what do you get on the asterisk console output ? Date: Mon, 1 Sep 2014 18:53:51 +0530 From: [email protected] To: [email protected] Subject: [asterisk-users] SIP Calls Not Working
Hello, I have two sip phones (zoiper). Earlier these used to communicate using the settings below for sip.conf and extensions.conf and now we asterisk 1.8.29.0, so these phones have stopped communicating. My question is that does 1.8.29.0 release require any more changes to be done to the sip.conf and extensions.conf to make the below work ? The sip.conf contains following enteries ================================== [100] type=friend username=100 secret=100 host=dynamic port=5060 dtmfmode=rfc2833 fromdomain=dynamic nat=no canreinvite=false context=exten-100 [101] type=friend username=101 secret=101 host=dynamic port=5060 dtmfmode=rfc2833 fromdomain=dynamic nat=no canreinvite=false context=exten-101 The extensions.conf contains ======================== [exten-100] exten => 101,1,Dial(SIP/101) ;exten => echo,1,Echo() ;exten => busytone,1,Playback(moh) ;exten => 101,n,Hangup() exten => 100,1,Answer() exten => 100,n,Hangup() [exten-101] exten => 101,1,Answer() exten => 101,n,Hangup() exten => 100,1,Dial(SIP/100) ;exten => _x.,1,Playback(moh) -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
