Echocanceller setting

Hello,

I’am using Digium card TE420 PCI Express with echocanceller VPMOCT128. I had enable it in chan_dahdi.conf echocancell=yes. When I’am calling from SIP to DAHDI echocanceller is used to streem from Asterisk to user.
voipgw1*CLI> dahdi show channel 2
Channel: 2
Description:
File Descriptor: 10
Span: 1
Extension: 210021610
Dialing: no
Context: dms_in
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: SS7
Radio: 0
Owner: DAHDI/2-1
Real: DAHDI/2-1
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
HW Gains (RX/TX): Disabled/Disabled
SW Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
(unless TDM bridged) currently ON
Wait for dialtone: 0ms
CIC: 2
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

When I’m calling from DAHDI to SIP echocanceller is not used at all and called party is hearing echo
voipgw1*CLI> dahdi show channel 124
Channel: 124
Description:
File Descriptor: 132
Span: 4
Extension: 226079674
Dialing: no
Context: dms_in
Caller ID: 210021699
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: SS7
Radio: 0
Owner: DAHDI/124-1
Real: DAHDI/124-1
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
HW Gains (RX/TX): Disabled/Disabled
SW Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
(unless TDM bridged) currently OFF
Wait for dialtone: 0ms
CIC: 127
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

I don’t know why echocanceller is not used? Is it normal behavionur or am’I wrong with something?
I thought that echocanceller is used to streem from Asterisk to user independently from call direction.

Thank you for remarks and advice

Regards

Lubor