hello
i install the astersik and sound by tar file of version , asterisk-1.2.2.tar.gz and asterisk-sounds-1.2.1.tar.gz .
i install it and run it by configuring to files sip.conf and extension.conf.
In sip.conf file,i write following code.
[general]
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
allow =all ; Note: Asterisk only uses the first host
context =bogon-calls ; in SRV records
domain=172.16.100.54 ; my asterisk server computer ip address
this much of changes i did in configuration file .then i dial one soft phone to another ,it shows that phone is ringing and when another pc soft phone pick up then it shows connected.but when i speak it gives noise sound and it is not clear.
so what to do for this problem , should i add anything in these configuration files?
you should upgrade to the latest version of asterisk. earlier versions had a bug that stopped bridging, i.e. no sound. this may or may not be the cause, but it’s my tuppence
Hi,
I think its not the problem of your version. And I dont know what’s the need of asterisk-sounds-1.2.1.tar.gz. If you knows please post it.
Again a version difference is there in ‘asterisk-1.2.2.tar.gz and asterisk-sounds-1.2.1.tar.gz’.
I suggest check your hardware settings.
My extensions.conf is:
[quote=“ijb”]Hi,
I think its not the problem of your version. And I dont know what’s the need of asterisk-sounds-1.2.1.tar.gz. If you knows please post it.
Again a version difference is there in ‘asterisk-1.2.2.tar.gz and asterisk-sounds-1.2.1.tar.gz’.
I suggest check your hardware settings.
My extensions.conf is:
hi ijb
thanks ,ya it is problem on my extension.conf file .
i change whatever u sent me and it works. i install asterisk-sounds-1.2.1.tar.gz, just for sound.i don’t know about much of it. but i think it is sample sounds.
once more thanks a lots for helping me.
Also is there any third party software like it records the user’s calling and stop time and
we can integate with asterisk server.