; side of the 9T.

;; IT IS I#P/*T9T T/ 9/T+ 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 steris% will respond to different ports for defined and

; undefined peers. 0or this reason it is recommended to /9-M 3+0I9+ 9T S+TTI9NS I9TA+

; N+9+*- S+4TI/9. 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 sectionwill

; be discoverable.

;

; In addition to these settings, steris% CalwaysC uses 8symmetric *TP8 mode asdefined by

; *04 "DGH; steris% will always send *TP pac%ets from the same port number itexpects

; to receive them on.

;

; The IP address used for media $audio, video, and text' in the S3P can also beoverridden by using

; the media:address configuration option. This is only applicable to the generalsection and

; can not be set per)user or per)peer.

;

; media:address = HF.HG."F.H

;

; Through the use of the res:stun:monitor module, steris% has the ability to detectwhen the

; perceived external networ% address has changed.


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; generate all outbound registrations on a networ% change, use the option below todisable

; this feature.

;

; subscribe:networ%:change:event = yes ; on by default

;

; I4+&ST9&T*9 usage can be disabled globally or on a per)peer basis using theicesupport

; configuration option.


36/51


; $reinvite' but only when the peer where the mediais being

; sent is %nown to not be behind a 9T $as the *TPcore can

; determine it based on the apparent IP address themedia

; arrives from'.

;directmedia=update ; Met a third option... use P3T+ for media pathredirection,

; instead of I9(IT+. This can be combined with8nonat8, as

; 8directmedia=update,nonat8. It implies 8yes8.

;directmedia=outgoing ;


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; session if the version number changes. This optionwill

; force asteris% to ignore the S3P session versionnumber

; and treat all S3P data as new data. This isreJuired

; for devices that send us non standard S3P pac%ets ; $observed with #icrosoft /4S'. y default thisoption is

; off.

;sdpsession=steris% PB ; llows you to change the S3P session name string,$s='

; -i%e the useragent parameter, the default useragent string

; also contains the steris% version.

;sdpowner=root ; llows you to change the username field in the S3Powner string, $o='

; This field #ST 9/T contain spaces

;encryption=no ;


38/51


; the will be set to database via realtime.

; If not present, defaults to 8yes8. 9ote1 realtimepeers will

; probably not function across reloads in the waythat you expect, if

; you turn this option off.

;rtautoclear=yes ; uto)+xpire friends created on the fly on the sameschedule

; as if it had 5ust registered6 $yes7no7Rseconds'

; If set to yes, when the registration expires, thefriend will

; vanish from the configuration until reJuestedagain. If set

; to an integer, friends expire within this number ofseconds

; instead of the registration interval.

;ignoreregexpire=yes ; +nabling this setting has two functions1

;

; 0or non)realtime peers, when their registrationexpires, the

; information will :not: be removed from memory orthe steris% database

; if you attempt to place a call to the peer, theexisting information

; will be used in spite of it having expired

;

; 0or realtime peers, when the peer is retrieved fromrealtime storage,

; the registration information will be usedregardless of whether

; it has expired or not; if it expires while the

realtime peer

; is still in memory $due to caching or otherreasons', the

; information will not be removed from realtimestorage

;))))))))))))))))))))))))))))))))))))))))) SIP 3/#I9 SPP/*T))))))))))))))))))))))))

; Incoming I9(IT+ and *+0+* messages can be matched against a list of 8allowed8

; domains, each of which can direct the call to a specific context if desired.

; y default, all domains are accepted and sent to the default context or the

; context associated with the user&peer placing the call.

; *+NIST+* to non)local domains will be automatically denied if a domain; list is configured.

;

; 3omains can be specified using1

; domain=Rdomain>,Rcontext@

; +xamples1

; domain=myasteris%.dom

; domain=customer.com,customer)context

;

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; In addition, all the 8default8 domains associated with a server should be

; added if incoming reJuest filtering is desired.

; autodomain=yes

;

; To disallow reJuests for domains not serviced by this server1

; allowexternaldomains=no

;domain=mydomain.tld,mydomain)incoming

; dd domain and configure incoming context

; for external calls to this domain

;domain=H.F.." ; dd IP address as local domain

; Mou can have several LdomainL settings

;allowexternaldomains=no ; 3isable I9(IT+ and *+0+* to non)local domains

; 3efault is yes

;autodomain=yes ; Turn this on to have steris% add local host

; name and local IP to domain list.

; fromdomain=mydomain.tld ;


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; 5bmaxsie = FEE ; #ax length of the 5itterbuffer in milliseconds.

; 5bresyncthreshold = HEEE ; Xump in the frame timestamps over which the5itterbuffer is

; resynchronied. seful to improve the Juality of the

voice, with ; big 5umps in&bro%en timestamps, usually sent fromexotic devices

; and programs. 3efaults to HEEE.

; 5bimpl = fixed ; Xitterbuffer implementation, used on the receivingside of a SIP

; channel. Two implementations are currently available) LfixedL

; $with sie always eJuals to 5bmaxsie' and LadaptiveL$with

; variable sie, actually the new 5b of IBF'. 3efaultsto fixed.

; 5btargetextra = "E ; This option only affects the 5b when 85bimpl =adaptive8 is set.

; The option represents the number of milliseconds bywhich the new 5itter buffer

; will pad its sie. the default is "E, so withoutmodification, the new

; 5itter buffer will set its sie to the 5itter valueplus "E milliseconds.

; increasing this value may help if your networ%normally has low 5itter,

; but occasionally has spi%es.

; 5blog = no ; +nables 5itterbuffer frame logging. 3efaults to LnoL.

;)))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))

>authentication@

; Nlobal credentials for outbound calls, i.e. when a proxy challenges your

; steris% server for authentication. These credentials override

; any credentials in peer&register definition if realm is matched.

;

; This way, steris% can authenticate for outbound calls to other

; realms. peer@ definitions

; Peer auth= override all other authentication settings if we match on realm

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;))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))

; 3+(I4+ 4/90IN*TI/9

;

; SIP entities have a 8type8 which determines their roles within steris%.

; C 0or entities with 8type=peer81

; 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 wor%. This means calls made from either direction won8t wor% if

; The peer is unregistered while host=dynamic or if the host is otherise not set to

; the correct IP of the sender.

; C 0or entities with 8type=user81

; steris% users handle inbound calls only $meaning they call steris%, steris%can8t

; call them' and are matched by their authoriation information $authname andsecret'.

; steris% doesn8t rely on their IP and will accept calls regardless of the hostsetting

; as long as the incoming SIP invite authories successfully.; C 0or entities with 8type=friend81

; steris% will create the entity as both a friend and a peer. steris% will accept

; calls from friends li%e it would for users, reJuiring only that the authoriation

; matches rather than the IP address. Since it is also a peer, a friend entity can

; be called as long as its IP is %nown to steris%. In the case of host=dynamic,

; this means it is necessary for the entity to register before steris% can callit.

;

; se remotesecret for outbound authentication, and secret for authenticating

; inbound reJuests. 0or historical reasons, if no remotesecret is supplied for an

; outbound registration or call, the secret will be used.

;; 0or device names, we recommend using only a), numerics $E)D' and underscore

;

; 0or local phones, type=friend wor%s most of the time

;

; If you have one)way audio, you probably have 9T problems.

; If steris% is on a public IP, and the phone is inside of a 9T device

; you will need to configure nat option for those phones.

; lso, turn on Jualify=yes to %eep the nat session open

;

; 4onfiguration options available

; ))))))))))))))))))))

; context; callingpres

; permit

; deny

; secret

; md?secret

; remotesecret

; transport

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; dtmfmode

; directmedia

; nat

; callgroup

; pic%upgroup

; language

; allow

; disallow

; insecure

; trustrpid

; progressinband

; promiscredir

; useclientcode

; accountcode

; setvar

; callerid

; amaflags

; callcounter

; busylevel

; allowoverlap

; allowsubscribe

; allowtransfer

; ignoresdpversion

; subscribecontext

; template

; videosupport

; maxcallbitrate

; rfcF!compensate

; mailbox

; session)timers; session)expires

; session)minse

; session)refresher

; t!pt:usertpsource

; regexten

; fromdomain

; fromuser

; host

; port

; Jualify

; %eepalive

; defaultip; defaultuser

; rtptimeout

; rtpholdtimeout

; sendrpid

; outboundproxy

; rfcF!compensate

; callbac%extension

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; registertrying

; timertH

; timerb

; JualifyfreJ

; t!pt:usertpsource

; contactpermit ; -imit what a host may register as $a neat tric%

; contactdeny ; is to register at the same IP as a SIP provider,

; contactacl ; then call oneself, and get redirected to that

; ; same location'.

; directmediapermit

; directmediadeny

; directmediaacl

; unsolicited:mailbox

; use:J!?E:reason

; maxforwards

; encryption

; description ; sed to provide a description of the peer in consoleoutput

; dtlsenable

; dtlsverify

; dtlsre%ey

; dtlscertfile

; dtlsprivate%ey

; dtlscipher

; dtlscafile

; dtlscapath

; dtlssetup

;

;))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))))

; 3T-S)S*TP 4/90IN*TI/9

;

; 3T-S)S*TP support is available if the underlying *TP engine in use supports it.

;

; dtlsenable = yes ; +nable or disable 3T-S)S*TP support

; dtlsverify = yes ; (erify that the provided peer certificate isvalid

; dtlsre%ey = GE ; Interval at which to renegotiate the T-Ssession and re%ey the S*TP session

; ; If this is not set or the value provided is Ere%eying will be disabled

; dtlscertfile = file ; Path to certificate file to present

; dtlsprivate%ey = file ; Path to private %ey for certificate file; dtlscipher = RSS- cipher string ; 4ipher to use for T-S negotiation

; ; list of valid SS- cipher strings can befound at1

; ;http1&&www.openssl.org&docs&apps&ciphers.html4IPA+*:ST*I9NS

; dtlscafile = file ; Path to certificate authority certificate

; dtlscapath = path ; Path to a directory containing certificateauthority certificates

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;callbac%extension=HF ; *egister with this server and reJuire callscoming bac% to this extension

;transport=udp,tcp ; This sets the transport type to udp for outgoing,and will

; ; accept both tcp and udp. 3efault is udp. Thefirst transport

; ; listed will always be used for outgoingconnections.

;unsolicited:mailbox="EH???FFDD ; If the remote SIP server sends an unsolicited #public)phone@$K,basic)options' ; another template inheriting basic)options

directmedia=yes

>my)codecs@$K' ; a template for my preferred codecs

disallow=all

allow=ilbc

allow=gFD

allow=gsm

allow=gF

allow=ulaw

; /r, more simply1

;allow=Kall,ilbc,gFD,gsm,gF,ulaw

>ulaw)phone@$K' ; and another one for ulaw)only

disallow=all

allow=ulaw

; gain, more simply1

;allow=Kall,ulaw

; and finally instantiate a few phones

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;

; >FH@$natted)phone,my)codecs'

; secret = pee%aboo

; >FH"@$natted)phone,ulaw)phone'

; secret = not:very:secret

; >FHG@$public)phone,ulaw)phone'

; secret = not:very:secret:either

; ...

;

; Standard configurations not using templates loo% li%e this1

;

;>grandstreamH@

;type=friend

;context=from)sip ;


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; Turn off silence suppression in B)-ite $LTransmit SilenceL=M+S'K

; 9ote that Blite sends 9T %eep)alive pac%ets, so Jualify=yes is not needed

;type=friend

;regexten=HF" ; snom@

;type=friend ; 0riends place calls and receive calls

;context=from)sip ; 4ontext for incoming calls from this user

;secret=blah

;subscribecontext=localextensions ; /nly allow SS4*I+ for local extensions

;language=de ; se Nerman prompts for this user

;host=dynamic ; This peer register with us

;dtmfmode=inband ; 4hoices are inband, rfcF!, or info

;defaultip=HDF.HG!.E.?D ; IP used until peer registers

;mailbox=HF"context,F"? ; #ailbox$)es' for message waiting indicator

;subscribemwi=yes ; /nly send notifications if this phone

; subscribes for mailbox notification

;vmexten=voicemail ; dialplan extension to reach mailbox

; sets the #essage)ccount in the #polycom@

;type=friend ; 0riends place calls and receive calls

;context=from)sip ; 4ontext for incoming calls from this user

;secret=blahpoly

;host=dynamic ; This peer register with us

;dtmfmode=rfcF! ; 4hoices are inband, rfcF!, or info

;defaultuser=polly ; sername to use in I9(IT+ until peer registers

;defaultip=HDF.HG!."E.HF ; 9ormally you do 9/T need to set this parameter

;disallow=all

;allow=ulaw ; dtmfmode=inband only wor%s with ulaw or alawK

;progressinband=no ; Polycom phones don8t wor% properly with LneverL

;>pingtel@

;type=friend

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;secret=blah

;host=dynamic

;insecure=port ; llow matching of peer by IP address without

; matching port number

;insecure=invite ; 3o not reJuire authentication of incoming I9(IT+s

;insecure=port,invite ; $both'

;Jualify=HEEE ; 4onsider it down if it8s H second to reply

; Aelps with 9T session

; Jualify=yes uses default value

;JualifyfreJ=GE ; Oualification1 Aow often to chec% for the

; host to be up in seconds

; Set to low value if you use low timeout for

; 9T of 3P sessions

;

; 4all group and Pic%up group should be in the range from E to G

;

;callgroup=H,)" ;


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; an attended transfer.

;>preH")asteris%@

;type=friend

;secret=digium

;host=dynamic

;rfcF!compensate=yes ; 4ompensate for pre)H." 3T#0 transmission fromanother steris% machine.

; Mou must have this turned on or 3T#0 reception willwor% improperly.

;t!pt:usertpsource=yes ; se the source IP address of *TP as the destinationIP address for 3PT- pac%ets

; if the nat option is enabled. If a single *TPpac%et is received steris% will %now the

; external IP address of the remote device. If portforwarding is done at the client side

; then 3PT- will flow to the remote device.

/ser con options. Peer conuration.

8 -------------------- -------------------

8 conte6t conte6t

8 callinpres callinpres

8 permit permit

8 +eny +eny

8 secret secret

8 m+#secret m+#secret

8 +tmfmo+e +tmfmo+e

8 canreinvite canreinvite

8 nat nat

8 callroup callroup

8 pickuproup pickuproup

8 lanuae lanuae

8 allo5 allo5

8 +isallo5 +isallo5

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8 insecure insecure

8 trustrpi+ trustrpi+

8 proressin7an+ proressin7an+

8 promiscre+ir promiscre+ir

8 useclientco+e useclientco+e

8 accountco+e accountco+e

8 setvar setvar

8 calleri+ calleri+

8 amaFas amaFas

8 call-limit call-limit

8 allo5overlap allo5overlap

8 allo5su7scri7e allo5su7scri7e

8 allo5transfer allo5transfer

8 su7scri7econte6t su7scri7econte6t

8 vi+eosupport vi+eosupport

8 ma6call7itrate ma6call7itrate

8 rfc2(33compensate mail7o6

8 t3(ptusertpsource username

8 from+omain

8 ree6ten

8 fromuser

8 host

8 port

8 @ualify

8 +efaultip

8 rtptimeout

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8 rtphol+timeout

8 sen+rpi+

8 out7oun+pro6y

8 rfc2(33compensate

8 t3(ptusertpsource

8 contactpermit 8 0imit 5hat a host may reister as =a neat

trick

8 contact+eny 8 is to reister at the same !P as a S!P

provi+er

8 8 then call oneself an+ et re+irecte+ to that

8 8 same location?"

sip extension

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