DTMF outgoing

Asterisk Asterisk Support
1

Hey,

i have an Asterisk box with some SIP phone Grandstream BudgeTone 102 and two ISDN card (chipset Winbond 6692) configured with mISDN.

Voice calls, inbound and outbound, works fine but tones DTMF generated by SIP phone Grandstream don’t reache remote party.

In file sip.conf, for parameter dtmfmode, i tryed inband (with codec ulaw), rcf2833, info but without results.

Please help me.

2

Could you post more detail per the troubleshooting guidelines here:

forums.digium.com/viewtopic.php?t=4208

3

I’m not familiar with your phone, but had a similar problem with PSTN incoming line to a SPA 3000 FXO port and into asterisk. DTMF tone not reaching Asterisk. I had to change to info on both the Asterisk SIP settings for the SPA3000 AND on the SPA3000 settings. Does the Grandstream also have such a setting that you need to set along with the Asterisk config?

p

4

I’m using Asterisk@Home 2.2

mISDN installed with beronet.com/download/install-misdn.tar.gz.

sip.conf:

[general]
context=messagenet
srvlookup=yes
language=it
disallow=all
allow=alaw
relaxdtmf=yes
dtmf=auto
dtmfmode=auto
nat=yes

[301]
type=friend
secret=
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=Interni

extensions.conf

exten => _X.,10,Answer()
exten => _X.,13,Dial(SIP/301)

exten => _0.,1,Dial(misdn/2/${EXTEN},T)

misdn.conf

[general]
debug=0
trace_calls=false
trace_dir=/var/log/
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
l1_info_ok=yes
clear_l3=no
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=misdn
language=it
nationalprefix=0
internationalprefix=00
rxgain=1
txgain=1
te_choose_channel=no
method=standard
dialplan=0
localdialplan=0
use_callingpres=yes
early_bconnect=yes
echocancelwhenbridged=no
echotraining=yes
echocancel=128

[Esterne]
ports=1,2
context=Esterne
msns=828621936

5

BudgeTone permit to configure inband, rfc2833 and info dtmfmode. This works fine with Asterisk (i.e. accessing VoiceMail) configuring same mode in sip.conf.

6

Hi,

I am getting the exact same problem when i try to dial out to PSTN.

I am using EUROisdn E1 line through British Telecom.

When i call a number this is what i get:

When i make the above call I get the British telecom message, the number you have dialed has not been recognised.

the following are my confs:

Zapata.conf

sip.conf

Extensions.conf

outbound.include

output of “pri show span 1”