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some warnings in docu

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Thomas Ries
2007-05-24 18:39:23 +00:00
parent c97532fdbb
commit 697790c151
1 changed files with 192 additions and 43 deletions
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doc/siproxd_guide.sgml
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@@ -17,7 +17,7 @@
<email>tries@users.sourceforge.net</email>
</address>
<copyright>
<year>2005</year>
<year>2005-2007</year>
<holder>Thomas Ries</holder>
</copyright>
<legalnotice>
@@ -37,6 +37,12 @@
<authorinitials>tries@users.sourceforge.net</authorinitials>
<revremark>Comment on Asterisk Scenario</revremark>
</revision>
<revision>
<revnumber>0.3</revnumber>
<date>2007-05-15</date>
<authorinitials>tries@users.sourceforge.net</authorinitials>
<revremark>New Asterisk Config Files</revremark>
</revision>
</revhistory>
</bookinfo>
<toc></toc>
@@ -44,10 +50,34 @@
<!-- We are done with the preliminaries, now we can start with
the body of the document -->
<!-- Chapter 1: Overview -->
<chapter label="" id="README">
<?dbhtml filename="siproxd_guide_c0.html">
<title>README</title>
<para>Important information, please read me!</para>
<sect1 label="">
<?dbhtml filename="siproxd_guide_c0s1.html">
<title>Important / Warning</title>
<para>As it still happens that people try to mix different NAT
traversal techologies together with siproxd I'll put some words
here:
<itemizedlist mark='bullet'>
<listitem><para>Do NOT USE anything like an STUN Server together
with siproxd.</para></listitem>
<listitem><para>Do NOT USE any additional techologies trying to
help in NAT traversal (additional firewall modules like
ip_nat_sip.ko or whatever fancy stuff may tempt you).
</para></listitem>
</itemizedlist>
If you do not follow the above rules, those other
"helping technologies" WILL DO CONFLICT with siproxd and result
in a mess.</para>
</sect1>
<!-- Chapter 1: Overview -->
<chapter label="1" id="Overview">
<?dbhtml filename="siproxd_guide_c1.html">
<title>Overview</title>
<para>Siproxd is an proxy/masquerading daemon for the SIP protocol.
It handles registrations of SIP clients on a private IP network
@@ -924,62 +954,181 @@ iptables -A INPUT -m udp -p udp -i ppp0 --dport 7070:7080 -j ACCEPT
<para>Asterisk configuration (SIP related part):</para>
<para>Note: Very important are the fromuser and fromdomain
keywords in the client section. They are required to have
Asterisk send the correct From headers in SIP dialogs.</para>
Asterisk send the correct From headers in SIP dialogs.
The used Asterisk version is 'SVN-branch-1.4-r62331M'.</para>
<para>With newer Asterisk versions, it is no longer required
to have a separate REGISTER definition, this can be made
implicit in the SIP trunk config.</para>
<screen>
sip.conf:
; sip.conf:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
defaultexpirey = 900
; codecs
useragent = PBX ; NOTE: some providers (e.g sipcall.ch) do simply
; not work with the default "AsteriskPBX"
; UA String.
; Network Settings
nat=never
localnet = 10.0.0.0/24
domain = 10.0.0.10
; Codecs
disallow=all
allow=gsm ; 13 Kbps
allow=ulaw ; 64 Kbps
allow=alaw ; 64 Kbps
autoframing = yes
; SIP Trunk to sipphone.com
; the SIP number is taken randomly for this example
register=17476691234:&lt;password>@proxy01.sipphone.com
; SIP Settings
canreinvite = no ; important!
[17476691234]
type=user
nat=never
context=from-pstn
canreinvite=no
; the following are just my settings I use, however
; I dont' consider them critical
allowexternaldomains = yes
allowexternalinvites = yes
allowguest = yes
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
autodomain = yes
callevents = no
compactheaders = no
dumphistory = no
g726nonstandard = no
ignoreregexpire = no
jbenable = no
jbforce = no
jblog = no
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 180
notifyringing = no
pedantic = no
promiscredir = no
recordhistory = no
relaxdtmf = no
rtcachefriends = no
rtsavesysname = no
rtupdate = no
sendrpid = yes
sipdebug = no
t1min = 100
progressinband = no
;register =
t38pt_udptl = no
trustrpid = no
usereqphone = no
videosupport = no
</screen>
<para>The Trunk definition looks like:</para>
<screen>
; users.conf:
[sipphone1]
username=17476691234
type=peer
qualify=2000
host=proxy01.sipphone.com
fromuser=17476691234
fromdomain=proxy01.sipphone.com
context=from-pstn
canreinvite=no
secret=&lt;password>
[general]
;
; Full name of a user
;
fullname = New User
userbase = 200
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
hasiax = no
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
host = dynamic
localextenlength = 3
allow_aliasextns = no
allow_an_extns = no
hasagent = no
hasdirectory = no
; local SIP extensions
[200]
username=200
type=friend
secret=XXXXXX
qualify=500
port=5060
pickupgroup=
nat=never
mailbox=
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid="Extension 200" <200>
allow=all
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; Local SIP UAs
; = locally connected phones. nothing special here.
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[201]
callwaiting = yes
cid_number = 201
context = local_sip
email = e@mail
fullname = Full Name
group =
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host = dynamic
mailbox = 201
secret = sip_password
threewaycalling = yes
zapchan =
registeriax = no
registersip = yes
vmsecret = 1234
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; SIP Trunks
; these are masqueraded via siproxd
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
[trunk_1]
disallow = all
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
callerid =
contact = 17476691234 ; IMPORTANT
context = DID_trunk_1
dialformat = ${EXTEN:1}
fromdomain = proxy01.sipphone.com
fromuser = 17476691234 ; IMPORTANT
group =
hasexten = no
hasiax = no
hassip = yes
host = proxy01.sipphone.com
insecure = very
port = 5060
provider =
registeriax = no
registersip = yes
secret = sip_password
trunkname = Custom - sipphone1234
trunkstyle = customvoip
username = 17476691234
</screen>
</sect1>
</chapter>