Audio problems in SIP calls

Asterisk Asterisk Support
1

Hi,
I am working on a java desktop application to allow audio calls between PCs.
I am using asterisk 11.17.1 and Ubuntu 14.04.

I start an audio call, but I have problems with the audio. Only one way is working.
I start a call from the PC 1 to a softphone 2, but from the telephone I can not hear nothing but a noise sound.

Both devices are in the same office, and using the same local net.

SIP.conf…
[general]
callcounter=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
svrlookup=no
disallow=all
allow=ulaw; ,alaw,gsm,g722 // I try all of them

Thanks in advance and I am sorry about my poor english

2

since this is asterisk 11, this could be a nat issue.
Try nat=yes

Also check that you have opened the rtp ports on the firewall. This should be your UDP ports considering its default settings