Weird behavior: the caller can't hear anything

Asterisk Asterisk Support
1

Hi everyone,

Everything was pretty good until we decided to test the conferencing feature. Now, the regular call (between 2) is not working anymore. Just the caller can’t hear anything but the other can, besides amp says ANSWERED.

Does anybody know why and how to fix it?

Thanks in advance
William

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

;port = 5060 ; Port to bind to (SIP is 5060)
;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
;disallow=all
;allow=ulaw
;allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
;callerid = Unknown

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.255.17 ; Address to bind to (all addresses on machine)
fromdomain=x.com
srvlookup=yes
videosupport=yes
language=en
nat=yes
externip=x.x.x.x
localnet=192.168.255.0/24
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

2

Sorry i forgot something

we are using Polycom sip phones, we are sure that the phone is ok (physical).
And our box has enough resources(mem and cpu).

Running on cent OS, asterisk at home 2.1.

Thanks

3

Im in no way an expert but isnt it nat=1 not nat=yes

4

You will need nat=yes on the phones that are outside of your network.

In sip.conf in the [general] section, you need

externip=xx.xx.xx.xx ; Replace xx.xx.xx.xx with your public IP
localnet=192.168.1.0/255.255.255.0 ; or whatever your local network is

You’ll probably want delete nat=yes from the [generals] unless ALL of your clients are on the outside.

-Trey

P.S. nat=1 and nat=yes all do the same thing.

5

try to put canreinvite=no in the sip.conf for each of the phones. it disallows a direct media path between the phones and makes it route through the asterisk server

6

Thanks kctrey,

your info solved my problem

Have i nice day