[asterisk-users] Issues with calling certain phone numbers...

[Luca Corti]

Hello,

I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbers
also via SIP through an AS5350 which has an E1 ISN PRI attached.

I have a PSTN operator number (say 012345678) routed to three SIP
extensions (01,21,20) and numbers to directly reach extensions from
outside (say 98765432XX, where 00 < XX < 99).

[outsidetoinside]

exten => 012345678,1,Dial(SIP/01,10,t);
exten => 012345678,n,Dial(SIP/21,10,t);
exten => 012345678,n,Dial(SIP/20,10,t);
exten => 012345678,n,Goto(1);

exten => _98765432XX,1,Dial(SIP/${EXTEN:8},60);
exten => _98765432XX,n,Hangup();


All two digits numbers dialed from extensions are routed to other
extensions, three digit numbers get routed to the PSTN Gateway.

[insidetooutside]

exten => 012345678,1,Dial(SIP/01);
exten => 012345678,n,Hangup();

exten => _98765432XX,1,Dial(SIP/${EXTEN:8});
exten => _98765432XX,n,Hangup();

exten => _XX.,1,Set(CALLERID(number)=012345678);
exten => _XX.,n,Dial(SIP/${EXTEN}@as5350);
exten => _XX.,n,Hangup();

exten => _XX,1,Dial(SIP/${EXTEN},30,t);
exten => _XX,n,Hangup();


The problem is that three digit numbers like 187 (which is a public
reachable PSTN number in my country, so I can reach it via the E1) is
not actually routed to the PSTN gateway (as it should).
I tried debugging SIP and see no request made to the AS5350. Is there a
command in the asterisk cli to debug how dialplan logic matches
requests? What could be crong?

TIA

Luca