Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten => 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running "sip debub" no messages are seen on screen. What additional routing informations are to be added to sip.conf, inorder to make it work . Thanx and regards sanchal _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users