i have asterisk (trixbox) up and running with 12 remote extensions defined. the system work for almost 12 hours without a problem. The outgoing calls can be made (we do not use incoming calls) with normal quality of voice (we use codec g711), but after 12 hours of continue work suddenly the calls lost part of the communication.
The people that receive the calls can hear the source voice, but the ones that made the calls can not here anything. In order to solve the problem we have to reboot the server once or twice.
we have read a lot of appends that say that problem of voice is most of the time because of nat, and we have review carefully our configurations without solving it. We really apreciate if someone give us some ideas. Thanks.
OUR SIP.CONF CONFIGURATION
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
registerattempts=0 ; we have just this today
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
externip=our gateway public ip
OUR RTP.CONF CONFIGURATION