[Asterisk-Users] OH323 not dial Modem[i4l]/g1

eltorio eltorio at lesmuids.com
Tue Aug 3 00:52:19 MST 2004


Hello everybody,

I have a strange comportment with oh323 and asterisk, I'start testing
asterisk but with this I can't understant plesae help me ! 
Thanks
Eltorio

----------------------------------------------------------
1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a
Modem[i4l] line
----------------------------------------------------------
Nothing happens when h323 to isdn even everything is ok for isdn to SIP and
isdn to h323 and sip to isdn!: trace

----------------------------------------------------------
2/ versions used
----------------------------------------------------------
Asterisk CVS-HEAD-07/29/04-19:00:52 built by root at compiere on a i686 running
Linux
Linux compiere 2.6.4-52-smp #1 SMP Wed Apr 7 02:11:20 UTC 2004 i686 i686
i386 GNU/Linux (Suse 9.1)
Oh323 0.6.3

-------------------------------------------------------
2-1 test FAILED NetMeeting(4001 on GnuGK) To ISDN Phone 221 (strip in phone
1 digit) so 5221 is dialed 221
----------------------------------------------------------
    -- Executing Dial("OH323/R21368", "Modem/g1:5221") in new stack
    -- Called g1:5221

(after more than 1 minute I stop Netmeeting call)

    -- H.323 call 'ip$192.168.3.1:30056/21368' cleared, reason 7 (Remote
user stopped calling)
    -- Hungup 'Modem[i4l]/ttyI1'
  == Spawn extension (lesmuids, 5221, 1) exited non-zero on 'OH323/R21368'
     -- Hungup 'OH323/R21368'

----------------------------------------------------------
2-2 test OK SIP Phone (on Asterisk) call 5221
----------------------------------------------------------
    -- Executing Dial("SIP/4122-d20d", "Modem/g1:5221") in new stack
Aug  3 02:54:22 WARNING[1141783472]: chan_modem_i4l.c:608 i4l_dial:
Outgoing MSN 4122 not allowed (see outgoingmsn=,, in modem.conf)
    -- Called g1:5221
    -- Modem[i4l]/ttyI1 answered SIP/4122-d20d
(blah blah.. and hang)
    -- Hungup 'Modem[i4l]/ttyI1'
  == Spawn extension (lesmuids, 5221, 1) exited non-zero on 'SIP/4122-d20d'
Strange think I can call a SIP extension

----------------------------------------------------------
2-3 test OK ISDN phone NetMeeting via Asterisk
----------------------------------------------------------
    -- Executing Goto("Modem[i4l]/ttyI0", "4001|1") in new stack
    -- Goto (lesmuids,4001,1)
     -- Executing Dial("Modem[i4l]/ttyI0", "OH323/4001") in new stack
    -- H.323 call to 4001 with codec ALAW
    -- Called 4001
    -- OH323/L20707 is ringing
    -- OH323/L20707 answered Modem[i4l]/ttyI0
    -- Hungup 'OH323/L20707'
  == Spawn extension (lesmuids, 4001, 1) exited non-zero on
'Modem[i4l]/ttyI0'
    -- H.323 call 'ip$localhost/20707' cleared, reason 1 (Cleared by local
user)
(blah blah)
    -- Hungup 'Modem[i4l]/ttyI0'

----------------------------------------------------------
3 Config
----------------------------------------------------------
Extensions.conf
[lesmuids]
;Accueil application (sur SIP/0, ou sur ISDN/400)

exten => 0,1,Wait,15				; Attend 15s (4 eme
sonnerie)
exten => 0,2,NoOp("Receive on 0")
exten => 0,3,Dial(SIP/4122&SIP/4123,60,tr)

exten => 1,1,Goto(4122,1)
exten => 2,1,Goto(4122,1)

exten => _[3-9],1,Goto(4001,1)

exten => 4001,1,Dial(OH323/4001)
exten => 4002,1,Dial(OH323/4002)
exten => 4008,1,Dial(OH323/408)
exten => 4122,1,Dial(SIP/4122,60,tr)
exten => 4123,1,Dial(SIP/4123,60,tr)

exten => _[5]ZXX,1,Dial(Modem/g1:${EXTEN})

oh323.conf

[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=1024
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
gatekeeper=DISCOVER
gatekeeperTTL=100
userInputMode=TONE
amaFlags=omit
context=lesmuids

[register]
context=lesmuids
gwprefix=41
gwprefix=5






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