Upgrading from 1.6 to 1.8 disabled dtmf capture


#1

We have an asterisk with Digium TE405 card of which 1st span is connected to our local telco supporting SS7.
previously we had asterisk 1.6 and the only purpose/use of the system was to dial out mobile phone number thru the telco ss7 switch, play some ivr recording and capture the digits entered by the person receiving the call. This was working perfectly fine with 1.6.x. We upgraded to 1.8 (compiling the same modules as in 1.6 with additional cel modules) to use the channel event logging. We used the same config files (except the modules.conf) but now everything is working fine except the dtmf key pressed at the receiving mobile is not being captured. Is there some modules that I should add in 1.8 which is optional in 1.6 for dtmf?

content of current system.conf (using the same file from 1.6) is

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
mtp2=16

chan_dahdi.conf is

[channels]
language=en
relaxdtmf=yes

[nt1-gsm-ss7-0]
signalling=ss7
ss7type=itu
linkset=1
group=1
networkindicator=national
ss7_called_nai=national
ss7_calling_nai=subscriber
pointcode=910
adjpointcode=833
defaultdpc=833
cicbeginswith=1
dahdichan=1-15
sigchan=47
signchan=16
echocancel=no

[nt1-gsm-ss7-1]
signalling=ss7
linkset=1
group=1
cicbeginswith=17
dahdichan=17-31

usecallingpres=yes

extension.conf is

[general]
static = yes
writeprotect = yes
; THIS WILL PROBABLY NEED TO BE CHANGED
clearglobalvars = yes

[globals]
LANGUAGE = en
DEV_HOST=192.168.1.101

[test]
;exten => s,1,answer()
;exten => s,n,Wait(15)
;exten => s,n,SendDTMF(09)
;exten => s,1,Playback(digits/7)
exten => s,1,Agi(agi://${DEV_HOST}:4573/customivr?${parameters})
exten => s,n,Hangup()

manager.conf is

[general]
displaysystemname = yes
enabled = yes
webenabled = no
port = 5038
bindaddr=0.0.0.0
displayconnects = yes

[ivr]
secret = cello
;IPs disallowed to connect to AMI Port i.e. 5038. All IPs disallowed except IP defined in permit section below.
deny=0.0.0.0/0.0.0.0
permit= 10.0.0.105/10.0.0.105
permit = 192.168.1.101/192.168.1.101
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate

modules.conf

[modules]
autoload=yes
noload => pbx_gtkconsole.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_console.so

listing for lib/asterisk/modules directory:

app_celgenuserevent.so chan_dahdi.so func_frame_trace.so
app_chanisavail.so chan_local.so func_global.so
app_db.so chan_multicast_rtp.so func_logic.so
app_dial.so codec_adpcm.so func_sysinfo.so
app_dumpchan.so codec_alaw.so pbx_config.so
app_echo.so codec_a_mu.so pbx_spool.so
app_exec.so codec_dahdi.so res_agi.so
app_originate.so codec_gsm.so res_calendar.so
app_playback.so codec_ulaw.so res_clialiases.so
app_readexten.so format_g719.so res_clioriginate.so
app_read.so format_gsm.so res_convert.so
app_saycounted.so format_pcm.so res_fax.so
app_senddtmf.so format_sln.so res_monitor.so
app_setcallerid.so format_wav_gsm.so res_mutestream.so
app_system.so format_wav.so res_pktccops.so
cdr_csv.so func_callcompletion.so res_rtp_asterisk.so
cdr_custom.so func_callerid.so res_rtp_multicast.so
cdr_manager.so func_cdr.so res_security_log.so
cdr_syslog.so func_channel.so res_smdi.so
cel_custom.so func_db.so res_stun_monitor.so
cel_manager.so func_dialplan.so

Any pointers to resolve this issue much appreciated.
thanks in advance.

Danilel