No more caller voice transmited since upgrade to 1.8

Hi folks,

I’ve been using an asterisk 1.6 server for about a year without trouble. Since last wednesday, when I upgraded to 1.8, I’m facing troubles. The voice of the caller is not transmited for internal calls (between two extensions managed by my server) but it works fine for other incoming or outgoing calls (which involve only one extension managed by my server).

I have a dozen extensions split on 3 sites. The problem occurs while calling from site A to site B but also from site A to site A.

The asterisk server is alone in the internet with a public static IP. All phones have a private IP and are natted. The routers have a public static IP.

The configuration files have not be changed during the upgrade. Relevant trimed files can be found here:


It sounds like NAT issue, uncomment the directmedia option, and change it to no b.[/b]

Asterisk has an additional “nat” parameter to
address NAT-related issues in incoming SIP or media sessions.
In particular, depending on the 'nat= ’ settings described below, Asterisk
may override the address/port information specified in the SIP/SDP messages,
and use the information (sender address) supplied by the network stack instead.
However, this is only useful if the external traffic can reach us.
The following settings are allowed (both globally and in individual sections):

nat = no ; Do no special NAT handling other than RFC3581
nat = force_rport ; Pretend there was an rport parameter even if there wasn’t
nat = comedia ; Send media to the port Asterisk received it from regardless
of where the SDP says to send it.
nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
nat = auto_comedia ; Set the comedia option if Asterisk detects NAT

Settings directmedia=no did fix my problem