Hi,
i have
- asterisk 13.23.1
- dahdi 2.11.1-r1 + dahdi-tools
- digium A8A card (8 lines)
- linksys spa921 phones (from our side)
SIP in/out calls work fine, inbound calls over dahdi trunk work too, but outbound calls over dahdi do not work: call starts, user phone rings, but i hear silence, no matter was call answered or not, and then message like ‘The number is not answering’. User also does not hear anything. No dial tones, no sound.
I’ve checked:
- RTP ports
- audio codecs
how can I identify the cause of the problem, any ideas? My config is listed below
sip.conf
[general]
context=default
allowguest=no
allowtransfer=yes
realm=evrogen.net
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
tos_sip=cs3
tos_audio=ef
tos_video=af41
maxexpiry=3600
minexpiry=60
defaultexpiry=120
alwaysauthreject=yes
externip=185.93.41.38
localnet=192.168.1.0/24
localnet=10.168.1.0/22
localnet=192.168.32.0/23
localnet=172.18.2.0/24
nat=yes
checkmwi=5
vmexten=voicemail
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
language=ru
relaxdtmf=no
trustrpid = no
sendrpid = yes
useragent=AstPbx.ru
dtmfmode = rfc2833
videosupport=yes
maxcallbitrate=1024
callevents=no
alwaysauthreject = yes
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=30
sipdebug = no
; For hints
limitonpeers=yes
limitonpeer=yes
call-limit=100
notifyringing=yes
;
recordhistory=no
dumphistory=no
t38pt_udptl = yes
registertimeout=60
registerattempts=0
directrtpsetup=no
canreinvite=no
jbenable = yes
jbforce = yes
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = adaptive
jblog = yes
defaultexpiry=3600
;register => sip_id:sip_pass@sipnet.ru/sip_id
[authentication]
[user](!)
type=friend
host=dynamic
context=numberplan-local
nat=no
qualify=yes
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
allowsubscribe=yes
subscribecontext = users
notifyringing = yes
notifyhold = yes
call-limit = 10
#include /etc/asterisk/sip_users.conf
#include /etc/asterisk/sip_trunks.conf
chan_dahdi.conf
[trunkgroups]
[channels]
language=ru
usecallerid=yes
hidecallerid=no
; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
;
;cidsignalling=v23
;
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see doc/India-CID.txt)
;
;cidstart=polarity
callerid=asreceived
callwaiting=yes
threewaycalling=yes
signalling = fxs_ks
transfer=yes
pulsedial=no
echocancel=yes
canpark=yes
echocancelwhenbridged=yes
faxdetect=incoming
usedistinctiveringdetection=no
callprogress=yes
usecallingpres=yes
callwaitingcallerid=yes
cancallforward=yes
callreturn=yes
hanguponpolarityswitch=no
group=0
context=306a-in
channel => 3
context=322a-in
channel => 1
;context=322-in
;channel => 7
group=1
context=325-in
;channel => 2,4-6
channel => 2,5,6
extensions.conf (part)
[general]
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
userscontext=default
#include extensions_custom.conf
#include extensions_services.conf
[globals]
INDIAL_TIMEOUT = 30
MENU_DIAL_TIMEOUT=20
INDIAL_OPTS = tkhxw
OUTDIAL_TIMEOUT = 3000
OUTDIAL_OPTS = TKH
EMERGENCY_PREFIX=911
EMERGENCY_LINE=DAHDI/4
TRUNK_1 = DAHDI/g1
;sipnet;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
TRUNK_2 = SIP/voipdiscount
;TRUNK_2 = SIP/sipnet
TRUNK_3 = SIP/sipnet
TRUNK_4 = IAX2/nomotech
EXT_PR=
;sipnet;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
MONITOR_EXEC = /etc/asterisk/scripts/2wav2ogg.sh
LIMIT_OUTGOING=7
CALLREC_PREFIX = /var/spool/asterisk/monitor
; Menu vars
WORKTIME=10:00-21:59 ; TODO: split worktime into start and end hours and automate weekend IVR
WORKDAYS=mon-fri
; timeouts for menus
TIMEOUT_RESPONSE=8
TIMEOUT_DIGIT=3
MENU_PAUSE=2
AFTER_ANSWER_PAUSE=1
QUEUE_TIMEOUT=60
[macro-trunkdial]
exten => s,1,Gosub(check-outgoing-limit,s,1)
exten => s,n,Gosub(call-record,${MACRO_EXTEN},1)
exten => s,n,Set(CALLERID(all)="Anonymous" <000>); We do not reveal our users to ISPs!
exten => s,n,GotoIf(${REGEX("^OOH323.*" ${ARG1})}?ooh323)
exten => s,n,Dial(${ARG1}/${ARG3}${ARG2},${OUTDIAL_TIMEOUT},${OUTDIAL_OPTS})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(ooh323),Set(ooh323peer=${CUT(ARG1,/,2)})
exten => s,n,Dial(OOH323/${ARG3}${ARG2}@${ooh323peer},${OUTDIAL_TIMEOUT},${OUTDIAL_OPTS})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Answer
exten => s-NOANSWER,n,Wait(${AFTER_ANSWER_PAUSE})
exten => s-NOANSWER,n,Playback(abonent)
exten => s-NOANSWER,n,Playback(ne-otvechaet)
exten => s-NOANSWER,n,Hangup
exten => s-BUSY,1,Answer
exten => s-BUSY,2,Playtones(busy)
exten => s-BUSY,3,Wait(5)
exten => s-BUSY,4,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1)