Trixbox mit Sipgate.at keine eingehenden Anrufe!

sersandy

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hallo user,

vielen Dank an das Forum, hab es wenigstens geschafft dass ich intern Tel kann und nach außen auch, aber leider geht inbound nicht. meine trixbox v.2804 läuft als vmware, bridged net.

nachdem ich ein kompletter newbie bin weiß ich auch gar nicht mit welchen befehlen ich auf shell-ebene mir die stati anzeigen kann, die Webgui alleine ist zuwenig, bzw. wie kann ich mir welche logs ansehen???ports 5060-61 hab ich forgewarded, mehr kann ich auf meinen linksys gar nicht einstellen.

als sip-softphne clients X-Lite 4.0. habe 2 extensions angelegt die sowohl intern als auch nach extern tel können. mein provider sipcall.at, meinte ich muß bei Asterisk in der sip.conf einen useragent=irgendwas außer asterisk stehen. keine ahnung ob man diesen uagent noch wo braucht.

meine sip.conf:

Code:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general 
;options that you might need set. For example: enable and force the sip jitterbuffer. 
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall.  For nat'ing you'd need to add the following lines: 
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade 
;from previous versions.  If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them 
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a 
;extension to work to for example, those go here.  So you have extension 
;1000 defined in your system you start by creating a line [1000](+) in this 
;file.  Then on the next line add the extra parameter that is needed.  
;When the sip.conf is loaded it will append your additions to the end of 
;that extension. 
;
#include sip_custom_post.conf
sip_general_additional:

Code:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

vmexten=*97
disallow=all
allow=ulaw
allow=alaw
allow=h263
allow=h264
videosupport=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
sip_general_custom:

Code:
context=from-pstn
srvlookup=yes
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
sip_additional_conf:

Code:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

[1000]
deny=0.0.0.0/0.0.0.0
type=friend
secret=1000
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
[email protected]
host=dynamic
dtmfmode=rfc2833
dial=SIP/1000
context=from-internal
canreinvite=no
callgroup=
callerid=device <1000>
accountcode=
call-limit=50

[2000]
deny=0.0.0.0/0.0.0.0
type=friend
secret=2000
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
[email protected]
host=dynamic
dtmfmode=rfc2833
dial=SIP/2000
context=from-internal
canreinvite=no
callgroup=
callerid=device <2000>
accountcode=
call-limit=50

[111111111111]
secret=xxxxxxxxxx
type=user
context=from-trunk

[sipcall.at]
host=212.333.333.333
username=111111111111
secret=xxxxxxxxxx
type=peer
falls noch was fehlt ich bin für jede hilfe dankbar