one way audio from pstn to sip only (b400p):solved

Hello
I have a problem with one way audio from pstn
this is the situation.

When i make calls from pstn to sip i the audio on the sip phone is muted ,
the calling party can hear me but, on the sip phone i can't hear a thing like

Everything else works fine calling sip to sip or sip to pstn.
all is ok.

I use asterisk 1.4.22 misdn 1.1.8

here is the debug log of call from pstn --> sip phone 101
what boders me is

fw*CLI>
P[ 0] MGMT: SSTATUS: L1_ACTIVATED
P[ 4] channel with stid:0 not in use!
P[ 4] set_channel: bc->channel:0 channel:1
P[ 4] I IND :NEW_CHANNEL oad:113039465 dad:3445 pid:3 state:none
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:0 keypad: sending_complete:1
P[ 4] Chan not existing at the moment bc->l3id:80027 bc:0x8218e0 event:NEW_CHANNEL port:4 channel:1
P[ 4] NO USERUESRINFO
P[ 4] --> found chan (preselected): 1
P[ 4] --> TRANSPARENT Mode
P[ 4] I IND :SETUP oad:113039465 dad:3445 pid:3 state:none
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] --> Bearer: Audio 3.1k
P[ 4] --> Codec: Alaw
P[ 4] --> Bearer: Audio 3.1k
P[ 4] --> Codec: Alaw
P[ 0] --> * NEW CHANNEL dad:3445 oad:113039465
P[ 4] read_config: Getting Config
P[ 4] --> CTON: Unknown
P[ 4] --> EXPORT_PID: pid:3
P[ 4] --> PRES: Restricted (0)
P[ 4] --> SCREEN: Unscreened (0)
P[ 4] I SEND:PROCEEDING oad:90113039465 dad:3445 pid:3
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] * Starting Ast ctx:ulaz1 dad:3445 oad:90113039465 with 's' extension
-- Executing [s@ulaz1] Dial("mISDN/1-u1", "SIP/101") in new stack
-- Called 101
Extension Changed *00101[lokali] new state Ringing for Notify User 101
P[ 4] BCHAN: bchan ACT Confirm pid:3
P[ 4] MGMT: SSTATUS: L2_ESTABLISH
-- SIP/101-008867e0 is ringing
P[ 4] * IND : ringing pid:3
P[ 4] --> * IND : ringing pid:3
P[ 4] I SEND:ALERTING oad:90113039465 dad:3445 pid:3
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] --> * SEND: State Ring pid:3
P[ 4] --> incoming_early_audio off
-- SIP/101-008867e0 answered mISDN/1-u1
P[ 4] --> * IND : -1! (stop indication) pid:3
Extension Changed *00101[lokali] new state InUse for Notify User 101
P[ 4] --> None
P[ 4] * ANSWER:
P[ 4] --> Connection is without BF encryption
P[ 4] --> None
P[ 4] --> empty cad using dad
P[ 4] I SEND:CONNECT oad:90113039465 dad:3445 pid:3
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:3445
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] --> * Unknown Indication:20 pid:3
P[ 4] Sending Control ECHOCAN_ON taps:128
P[ 4] I IND :CONNECT_ACKNOWLEDGE oad:90113039465 dad:3445 pid:3 state:CONNECTED
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:3445
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] --> * Unknown Indication:20 pid:3
== Spawn extension (ulaz1, s, 1) exited non-zero on 'mISDN/1-u1'
P[ 4] * IND : HANGUP pid:3 ctx:ulaz1 dad:s oad:90113039465 State:CONNECTED
P[ 4] --> l3id:80027
Extension Changed *00101[lokali] new state Idle for Notify User 101
P[ 4] --> cause:16
P[ 4] --> out_cause:16
P[ 4] --> state:CONNECTED
P[ 4] I SEND:DISCONNECT oad:90113039465 dad:3445 pid:3
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:3445
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] --> Channel: mISDN/1-u1 hanguped new state:CLEANING
P[ 4] I IND :RELEASE oad:90113039465 dad:3445 pid:3 state:CLEANING
P[ 4] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:3445
P[ 4] --> info_dad: onumplan:2 dnumplan:0 rnumplan: cpnnumplan:0
P[ 4] --> caps:Audio 3.1k pi:3 keypad: sending_complete:1
P[ 4] ast_hangup already called, so we have no ast ptr anymore in event(RELEASE)
P[ 4] --> No need to queue hangup
P[ 4] Cannot hangup chan, no ast
P[ 4] $$$ CLEANUP CALLED pid:3
P[ 4] $$$ Cleaning up bc with stid :10010400 pid:3
P[ 4] Sending Control ECHOCAN_OFF
P[ 4] BCHAN: MGR_DELLAYER|CNF pid:3
P[ 4] MGMT: SSTATUS: L2_RELEASED
P[ 0] MGMT: SSTATUS: L1_DEACTIVATED
fw*CLI>

the card model is b400p configured in TE mode

one more thing i expirince the same problem described here

bugs.digium.com/view.php?id=10631

but i user 1.4.22 asterisk and that asterisk is patched with 87971-1 89791-2 patch .

could you try to install another version of mISDN or asterisk? It looks like a software configuration issue.

Is not a problem
tnx for the answer could you recommend me version of asterisk and misdn
to install.
I can show you my dial plan for sip phone but is simple

One more question if the my netmod device has 100omh termination enabled on S0 port should termination on the card be enabled or not?

you can try ealier 1 or 2 version, such as 1.4.20 and 1.1.7

S0 should be useless when working under NT mode.

Ok this is the solution for ubuntu hardy amd 64
and misdn 1.1.8 openvox b400p and default kernel.

instalated asterisk verision 1.6.0.1 and dahdi 2.0
i also needed to patch asterisk
this patch with bugs.digium.com/view.php?id=13491
Pecos asterisk is unable to place outgoing calls on TE port.
any way this is the solution for now.
for some wierd resaon i thing the bug is misdn_chan
as described heer bugs.digium.com/view.php?id=10631