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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+SIP+Connections">Asterisk SIP Connections</a></h2>
<h4>Page <b>edited</b> by <a href="https://wiki.asterisk.org/wiki/display/~mdavenport">Malcolm Davenport</a>
</h4>
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<h4>Changes (1)</h4>
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<tr><td class="diff-snipped" >...<br></td></tr>
<tr><td class="diff-unchanged" >language=en <br>disallow=all <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-changed-words">allow=ulaw:20,alaw:20,g722<span class="diff-deleted-chars"style="color:#999;background-color:#fdd;text-decoration:line-through;">:20</span></span> <br></td></tr>
<tr><td class="diff-unchanged" >insecure=no <br>trustrpid=yes <br></td></tr>
<tr><td class="diff-snipped" >...<br></td></tr>
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</div> <h4>Full Content</h4>
<div class="notificationGreySide">
<p>Fishook: We need a single spot to compile the definitive guide to Asterisk's SIP capabilities.</p>
<h2><a name="AsteriskSIPConnections-Configuration"></a>Configuration</h2>
<div class='panelMacro'><table class='infoMacro'><colgroup><col width='24'><col></colgroup><tr><td valign='top'><img src="/wiki/images/icons/emoticons/information.gif" width="16" height="16" align="absmiddle" alt="" border="0"></td><td>For starters, here are all of the configuration options for the general section and the configuration of a peer, stripped of comments. It's not a short list</td></tr></table></div>
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<pre>
[general]
context=default
allowguest=no
math_auth_username=yes
allowoverlap=yes
allowtransfer=yes
realm=asterisk
domainsasrealm=no
udpbindaddr=0.0.0.0:5060
disallowed_methods=
tcpenable=yes
tcpbindaddr=0.0.0.0:5060
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
srvlookup=yes
pedantic=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=af41
cos_sip=3
cos_audio=5
cos_video=4
cos_text=3
maxexpiry=3600
minexpiry=60
defaultexpiry=120
mwiexpiry=3600
maxforwards=70
qualifyfreq=60
qualifygap=100
qualifypeers=1
notifymimetype=text/plain
buggymwi=no
mwi_from=
vmexten=asterisk
preferred_codec_only=yes
disallow=all
allow=ulaw:20,alaw:20,g719,siren14,siren7,g722,slin16,slin:20,g726:20,g726aal2:20,adpcm:20,gsm:20,ilbc:30,speex16,speex:20,lpc10:20,g729:20,g723:30,h264,mpeg4,h263p,h263,h261,png,jpeg,t140,red
mohinterpret=default
mohsuggest=default
parkinglot=plaza
language=en
relaxdtmf=no
trustrpid=yes
sendrpid=yes
rpid_update=yes
prematuremedia=no
progressinband=never
useragent=Asterisk PBX
promiscredir=no
usereqphone=no
dtmfmode=rfc2833
compactheaders=yes
videosupport=yes
maxcallbitrate=384
callevents=yes
authfailureevents=yes
alwaysauthreject=yes
auth_options_requests=yes
g726nonstandard=no
;outboundudpproxy=udp://proxy.provider.domain:5060
;outboundtcpproxy=tcp://proxy.provider.domain:5060
;ouboundtlsproxy=tls://proxy.provider.domain:5061
matchexternaddrlocally=no
dynamic_exclude_static=yes
contactdeny=0.0.0.0/0.0.0.0
contactpermit=192.168.0.0/255.255.0.0
engine=asterisk
regcontext=sipregistrations
regextenonqualify=no
shrinkcallerid=yes
use_q850_reason=no
;tlscertfile=/path/to/certificate.pem
;tlsprivatekey=/path/to/private.pem
;tlscafile=/path/to/certificate.ca
;tlscadir=/path/to/ca/dir
tlsdontverifyserver=no
tlscipher=ALL
tlsclientmethod=tlsv1
t1min=100
timert1=500
timerb=32000
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=0
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas
sipdebug=no
recordhistory=yes
dumphistory=yes
allowsubscribe=yes
subscribecontext=default
notifyringing=yes
notifyhold=yes
notifycid=yes
callcounter=yes
t38pt_udptl=yes,fec,maxdatagram=400
faxdetect=yes
;register=>[peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;register=>mypeer?tls://myuser@domain:password:authuser@my.host.com:5060/myextension~600
registertimeout=20
registerattempts=10
;mwi => user[:secret[:authuser]]@host[:port]/mailbox
;mwi=>1234:password:authuser@myauthportprovider.com:6969/1234
;localnet=192.168.0.0/255.255.0.0
;externaddr=12.34.56.78:9900
;externtcpport=9900
;externtslport=12600
;externhost=my.host.name
;externrefresh=180
nat=no
;media_address=172.16.42.1
subscribe_network_change_event=yes
directmedia=yes
directrtpsetup=no
;directmediadeny=0.0.0.0/0
;directmediapermit=192.168.0.0/255.255.0.0
ignoresdpversion=no
sdpsession=Asterisk PBX
sdpowner=root
encryption=no
rtcachefriends=yes
rtsavesysname=no
rtupdate=yes
rtautoclear=yes
ignoreregexpire=no
autodomain=yes
allowexternaldomains=no
;domain=mydomain.tld,mydomain-incoming
;fromdomain=mydomain.tld
snom_aoc_enabled=yes
jbenable=no
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jbtargetextra=40
jblog=no
[authentication]
;auth=<user>:<secret>@<realm>
;auth=<user>#<md5secret>@<realm>
[mypeer]
context=default
callingpres=allowed_passed_screen
deny=0.0.0.0/0.0.0.0
permit=192.168.0.60/255.255.255.0
;secret=AbCdEfG#$!
;md5secret=
;remotesecret=
transport=udp,tcp,tls
encrption=no
dtmfmode=rfc2833
directmedia=yes
nat=no
callgroup=1
pickupgroup=1
language=en
disallow=all
allow=ulaw:20,alaw:20,g722
insecure=no
trustrpid=yes
sendrpid=yes
progressinband=never
promiscredir=no
useclientcode=yes
;setvar=
callerid=My Peer <800 555 123>
;amaflags=
callcounter=yes
busylevel=2
allowoverlap=yes
allowsubscribeyes
allowtransfer=yes
ignoresdpversion=no
subscribecontext=default
;template=
videosupport=yes
maxcallbitrate=384
rfc2833compensate=no
mailbox=mypeer@default
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas
t38pt_usertpsource=no
;regexten=1234
;fromdomain=provider.sip.domain
;fromuser=yourusername
;host=dynamic
;port=5060
qualify=yes
;defaultip=
;defaultuser=
rtptimeout=60
rtphodltimeout=300
;outboundproxy=my.proxy.tld
;callbackextension=1234
registertrying=100
timert1=500
timerb=32000
qualifyfreq=60
contactdeny=0.0.0.0/0.0.0.0
contactpermit=192.168.0.0/255.255.0.0
;diretmediadeny=0.0.0.0/0
;directmediapermit=192.168.0.0/255.255.0.0
;unsolicited_mailbox=1234@SIP_Remote
use_q850_reason=no
maxforwards=70
encryption=no
</pre>
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