Hallo,
habe mich streng an betateilchens Tutorial gehalten, bekomme aber die Rufzuweisung meiner 3 Sipgateaccounts nicht hin.
D.h. wenn ich auf meinen Nummern anrufe, werde diese zwar angenommen,
ich kann diese jedoch nicht den einzelnen Teilnehmern zuordnen... !?
Meine Sip.conf:
und meine extensions.conf
würde mich wie immer über jeden Tip freuen
Gruß DD
habe mich streng an betateilchens Tutorial gehalten, bekomme aber die Rufzuweisung meiner 3 Sipgateaccounts nicht hin.
D.h. wenn ich auf meinen Nummern anrufe, werde diese zwar angenommen,
ich kann diese jedoch nicht den einzelnen Teilnehmern zuordnen... !?
Meine Sip.conf:
Code:
; sip.conf - optimized
; find original file in /usr/share/doc/asterisk1.4-config/config.samples/sip.conf
[general]
; --- general options ---
bindaddr = 0.0.0.0
bindport = 5060
language = de
context = inbound
subscribecontext = internal
;regcontext = internal
;realm = mydomain.tld
;useragent = Asterisk PBX 1.4
usereqphone = no
;relaxdtmf = yes
;compactheaders = yes
notifyringing = yes
notifyhold = yes
;alwaysauthreject = yes
;callerid = whatever
;outboundproxy = proxy.provider.domain
;outboundproxyport = 5060
;pedantic = no
;sipdebug = no
;dumphistory = yes
;recordhistory = yes
disallow = all
allow = alaw
allow = ulaw
allow = gsm
;autoframing = no
tos_sip = cs3
tos_audio = ef
tos_video = af41
qualify = 5000
;t1min = 100
callevents = yes
;maxcallbitrate = 384
;g726nonstandard = yes
useclientcode = yes
dtmfmode = auto
nat = route
allowtransfer = yes
limitonpeers = yes
trustrpid = yes
sendrpid = yes
canreinvite = no
insecure = no
progressinband = never
promiscredir = yes
videosupport = yes
allowoverlap = no
allowsubscribe = yes
t38pt_udptl = yes
rfc2833compensate = yes
;buggymwi = yes
; --- Voicemail settings ---
checkmwi = 10
;vmexten = mailbox
;notifymimetype = text/plain
; --- SIP domain support ---
allowguest = no
allowexternaldomains = yes
;autodomain = yes
;domain = mydomain.tld,mydomain-incoming
;fromdomain = mydomain.tld
autocreatepeer = no
srvlookup = yes
;localnet = 10.0.0.0/255.0.0.0
;localnet = 172.16.0.0/255.240.0.0
;localnet = 169.254.0.0/255.255.0.0
;localnet = 192.168.0.0/255.255.0.0
; either specify your "official" external ip address here or
; use the astertools service for getting this ip address everytime
; you reload or restart asterisk
;externip = 200.201.202.203
#exec wget -q -O - http://services.astertools.com/sip_externip.php
matchexterniplocally = no
; still experimental, do not use!
;externhost = foo.dyndns.net
;externrefresh = 10
; --- RealTime settings ---
rtcachefriends = yes
rtsavesysname = yes
rtupdate = yes
ignoreregexpire = yes
rtautoclear = no
; --- RTP settings ---
;rtptimeout = 60
;rtpholdtimeout = 300
;rtpkeepalive = 90
;directrtpsetup = no
; --- MusicOnHold settings ---
mohinterpret = default
;musicclass = default
;musiconhold = default
;mohsuggest = default
; --- Registration settings ---
; outbound
registertimeout = 20
registerattempts = 10
; inbound
maxexpiry = 3600
minexpiry = 60
defaultexpiry = 900
; --- Jitter Buffer settings ---
jbenable = yes
jbforce = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed
jblog = no
; --- outbound registrations ---
;register => user[:secret[:authuser]]@host[:port][/extension]
;register => 1234:[email protected]
register => 13900000:[email protected]/13900000
[13900000]
type=peer
username=13900000
fromuser=13900000
secret=123XX
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=no
disallow=all
allow=ulaw
;SIPGATE Line2
register => 13900002:[email protected]/13900002
[13900002]
type=peer
username=13900002
fromuser=13900002
secret=123XX
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=no
disallow=all
allow=ulaw
;SIPGATE Line3
register => 13900003:[email protected]/13900003
[13900003]
type=peer
username=13900003
fromuser=13900003
secret=123XX
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=no
disallow=all
allow=ulaw
[sipgate_de_in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
disallow=all
allow=ulaw
context=inbound
[authentication]
;auth = 1234:[email protected]
#include "sip.d/*.conf"und meine extensions.conf
Code:
[inbound]
;Übergebe Gespräch von extern
exten = 13900000,1,Goto(inbound,10,1)
exten = 13900001,1,Goto(internal,99,1)
exten = 13900002,1,Goto(inbound,11,1)würde mich wie immer über jeden Tip freuen
Gruß DD
Zuletzt bearbeitet:
