I need help to dial out via PSTN (FXO) port

Everytime i dial out via FXO 3 i get this output:

dahdi*CLI> == Using SIP RTP CoS mark 5 -- Executing [8095377584@phone-201:1] Dial("SIP/201-00000006", "dahdi/3/8095377584") in new stack [Nov 24 07:02:27] WARNING[2787]: app_dial.c:2030 dial_exec_full: Unable to create channel of type 'dahdi' (cause 17 - User busy) == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/201-00000006' status is 'BUSY' dahdi*CLI>

But if i dial via FXS port i get his output:

dahdi*CLI> == Using SIP RTP CoS mark 5 -- Executing [2@phone-201:1] Dial("SIP/201-00000007", "dahdi/4/222") in new stack -- Called 4/222 -- DAHDI/4-1 is ringing dahdi*CLI>

THAT’S MEAN IF I CALL VIA FXS THE CALL IS OK, BUT IF I CALL VIA FXO, I GET THE ERROR. How to fix the problem?

The output of dahdi show channels is:

dahdi*CLI> dahdi show channels Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service 3 from-pstn default In Service 4 from-internal default In Service dahdi*CLI>

I’m using asterisk-1.8.0 dahdi-linux-2.4.0, and dahdi-tools-2.4.0

Show us your system.conf and your chan_dahdi.conf and the output for lsdahdi

I fixed the problem :smiley:
The solution was that i have everything configured but the PSTN cable wasn’t connected to the Pbx.
But… now i sitll having problem to detect hangup, if you call me and the IVR answer yo the call and you hangup, the asterisk still executing others prioritys and 10 seconds after, the asterisk hangup the line. Do you know a solution to detect hangup inmediatly clients hangup the line when the IVR answer?