Outgoing and incoming calls on the card TDM400P impossible

myydoo
Joined: Fri 10 of Jul, 2009

Outgoing and incoming calls on the card TDM400P impossible

Posted:Fri 10 of Jul, 2009 (10:16 UTC)
Hello I have installed asterisk on debian etch , Here is the version
Zaptel-1.4.12.1
libpri-1.4.9
asterisk-addons-1.4.8
asterisk-1.4.24.1

Card TDM400P with 4 FXO port is installed on this server
2 PSTN lines are connected to the 1 and 2 ports on the card for incoming and outgoing PSTN call
Calls between internal SIP user is OK
When I make Outgoing call from my internal SIP phone(1022) to PSTN(33052369) through the Digium card, the external called
phone does not ring. see below the CLI during this call

SIPSERVER*CLI>
    -- Executing [33052369@default:1] Answer("SIP/1022-081e5450", "") in new stack
    -- Executing [33052369@default:2] Playback("SIP/1022-081e5450", "holdon") in new stack
    -- <SIP/1022-081e5450> Playing 'holdon' (language 'fr')
    -- Executing [33052369@default:3] ChanIsAvail("SIP/1022-081e5450", "Zap/g1/33052369|sj") in new stack
    -- Hungup 'Zap/1-1'
    -- Executing [33052369@default:4] Dial("SIP/1022-081e5450", "Zap/g1/33052369") in new stack
    -- Called g1/33052369
    -- Zap/1-1 answered SIP/1022-081e5450
    -- Hungup 'Zap/1-1'
  == Spawn extension (default, 33052369, 4) exited non-zero on 'SIP/1022-081e5450'

SIPSERVER*CLI>


When I make call from a PSTN outside number(33052369) to the PSTN line connected on digium card , the call is transferred
to the standard internal sip phone(1014), then I pik up the 1014 phone, but there is no voice from outside phone
(33052369) .On the outside phone (33052369) I have infinite ringing tonality. see below the CLI

-- Starting simple switch on 'Zap/1-1'
[Jul 10 11:59:42] NOTICE[3267]: chan_dahdi.c:6525 ss_thread: Got event 18 (Ring Begin)...
[Jul 10 11:59:44] NOTICE[3267]: chan_dahdi.c:6525 ss_thread: Got event 2 (Ring/Answered)...
    -- Executing [s@entrant:1] Answer("Zap/1-1", "") in new stack
    -- Executing [s@entrant:2] Dial("Zap/1-1", "SIP/1014") in new stack
    -- Called 1014
    -- SIP/1014-081e7008 is ringing
    -- SIP/1014-081e7008 answered Zap/1-1
  == Spawn extension (entrant, s, 2) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'
[Jul 10 12:00:31] NOTICE[2283]: chan_dahdi.c:6836 handle_init_event: Alarm cleared on channel 2
SIPSERVER*CLI>


Please how to resolv this problem
See below some test that I made and file

SIPSERVER:~#  /sbin/ztcfg -vv

Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels to configure.

SIPSERVER:~#
SIPSERVER:~#





extension.conf



[general]

[default]

;Appel entrant du PSTN

exten => s,1,Answer
;exten => s,2,ChanIsAvail(SIP/1014,sj)
exten => s,2,Dial(SIP/1014)
exten => s,3,Hungup()
exten => s,102,Answer()
exten => s,103,Dial(SIP/1015)
exten => s,104,Hungup()



;;Appel sortant vers PSTN

;exten => _3XXXXXXX,1,Answer
;exten => _3XXXXXXX,2,Playback(holdon)
;exten => _3XXXXXXX,3,Dial(Zap/g1/${EXTEN}) ; Expose all of 256-428 
;exten => _3XXXXXXX,4,hangup()


;exten => _3XXXXXXX,1,Answer
;exten => _3XXXXXXX,2,Playback(holdon)
;exten => _3XXXXXXX,3,ChanIsAvail(Zap/1&Zap/2,j)
;exten => _3XXXXXXX,4,NoOp(${AVAILORIGCHAN})
;exten => _3XXXXXXX,5,Dial(${AVAILORIGCHAN}/${EXTEN})
;exten => _3XXXXXXX,6,hangup()
;exten => _3XXXXXXX,102,Answer
;exten => _3XXXXXXX,103,Playback(call-waiting)
;exten => _3XXXXXXX,104,Playback(call-waiting)
;exten => _3XXXXXXX,105,hangup()



exten => _3XXXXXXX,1,Answer
exten => _3XXXXXXX,2,Playback(holdon)
exten => _3XXXXXXX,3,ChanIsAvail(Zap/g1/${EXTEN},sj)
exten => _3XXXXXXX,4,Dial(Zap/g1/${EXTEN})
exten => _3XXXXXXX,5,Hungup()
exten => _3XXXXXXX,102,Answer
exten => _3XXXXXXX,103,Playback(call-waiting)
exten => _3XXXXXXX,104,Playback(call-waiting)
exten => _3XXXXXXX,105,hangup()


;Appel interne

exten => _1XXX,1,Answer
exten => _1XXX,2,Playback(holdon)
;exten => _1XXX,3,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten => _1XXX,3,Dial(SIP/${EXTEN})
exten => _1XXX,4,Hangup




exten => 8400,1,Answer
exten => 8400,2,MusicOnHold(default)


exten => 8500,1,VoicemailMain
exten => 8500,2,Hangup


exten => 8888,1,Goto(s,1)






zapata.conf


[channels]

language=fr
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=sortant
nationalprefix=0
internationalprefix=00

rxgain=0.0
txgain=0.0 ; -> sortie

echocancel = yes
echocancelwhenbridged = yes
echotraining=200

threewaycalling=yes
transfer=yes

busydetect=yes
;callprogress=yes
hanguponpolarityswitch

signalling = fxs_ks
context=entrant
group=1
channel=>1-2
;channel=>1
;channel=>1-4



How to resolv this problem ?
Please help