Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following settings: sip.conf: [sipphone-1] type=friend host=dynamic dtmfmode=inband username=sipphone-1 secret=blablabla extensions.conf exten => 100,1,dial(SIP/sipphone-1) then I issued a reload on the asterisk command console I am using X-lite as SIP softphone. I configured the SIP proxy as given on the instructions on the site voip-info.org/wiki-Asterisk+phone+xten+xlite I dialed the 1000 extension, and got connected, but there is no sound. I know that i should hear the demo comunication, but there is no sound. What am i doing wrong? Any help is welcome Regards Bozhidar
I'm by no means an asterisk Guru, just trying to get is together my self. How ever, no sound issues usually relate to blocked ports on your router / firewall. If your extension 1000 is an IAX connection, check your rtp.conf, and perhaps narrow the port range, allow port forwarding on this range (UDP) and port 5060 to your asterisk server. This seemed to do the trick for me. Hope this is of some use. Regards Chris -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of bozidar@mt.net.mk Sent: 18 January 2005 14:22 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with demo on asterisk Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following settings: sip.conf: [sipphone-1] type=friend host=dynamic dtmfmode=inband username=sipphone-1 secret=blablabla extensions.conf exten => 100,1,dial(SIP/sipphone-1) then I issued a reload on the asterisk command console I am using X-lite as SIP softphone. I configured the SIP proxy as given on the instructions on the site voip-info.org/wiki-Asterisk+phone+xten+xlite I dialed the 1000 extension, and got connected, but there is no sound. I know that i should hear the demo comunication, but there is no sound. What am i doing wrong? Any help is welcome Regards Bozhidar _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users