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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>
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<h4>Changes (2)</h4>
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<tr><td class="diff-unchanged" >Fishook: We need a single spot to compile the definitive guide to Asterisk's SIP capabilities. <br> <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">h2. Configuration <br> <br>{info} <br>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 <br>{info} <br> <br>{noformat} <br> <br>[general] <br>context=default <br>allowguest=no <br>math_auth_username=yes <br>allowoverlap=yes <br>allowtransfer=yes <br>realm=asterisk <br>domainsasrealm=no <br>udpbindaddr=0.0.0.0:5060 <br>disallowed_methods= <br>tcpenable=yes <br>tcpbindaddr=0.0.0.0:5060 <br>tlsenable=yes <br>tlsbindaddr=0.0.0.0:5061 <br>srvlookup=yes <br>pedantic=yes <br>tos_sip=cs3 <br>tos_audio=ef <br>tos_video=af41 <br>tos_text=af41 <br>cos_sip=3 <br>cos_audio=5 <br>cos_video=4 <br>cos_text=3 <br>maxexpiry=3600 <br>minexpiry=60 <br>defaultexpiry=120 <br>mwiexpiry=3600 <br>maxforwards=70 <br>qualifyfreq=60 <br>qualifygap=100 <br>qualifypeers=1 <br>notifymimetype=text/plain <br>buggymwi=no <br>mwi_from= <br>vmexten=asterisk <br>preferred_codec_only=yes <br>disallow=all <br>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 <br>mohinterpret=default <br>mohsuggest=default <br>parkinglot=plaza <br>language=en <br>relaxdtmf=no <br>trustrpid=yes <br>sendrpid=yes <br>rpid_update=yes <br>prematuremedia=no <br>progressinband=never <br>useragent=Asterisk PBX <br>promiscredir=no <br>usereqphone=no <br>dtmfmode=rfc2833 <br>compactheaders=yes <br>videosupport=yes <br>maxcallbitrate=384 <br>callevents=yes <br>authfailureevents=yes <br>alwaysauthreject=yes <br>auth_options_requests=yes <br>g726nonstandard=no <br>;outboundudpproxy=udp://proxy.provider.domain:5060 <br>;outboundtcpproxy=tcp://proxy.provider.domain:5060 <br>;ouboundtlsproxy=tls://proxy.provider.domain:5061 <br>matchexternaddrlocally=no <br>dynamic_exclude_static=yes <br>contactdeny=0.0.0.0/0.0.0.0 <br>contactpermit=192.168.0.0/255.255.0.0 <br>engine=asterisk <br>regcontext=sipregistrations <br>regextenonqualify=no <br>shrinkcallerid=yes <br>use_q850_reason=no <br>;tlscertfile=/path/to/certificate.pem <br>;tlsprivatekey=/path/to/private.pem <br>;tlscafile=/path/to/certificate.ca <br>;tlscadir=/path/to/ca/dir <br>tlsdontverifyserver=no <br>tlscipher=ALL <br>tlsclientmethod=tlsv1 <br>t1min=100 <br>timert1=500 <br>timerb=32000 <br>rtptimeout=60 <br>rtpholdtimeout=300 <br>rtpkeepalive=0 <br>session-timers=originate <br>session-expires=600 <br>session-minse=90 <br>session-refresher=uas <br>sipdebug=no <br>recordhistory=yes <br>dumphistory=yes <br>allowsubscribe=yes <br>subscribecontext=default <br>notifyringing=yes <br>notifyhold=yes <br>notifycid=yes <br>callcounter=yes <br>t38pt_udptl=yes,fec,maxdatagram=400 <br>faxdetect=yes <br>;register=>[peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] <br>;register=>mypeer?tls://myuser@domain:password:authuser@my.host.com:5060/myextension~600 <br>registertimeout=20 <br>registerattempts=10 <br>;mwi => user[:secret[:authuser]]@host[:port]/mailbox <br>;mwi=>1234:password:authuser@myauthportprovider.com:6969/1234 <br>;localnet=192.168.0.0/255.255.0.0 <br>;externaddr=12.34.56.78:9900 <br>;externtcpport=9900 <br>;externtslport=12600 <br>;externhost=my.host.name <br>;externrefresh=180 <br>nat=no <br>;media_address=172.16.42.1 <br>subscribe_network_change_event=yes <br>directmedia=yes <br>directrtpsetup=no <br>;directmediadeny=0.0.0.0/0 <br>;directmediapermit=192.168.0.0/255.255.0.0 <br>ignoresdpversion=no <br>sdpsession=Asterisk PBX <br>sdpowner=root <br>encryption=no <br>rtcachefriends=yes <br>rtsavesysname=no <br>rtupdate=yes <br>rtautoclear=yes <br>ignoreregexpire=no <br>autodomain=yes <br>allowexternaldomains=no <br>;domain=mydomain.tld,mydomain-incoming <br>;fromdomain=mydomain.tld <br>snom_aoc_enabled=yes <br>jbenable=no <br>jbforce=no <br>jbmaxsize=200 <br>jbresyncthreshold=1000 <br>jbimpl=fixed <br>jbtargetextra=40 <br>jblog=no <br> <br>[authentication] <br>;auth=<user>:<secret>@<realm> <br>;auth=<user>#<md5secret>@<realm> <br> <br>[mypeer] <br>context=default <br>callingpres=allowed_passed_screen <br>deny=0.0.0.0/0.0.0.0 <br>permit=192.168.0.60/255.255.255.0 <br>;secret=AbCdEfG#$! <br>;md5secret= <br>;remotesecret= <br>transport=udp,tcp,tls <br>encrption=no <br>dtmfmode=rfc2833 <br>directmedia=yes <br>nat=no <br>callgroup=1 <br>pickupgroup=1 <br>language=en <br>disallow=all <br>allow=ulaw:20,alaw:20,g722 <br>insecure=no <br>trustrpid=yes <br>sendrpid=yes <br>progressinband=never <br>promiscredir=no <br>useclientcode=yes <br>;setvar= <br>callerid=My Peer <800 555 123> <br>;amaflags= <br>callcounter=yes <br>busylevel=2 <br>allowoverlap=yes <br>allowsubscribeyes <br>allowtransfer=yes <br>ignoresdpversion=no <br>subscribecontext=default <br>;template= <br>videosupport=yes <br>maxcallbitrate=384 <br>rfc2833compensate=no <br>mailbox=mypeer@default <br>session-timers=originate <br>session-expires=600 <br>session-minse=90 <br>session-refresher=uas <br>t38pt_usertpsource=no <br>;regexten=1234 <br>;fromdomain=provider.sip.domain <br>;fromuser=yourusername <br>;host=dynamic <br>;port=5060 <br>qualify=yes <br>;defaultip= <br>;defaultuser= <br>rtptimeout=60 <br>rtphodltimeout=300 <br>;outboundproxy=my.proxy.tld <br>;callbackextension=1234 <br>registertrying=100 <br>timert1=500 <br>timerb=32000 <br>qualifyfreq=60 <br>contactdeny=0.0.0.0/0.0.0.0 <br>contactpermit=192.168.0.0/255.255.0.0 <br>;diretmediadeny=0.0.0.0/0 <br>;directmediapermit=192.168.0.0/255.255.0.0 <br>;unsolicited_mailbox=1234@SIP_Remote <br>use_q850_reason=no <br>maxforwards=70 <br>encryption=no <br>{noformat} <br></td></tr>
<tr><td class="diff-added-lines" style="background-color: #dfd;">Content coming. WIP beginning here: <br>[~mdavenport:1.8 SIP Scratchpad] <br></td></tr>
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</div> <h4>Full Content</h4>
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<p>Fishook: We need a single spot to compile the definitive guide to Asterisk's SIP capabilities.</p>
<p>Content coming. WIP beginning here:<br/>
<a href="/wiki/display/~mdavenport/1.8+SIP+Scratchpad" title="1.8 SIP Scratchpad">1.8 SIP Scratchpad</a></p>
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