>From your descriptions of your needs, you would be better served with anAAH installation. Easier to understand than hand coding your contexts. That aside, here are few answers... Look here for more... www.voip-info.org Routing to the VoIP is just a matter of dial plan matching (see dial plan at voip-info) Codecs on Asterisk require the license. Your phone may support the codec but your server needs a license to do so. Hold and transfer are usually part of the phone itself. For example, my Polycom holds the line and lets me transfer from itself. Otherwise, I can transfer a call via * by dialing # and the extension target. This is a blind transfer so the target extension just gets the call without you even talking to them. Consultative transfer is different, details located at www.voip-info.org Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of iMRAN Sent: Wednesday, April 20, 2005 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 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=> _1N1NXXNXXXXXX,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