Leo Salas wrote:
I am just learing some Linux and have been able to setup Asterisk samples and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1 analouge phone connected to port 4 of Digium TDM-400 with appropriate cards installed to dial out on. I wish to dial to the outside via PTSN line. I am lost on the instructions. Can anyone help with Extensions.conf and sap.conf. 3 extensions are needed.
Thanks for help.
Leo
I am using same setup running smoothly i am sending you my configs file hope you enjoy it currently i am dialing my sip phone to pstn and sip to dial anywhere in the world and also via pstn i am dialing my sip as well as locally fairly comfortably.
you change as per your requirement don't just copy try to understand the concept,logic behind that and you will be a happy man.
extensions.conf
[general] static=yes writeprotect=no
[outgoing] exten => _XXXXXX,1,Dial(Zap/4/${EXTEN}) exten => _111XXXXXX,1,Dial(Zap/4/${EXTEN}) exten => _1XXXXXXXXX,Dial(IAX Account) exten => _1XXXXXXXXX,Dial(SIP Account) exten => 103,1,Dial(Zap/1) exten => 123,1,VoiceMailMain exten => 101,1,Dial(SIP/101,20)
[from-sip] include => outgoing
sip.conf [general] port=5060 bindaddr=192.168.10.193 context=sip disallow=all allow=gsm nat=1
[101] username=101 type=friend host=dynamic secret=1234 context=from-sip callerid="101" dtmfmode=rfc2833 canreinvite=no ;disallow=all ;allow=gsm qualify=1000
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users