Luca Corti
2006-Oct-02 06:50 UTC
[asterisk-users] Issues with calling certain phone numbers...
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
Marco Mouta
2006-Oct-02 08:22 UTC
[asterisk-users] Issues with calling certain phone numbers...
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/....) On 10/2/06, Luca Corti <luca@leenoox.net> wrote:> > 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 > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Com os melhores cumprimentos, Marco Mouta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061002/56212edc/attachment.htm