Hello,
i am moving from static conf to realtime. I had the following .conf files
user.conf
[general]
fullname = New User
userbase = 6000
vmsecret =
hassip = yes
hasiax = yes
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
language = es
vmexten = 9998
[VOIP_Service]
context = DID_VOIP_service
host = sip.voip_service.com
username = 99fte32mCJ
secret = ecTeQzZ6Tki1YxpO
trunkname = VOIP_Service ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
hasvoicemail = no
fromuser = 99fte32mCJ
insecure = no
disallow = all
allow = ulaw,alaw,ilbc,gsm
.
.
.
sip.conf
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
allowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = yes
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain =
dtmfmode = rfc2833
dumphistory = no
externrefresh = 10
fromdomain =
g726nonstandard = no
jbenable = no
jbforce = no
jbimpl =
jblog = no
jbmaxsize =
jbresyncthreshold =
language = sp
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
mohsuggest =
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = no
rtpholdtimeout = 10
rtptimeout = 10
sendrpid = no
sipdebug = no
subscribecontext =
t1min = 100
t38pt_udptl = no
tos_audio = ef
tos_sip = CS3
tos_video = AF41
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no
mwi_from =
disallow = all
allow = ulaw,alaw,gsm,g729,ilbc
[siptouser]
username = siptouser
type = friend
context = sip
secret = password
host = dynamic
nat = yes
dtmfmode = auto
canreinvite = yes ;(possibly set to yes if you know what you are doing)
qualify = yes
incominglimit = 1
outgoinglimit = 1
call-limit = 1
busylevel = 1
I add a sip_buddies table to my mysql and move all my user information from user.conf to sip.conf for the above user but getting contex error loding module.
I am planing to have 500 new user in the table.
Do I have to enter all my users to the buddies’ table or some can stay in the sip.conf file?
kind reagrds,
Al