Sorry if this question already asked. I'm implementing Voip with asterisk and grandstream gxw4108, according from the manual, for connecting with PSTN I must configure one SIP account and assign that for dialing the PSTN so in my sip.conf I configure SIP account(extension) :
[1401] type=friend username=1401 secret=1401 host=dynamic context=my-office insecure=port in my extension.con [my-office] exten=>1401,1,Dial(SIP/1401,60) exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401) but the problem is when I dial the number for the PSTN it's run/dial on internal extension, from the asterisk guru website it's wrote to separate the incoming and out going in sip.conf [1401] type=friend username=1401 secret=1401 host=dynamic context=my-office-in insecure=port [1401] type=friend username=1401 secret=1401 host=dynamic context=my-office-out insecure=port in extension.conf [my-office-in] exten=>1401,1,Dial(SIP/1001,60) [my-office-out] exten=>_NXXNXXXX,1,Dial(SIP/${EXTEN}@1401) but still with this won't work too My question it's Is it my configuration true/correct or if there any other way for my problem I'm using 1 Stage Dialing and the asterisk server and Grandstream using different IP Address 192.168.101.xxx (for asterisk server) and 192.168.14.xxx (for grandstream gateway) thank you for helping -- Regards Samuel Sappa, -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users