*** empty log message ***
This commit is contained in:
Thomas Ries
parent
ab0ed062bc
commit
f1d70b2fe5
@ -27,6 +27,7 @@ EXTRA_DIST = siproxd.conf.example \
|
||||
RFC3261_compliance.txt \
|
||||
sample_cfg_budgetone.txt \
|
||||
sample_cfg_x-lite.txt \
|
||||
sample_asterisk.txt \
|
||||
siproxd_guide.sgml
|
||||
#
|
||||
# docbook stuff
|
||||
|
||||
236
doc/asterisk_sample.txt
Normal file
236
doc/asterisk_sample.txt
Normal file
@ -0,0 +1,236 @@
|
||||
Example Setup, Asterisk running on the same machine as siproxd (= NAT host).
|
||||
|
||||
I use a iptables rule to redirect all outgoing SIP traffix from Asterisk
|
||||
to siproxd.
|
||||
|
||||
/etc/sysconfig/iptables:
|
||||
------------------------
|
||||
*nat
|
||||
######################################################################
|
||||
# NAT: redirect locally generated packets
|
||||
:OUTPUT - [0:0]
|
||||
#########################
|
||||
#
|
||||
# Asterisk Traffix via local siproxd. (must use DNAT to inbound IF! not REDIRECT)
|
||||
-A OUTPUT -o ppp+ -p udp --sport 5061 -j DNAT --to-destination 192.168.1.1:5060
|
||||
|
||||
COMMIT
|
||||
|
||||
|
||||
|
||||
|
||||
/etc/asterisk/sip.conf:
|
||||
-----------------------
|
||||
[general]
|
||||
context = default
|
||||
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
|
||||
bindport = 5061 ; use a different port than 5060, as that port will be
|
||||
; occupied by siproxd!
|
||||
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
|
||||
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
|
||||
|
||||
; g726 sounds very bad, useless!
|
||||
; g722 is a dead end, no conversion from/to possible
|
||||
; g729 sound like through a long metal tube
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
autoframing = yes
|
||||
|
||||
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
|
||||
t38pt_udptl = no
|
||||
trustrpid = no
|
||||
usereqphone = no
|
||||
videosupport = no
|
||||
;
|
||||
; the following is required when using siproxd with local DNAT rule
|
||||
nat=never
|
||||
externip=192.168.1.1
|
||||
;
|
||||
localnet = 192.168.0.0/16 ; my inbound network with local UAs
|
||||
domain = 192.168.1.1 ; inbound IP of host running Asterisk and siproxd
|
||||
domain = mynatfirewall ; -"-
|
||||
canreinvite = no
|
||||
|
||||
useragent = PBX ; sipcall.ch (and others?) require UA string
|
||||
; to be different from "AsteriskPBX"
|
||||
|
||||
[authentication]
|
||||
;---end---
|
||||
|
||||
|
||||
/etc/asterisk/users.conf
|
||||
------------------------
|
||||
[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 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 UAs
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
[201]
|
||||
callwaiting = yes
|
||||
cid_number = 201
|
||||
context = local_sip
|
||||
email = email@host.xx
|
||||
fullname = Full Name
|
||||
group =
|
||||
hasagent = yes
|
||||
hasdirectory = yes
|
||||
hasiax = no
|
||||
hasmanager = no
|
||||
hassip = yes
|
||||
hasvoicemail = yes
|
||||
host = dynamic
|
||||
mailbox = 201
|
||||
secret = <secret for AUTH>
|
||||
threewaycalling = yes
|
||||
zapchan =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
vmsecret = <secret for voicemail>
|
||||
|
||||
[202]
|
||||
callwaiting = yes
|
||||
cid_number = 202
|
||||
context = local_sip
|
||||
email = email@host.xx
|
||||
fullname = Full Name
|
||||
group =
|
||||
hasagent = yes
|
||||
hasdirectory = yes
|
||||
hasiax = no
|
||||
hasmanager = no
|
||||
hassip = yes
|
||||
hasvoicemail = yes
|
||||
host = dynamic
|
||||
mailbox = 202
|
||||
secret = <secret for AUTH>
|
||||
threewaycalling = yes
|
||||
zapchan =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
vmsecret = <secret for voicemail>
|
||||
|
||||
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; SIP Trunks
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; sipphone.com
|
||||
[trunk_1]
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
callerid =
|
||||
contact = 1747669xxxx
|
||||
context = DID_trunk_1
|
||||
dialformat = ${EXTEN:1}
|
||||
fromdomain = proxy01.sipphone.com
|
||||
fromuser = 1747669xxxx
|
||||
group =
|
||||
hasexten = no
|
||||
hasiax = no
|
||||
hassip = yes
|
||||
host = proxy01.sipphone.com
|
||||
insecure = very
|
||||
port = 5060
|
||||
provider =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
secret = <secret for AUTH>
|
||||
trunkname = Custom - sipphone1341
|
||||
trunkstyle = customvoip
|
||||
username = 1747669xxxx
|
||||
|
||||
; sipcall.ch
|
||||
[trunk_3]
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
callerid =
|
||||
contact = 4132511xxxx
|
||||
context = DID_trunk_3
|
||||
dialformat = ${EXTEN:1}
|
||||
fromdomain = sip.backbone.ch
|
||||
fromuser = 4132511xxxx
|
||||
group =
|
||||
hasexten = no
|
||||
hasiax = no
|
||||
hassip = yes
|
||||
host = sip.backbone.ch
|
||||
insecure = very
|
||||
port = 5060
|
||||
provider =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
secret = <secret for AUTH>
|
||||
trunkname = Custom - sipcall
|
||||
trunkstyle = customvoip
|
||||
username = 4132511xxxx
|
||||
|
||||
;---end---
|
||||
236
doc/sample_asterisk.txt
Normal file
236
doc/sample_asterisk.txt
Normal file
@ -0,0 +1,236 @@
|
||||
Example Setup, Asterisk running on the same machine as siproxd (= NAT host).
|
||||
|
||||
I use a iptables rule to redirect all outgoing SIP traffix from Asterisk
|
||||
to siproxd.
|
||||
|
||||
/etc/sysconfig/iptables:
|
||||
------------------------
|
||||
*nat
|
||||
######################################################################
|
||||
# NAT: redirect locally generated packets
|
||||
:OUTPUT - [0:0]
|
||||
#########################
|
||||
#
|
||||
# Asterisk Traffix via local siproxd. (must use DNAT to inbound IF! not REDIRECT)
|
||||
-A OUTPUT -o ppp+ -p udp --sport 5061 -j DNAT --to-destination 192.168.1.1:5060
|
||||
|
||||
COMMIT
|
||||
|
||||
|
||||
|
||||
|
||||
/etc/asterisk/sip.conf:
|
||||
-----------------------
|
||||
[general]
|
||||
context = default
|
||||
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
|
||||
bindport = 5061 ; use a different port than 5060, as that port will be
|
||||
; occupied by siproxd!
|
||||
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
|
||||
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
|
||||
|
||||
; g726 sounds very bad, useless!
|
||||
; g722 is a dead end, no conversion from/to possible
|
||||
; g729 sound like through a long metal tube
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
autoframing = yes
|
||||
|
||||
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
|
||||
t38pt_udptl = no
|
||||
trustrpid = no
|
||||
usereqphone = no
|
||||
videosupport = no
|
||||
;
|
||||
; the following is required when using siproxd with local DNAT rule
|
||||
nat=never
|
||||
externip=192.168.1.1
|
||||
;
|
||||
localnet = 192.168.0.0/16 ; my inbound network with local UAs
|
||||
domain = 192.168.1.1 ; inbound IP of host running Asterisk and siproxd
|
||||
domain = mynatfirewall ; -"-
|
||||
canreinvite = no
|
||||
|
||||
useragent = PBX ; sipcall.ch (and others?) require UA string
|
||||
; to be different from "AsteriskPBX"
|
||||
|
||||
[authentication]
|
||||
;---end---
|
||||
|
||||
|
||||
/etc/asterisk/users.conf
|
||||
------------------------
|
||||
[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 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 UAs
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
[201]
|
||||
callwaiting = yes
|
||||
cid_number = 201
|
||||
context = local_sip
|
||||
email = email@host.xx
|
||||
fullname = Full Name
|
||||
group =
|
||||
hasagent = yes
|
||||
hasdirectory = yes
|
||||
hasiax = no
|
||||
hasmanager = no
|
||||
hassip = yes
|
||||
hasvoicemail = yes
|
||||
host = dynamic
|
||||
mailbox = 201
|
||||
secret = <secret for AUTH>
|
||||
threewaycalling = yes
|
||||
zapchan =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
vmsecret = <secret for voicemail>
|
||||
|
||||
[202]
|
||||
callwaiting = yes
|
||||
cid_number = 202
|
||||
context = local_sip
|
||||
email = email@host.xx
|
||||
fullname = Full Name
|
||||
group =
|
||||
hasagent = yes
|
||||
hasdirectory = yes
|
||||
hasiax = no
|
||||
hasmanager = no
|
||||
hassip = yes
|
||||
hasvoicemail = yes
|
||||
host = dynamic
|
||||
mailbox = 202
|
||||
secret = <secret for AUTH>
|
||||
threewaycalling = yes
|
||||
zapchan =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
vmsecret = <secret for voicemail>
|
||||
|
||||
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; SIP Trunks
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; sipphone.com
|
||||
[trunk_1]
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
callerid =
|
||||
contact = 1747669xxxx
|
||||
context = DID_trunk_1
|
||||
dialformat = ${EXTEN:1}
|
||||
fromdomain = proxy01.sipphone.com
|
||||
fromuser = 1747669xxxx
|
||||
group =
|
||||
hasexten = no
|
||||
hasiax = no
|
||||
hassip = yes
|
||||
host = proxy01.sipphone.com
|
||||
insecure = very
|
||||
port = 5060
|
||||
provider =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
secret = <secret for AUTH>
|
||||
trunkname = Custom - sipphone1341
|
||||
trunkstyle = customvoip
|
||||
username = 1747669xxxx
|
||||
|
||||
; sipcall.ch
|
||||
[trunk_3]
|
||||
disallow = all
|
||||
allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
|
||||
callerid =
|
||||
contact = 4132511xxxx
|
||||
context = DID_trunk_3
|
||||
dialformat = ${EXTEN:1}
|
||||
fromdomain = sip.backbone.ch
|
||||
fromuser = 4132511xxxx
|
||||
group =
|
||||
hasexten = no
|
||||
hasiax = no
|
||||
hassip = yes
|
||||
host = sip.backbone.ch
|
||||
insecure = very
|
||||
port = 5060
|
||||
provider =
|
||||
registeriax = no
|
||||
registersip = yes
|
||||
secret = <secret for AUTH>
|
||||
trunkname = Custom - sipcall
|
||||
trunkstyle = customvoip
|
||||
username = 4132511xxxx
|
||||
|
||||
;---end---
|
||||
Loading…
x
Reference in New Issue
Block a user