; 2003-04-24 05:06 GMT jtodd@loligo.com ; ; This is the sip.conf file for John Todd's Asterisk ; server. Asterisk can be found on http://www.asterisk.org/ ; ; More recent versions of this file can be found on: ; http://www.loligo.com/asterisk/ ; ; ; SIP Configuration for Asterisk ; [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip ; Default for incoming calls callerid=No CallID ; These register statements are to REGISTER my Asterisk server ; with certain accounts on remote SIP servers. The inoc-dba, ; FWD, and iconnect (deltathree) registries work, the one for ; coloco does not, due to Vocal problems. ; ; These register= statements must be in context [general] ; ; Note that registering against (some?) Vocal 1.4 servers fails. Bug. ; ; All inbound SIP calls end up in context [from-sip] in extensions.conf ; with the extension of the called number as a match pattern ; ; You can have usernames with "@" signs in them, as the line is parsed ; from rear to front (Thanks, Mark!) ; register=32767:foo@inoc-dba.pch.net/32767 register=14155551212:9876@sipauth.deltathree.com/14155551212 register=11001:pass-da-word@fwd.pulver.com/11001 register=jtodd:wordtopass@iptel.org/1234567 ; Other general settings for the SIP channels that are used ; ; tos=['lowdelay', 'throughput', 'reliability', 'mincost', or 'none'] ; Sets Type Of Service flags (?) ; ; maxexpirey=3600 ; Max duration (seconds) of incoming registration allowed ; ; defaultexpirey=120 ; Default length of incoming/outoing registrations ; ; ; Under each peer/user/friend, you can specify some additional notes: ; tos=['lowdelay', 'throughput', 'reliability', 'mincost', or 'none'] ; ; The "iconnect" SIP peer is the outbound leg of the iconnecthere.com ; SIP gateway service. Note that the username and password below ; are different than the "register=" line above. I don't exactly ; know why iconnecthere.com does it this way, but that's fine with ; me. I only use the "iconnect" peer below to pass outgoing calls ; to their service. Inbound calls are handled out of [general] ; ; The "dtmfmode=inband" is due to the issues that iconnect seems to ; have with RFC2833 DTMF passing, so I send them the tones. I don't ; know if this works 100% or not; "untested" as of 2003-04-07. ; [iconnect] type=friend secret=9876 username=52671573 host=sipauth.deltathree.com dtmfmode=inband ; The "fwd" SIP peer is Free World Dialup, run by Jeff Pulver. ; Personally, I haven't found much use for it, but other people ; swear by it, so I've included it... ; [fwd] type=friend secret=pass-da-word username=11001 host=fwd.pulver.com ; The "iptel" SIP peer is a service provided by IPTEL.org as a ; SIP directory. ; [iptel] type=friend secret=wordtopass username=jtodd host=iptel.org ; The "coloco" SIP peer is a Vocal system at Coloco, Inc. in ; Laurel, MD. They are a SIP gateway provider, and I have ; an account with them for local Maryland SIP dialtone. ; http://www.coloco.com/ ; ; This simply dumps calls at a Cisco 3640 via SIP. There ; is no username/password required, since this is simply a ; SIP gateway, and not a proxy. Protection provided by ; ACLs on the router. ; ; [coloco] context=coloco type=friend host=198.180.62.154 dtmf=rfc2833 ; INOC-DBA is a terribly useful SIP-only gateway for ; AS# holders. If you don't know what an AS# is, and if you ; don't have one, this config subset won't be useful to you. ; Contact Woody for an extension assignment. ; ; Note: bogus IP address; get from woody yourself ; [inoc-dba] type=friend host=230.61.218.90 username=32767 secret=yoyoyopassword ; My SIP phones in the house/office are listed below ; ; By my own convention (and to save my sanity) I have given ; all the ATA-186 devices in my network "extensions" that ; are numeric. I picked "2200" to start for historical ; laziness. Again, for my sanity, I also made the usernames ; on the phone identical to the extension number of the ; channel. ; ; I distribute IP addresses with a DHCP server to these phones, ; on the network 204.91.156.0/24 or behind a NAT (for some) ; ; All the Cisco ATA-186 units I have are on "real" IP addresses, ; and I haven't experimented with NAT yet. NAT has been a real ; PITA with the ATA's, and I don't relish having to make it work. ; ; Anytime a call is dialed from "2203" or "2204", it will be ; parsed by the context [intern] in the extensions.conf file ; ; If you want the message waiting light to light up, or stutter ; dialtone, you can set mailbox=[mbox#] - note you can also ; set a comma-delimited list of mailboxes if you want multiples. ; ; You can use "permit=x.x.x.x/yy" lists for hosts that are dynamic ; ; You don't need to register SIP peers that are static (host=) ; ; Setting "nat=1" is required to get the "Via:" headers to be configured ; correctly so that the Ciscos (and others?) will re-register when ; they are behind a NAT. If nat=1 is set, it will not harm non-NAT ; connections on the Ciscos, but there are reports of other phones ; (pingtel) not working if you have nat=1 set and the phone is not ; behind a NAT. ; [2203] type=friend username=2203 secret=c98wh320e host=204.91.156.6 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 [2204] type=friend username=2204 secret=c82ncc8e9w mailbox=2203 host=204.91.156.6 context=intern canreinvite=yes dtmfmode=rfc2833 [2205] type=friend username=2205 secret=nonc282dwa mailbox=2205 host=dynamic context=intern dtmfmode=rfc2833 nat=1 [2206] type=friend username=2206 secret=cno2o093 mailbox=2206 host=dynamic context=intern canreinvite=yes nat=1 [2207] type=friend username=2207 secret=mhh32c02n host=dynamic context=intern canreinvite=yes nat=1