[Asterisk-Users] Problem with demo on asterisk

[Chris Blunt]

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 at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
bozidar at mt.net.mk
Sent: 18 January 2005 14:22
To: asterisk-users at 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
http://www.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
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