Problem to SIP call to PSTN line

Asterisk Asterisk Support
1

Hi,
I’m new of this forum and i don’t have strong skill to asterisk :frowning:
This is my sistem

Ubuntu 9.01
Astyerisk 1.6.0.6
Dahdi 2.2.0
OpenVox A400P + 1FXO

Can yuo help me to configuration astersik/dahdi for incominc the SIP call to PSTN line.
I don’t have problem to call SIP to SIP in my Lan and PSTN call to SIP :smile:
Thanks

2

Maybe, if you provide the information needed to debug the problem.

3

Hi david55
thanks for your interest in my problem.
I have configuration 2 SIP phone.
This is the customization for the default file of daidhi end asterik

chan_dahdi.conf

context=from-pstn
signalling=fxs_ks
callerid=asreceived
group=0
channel=1

sip.conf

[a]
type=friend
host=dynamic
language=it
context=home
nat=yes
dtmfmode=rfc2833
dial=SIP/101
callerid=a <101>
username=a
secret=a

[c]
type=friend
host=dynamic
language=it
context=home
nat=yes
dtmfmode=rfc2833
dial=SIP/102
callerid=b <102>
username=b
secret=b

extensions.conf

[home]
exten = 101,1,Dial(SIP/a,20)
exten = 101,2,Hangup
exten = 102,1,Dial(SIP/c,20)
exten = 102,2,Hangup

[from-pstn]
exten = s,1,Dial(SIP/a,20)
exten = s,2,Hangup

a to c ok
pstn call to a ok
a to pstn line KO
Can you help me?

4

There are no Dial application calls referencing the dahdi channels!

5

You can write me a sample configuration to solve my problem?