Hello,
i build a Image v17.01 rc2 for my VGV7510KW22 with the Patch from Stefan Koch
(o26431-vmmc_v3.tar.gz)
https://drive.google.com/file/d/0B9Z_KppE2R8sMjhVSHRXWElLdE0/view?usp=sharing
Now i got a dial-tone, but can not call a telefonnumber.
What did i wrong ?
here are the Logfiles & Config Files: https://www.file-upload.net/download-12305299/asterisk.zip.html
Thanks
look inside the asterisk console:
asterisk -rand analyse what is goin wrong
It describe in the same thread read from here: https://forum.openwrt.org/viewtopic.php?pid=311920#p311920
here a parts from my modified config i hope it will work
I am direct connected to the Internet.
/etc/asterisk/extensions.conf
[globals]
FXS1_SNUM=21
FXS2_SNUM=22
[default]
; emty for security reasons
[foreign_in]
exten => _[+0-9A-Za-z].,1,NoOp(Foreign in Call from: ${CALLERID(all)})
exten => _3XX,2,Gosub(intern,${EXTEN},1)
exten => USERID_SG,2,Gosub(intern,300,1)
exten => USERID_FVD,2,Gosub(intern,300,1)
exten => _X!,2,NoOp(Wrong exten: ${EXTEN})
exten => _[+0-9A-Za-z].,n,Hangup
; this is your default context (see sip.conf: context=foreign_in ) all calls from foreign came here, be shure that you can NEVER NEVER NEVER dial out !
; only right extension will be handle here, usefull because crimminals try to telefon with your device, if each call goes to intern your telefon will ring non stop.
;
;
;
[foreign_out]
exten => _90.,1,Gosub(sipgatede,${EXTEN:2},1) ; if want to use this provider with an exact number (90 will delete)
exten => _91.,1,Gosub(freevoipdealcom,${EXTEN:2},1)
exten => _0X.,1,Gosub(freevoipdealcom,+49${EXTEN:1},1) ; country intern default (0 will replace by +49)
exten => _[+0-9A-Za-z].,n,Hangup
;
;
;
[all]
exten => _3XX,1,Gosub(intern,${EXTEN},1)
exten => _[09]X.,1,Gosub(foreign_out,${EXTEN},1)
exten => _[+0-9A-Za-z].,n,Hangup
;
;
;
[intern]
exten => 300,1,Dial(local/s@ltq1_in/n&local/s@ltq2_in/n&SIP/311) ; all, the both tapi device and the sip account ringing, the first pick up get the call
exten => 301,1,Goto(tel1_in,s,1)
exten => 302,1,Goto(tel2_in,s,1)
exten => _31X,1,Dial(SIP/${EXTEN})
exten => _[+0-9A-Za-z].,n,Hangup
exten => _[+0-9A-Za-z].,n,Return()
;
;
;
; ##############################################################
;
; draufsicht auf die Rückseite
; o2Box-6431
; ------------------------------
; | |
; | |
; | |
; | n f u |
; | | | | |
; | 1 2 |
; -------------------------------
[tel1_out]
exten => _X.,1,Set(CALLERID(num)=${FXS1_NUM})
exten => ${FXS2_SNUM},2,Goto(ltq2_in,s,1) ; direct call by dialing 22 from TAPI 1 to 2
exten => _X!,2,Gosub(all,${EXTEN},1)
;
[tel2_out]
exten => _X.,1,Set(CALLERID(num)=${FXS1_NUM})
exten => ${FXS1_SNUM},2,Goto(ltq1_in,s,1) ; direct call by dialing 21 from TAPI 2 to 1
exten => _X!,2,Gosub(all,${EXTEN},1)
;
;
;
[ltq1_out]
exten => _X!,1,Dial(local/${EXTEN}@tel1_out/n)
;
[ltq2_out]
exten => _X!,1,Dial(local/${EXTEN}@tel2_out/n)
;
;
;
;
[tel1_in]
exten => s,1,Dial(local/s@ltq1_in/n)
exten => s,2,Hangup
;
[tel2_in]
exten => s,1,Dial(local/s@ltq2_in/n)
exten => s,2,Hangup
;
;
;
;
[ltq1_in]
exten => s,1,Dial(TAPI/1,60,t)
exten => s,2,Hangup
;
[ltq2_in]
exten => s,1,Dial(TAPI/2,60,t)
exten => s,2,Hangup
;
;
;
;
[lantiq1]
include => ltq1_out
;
[lantiq2]
include => ltq2_out
;/etc/asterisk/sip.conf
[general]
context=foreign_in ; Default context for incoming calls. Defaults to 'default'
allowoverlap=yes ; Disable overlap dialing support. (Default is yes)
; udpbindaddr=[::]:5060
udpbindaddr=0.0.0.0:5060
tcpenable=no ; Enable server for incoming TCP connections (default is no)
transport=udp ; Set the default transports. The order determines the primary default transport.
srvlookup=yes
subscribe_network_change_event=yes
nat=force_rport,comedia
;
;
;
register => USERID_SG:PASS_SG@sipgate.de:5060/USERID_SG
register => USERID_FVD:PASS_FVD@sip.freevoipdeal.com:5060/USERID_FVD
;
;
;
[sipgatede]
context=foreign_in
type=peer
host=sipgate.de
port=5060
fromdomain=sipgate.de
insecure=port,invite
;
dtmfmode=rfc2833
disallow=all
allow=gsm,alaw,ulaw
;
defaultuser=USERID_SG
fromuser=USERID_SG
secret=PASS_SG
callerid=TELNUM_SG
;
;
;
[freevoipdealcom]
context=foreign_in
type=peer
host=sip.freevoipdeal.com
port=5060
fromdomain=sip.freevoipdeal.com
insecure=port,invite
;
dtmfmode=rfc2833
disallow=all
allow=gsm,alaw,ulaw
;
defaultuser=USERID_FVD
fromuser=USERID_FVD
secret=PASS_FVD
;
;
;
;
[311]
callerid=311
context=all
type=friend
host=dynamic
defaultuser=311
secret=secretpass-for-311
;
videosupport=yes
allow=ulaw,alaw
;
deny=0.0.0.0/0.0.0.0 ; only allow from LAN
permit=192.168.0.0/16
permit=172.16.0.0/12
permit=10.0.0.0/8
;/etc/asterisk/rtp.conf
[general]
rtpstart=15000
rtpend=16000
; define the portrange where the rtp will be used/etc/asterisk/lantiq.conf
[interfaces]
;
; Number of FXS ports (default: 2)
channels = 2
per_channel_context = onI can not garantee that this config will work, for any damage or cost that this description produce I do not take responsibility.
Please be really carefull, if it is possible to call from the default context (in this example [foreign_in]) to extern it can cost thousends of Money.
Because crimminals try to telefon over your maschine.
If you think it will not happens on you: It happens on me when i do my first experiments with Asterisk.
What a luck that the rest of my extensions.conf was a fail to.
1 - first experiments should only be do via prepaid account with full cost controll.
2 - read the-asterisk-book and understand the part with the extentions
3 - be shure that nobody can dial out from the default context !!!!
4 - permit SIP-accounts that will be used from LAN only, the rechability from LAN only
deny=0.0.0.0/0.0.0.0 ; only allow from LAN
permit=192.168.0.0/16
permit=172.16.0.0/12
permit=10.0.0.0/8
5 - use Long and complex passwords
If you are German you can read what happen If my advice is not followed.
in some constallations it is possible that the dial out fail: cause 20 subscriber absend
read this: https://forum.openwrt.org/viewtopic.php?id=69371
I solve this for me via:
/etc/hotplug.d/iface/90-asterisk-restart
#!/bin/sh
if test "$ACTION" == "ifup" || test "$ACTION" == "ifdown" ; then
case $DEVICE in
pppoe-wan) /etc/init.d/asterisk enabled && /etc/init.d/asterisk restart ;;
esac
fi
but this solution is very rustical