[Asterisk-Users] Newbie - VoIP route SIP calls to provider
Dear Pros, Can anyone be kind enough to guide me to route calls to my SIP carrier. I have configured * to as local PBX from Softphones to hardphones and vice versa, the hardphone i have is AudioCodec MP108 8 FXS port gateway. SIP.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=g729 [1000] type=friend username=1000 host=dynamic context=internal ;canreinvite=yes dtmfmode=rfc2833 [2000] type=friend username=2000 secret=password2 host=dynamic context=internal ;canreinvite=yes dtmfmode=rfc2833 extension.conf [general] static=yes writeprotect=yes [globals] PHONE1=SIP/1000 PHONE2=SIP/2000 [international] ignorepat = 88 exten= _1N1NXXNXX,1,Dial ??? [internal] include = local-sip [local-sip] exten = 1000,1,Dial(${PHONE1},40,t) exten = 1000,2,Hangup exten = 2000,1,Dial(${PHONE2},40,t) exten = 2000,2,Hangup exten = 1001,1,Dial(${PHONE3},40,t) exten = 1001,2,Hangup i want user to dial 88 and they will get a tone and dial US or UK number from local-sip context. the provider only gave me a IP to route my SIP traffic and needs no registration, can any please help me how to write the International context in extension.conf also what i shld do in sip.conf.. my audiocodec supports g729 and g723 codec so do i need to aqquire license for G729 from digium, if yes then why? last if possible can you also please tell me wht i need to add on my context so user can while in calling put the call on hold and transfer to another sip phone. I thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie - VoIP route SIP calls to provider
Dear Pros, Can anyone be kind enough to guide me to route calls to my SIP carrier. I have configured * to as local PBX from Softphones to hardphones and vice versa, the hardphone i have is AudioCodec MP108 8 FXS port gateway. SIP.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=g729 [1000] type=friend username=1000 host=dynamic context=internal ;canreinvite=yes dtmfmode=rfc2833 [2000] type=friend username=2000 secret=password2 host=dynamic context=internal ;canreinvite=yes dtmfmode=rfc2833 extension.conf [general] static=yes writeprotect=yes [globals] PHONE1=SIP/1000 PHONE2=SIP/2000 [international] ignorepat = 88 exten= _1N1NXXNXX,1,Dial ??? [internal] include = local-sip [local-sip] exten = 1000,1,Dial(${PHONE1},40,t) exten = 1000,2,Hangup exten = 2000,1,Dial(${PHONE2},40,t) exten = 2000,2,Hangup exten = 1001,1,Dial(${PHONE3},40,t) exten = 1001,2,Hangup i want user to dial 88 and they will get a tone and dial US or UK number from local-sip context. the provider only gave me a IP to route my SIP traffic and needs no registration, can any please help me how to write the International context in extension.conf also what i shld do in sip.conf.. my audiocodec supports g729 and g723 codec so do i need to aqquire license for G729 from digium, if yes then why? last if possible can you also please tell me wht i need to add on my context so user can while in calling put the call on hold and transfer to another sip phone. I thankyou all for reading this mail and i hope someone will be kind enough to help. Best regards, Imran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users