No sound on pickup, but works after hold on/off


I am newbie in Asterisk, and facing strange behaviour on two implementations I make in parallel :

I have the minimum configuration for sip.conf and extensions.conf, that permit to register two local sip softphones, play echotest and Dial the two softphone.

The two softphones work fine with echotest.

When I dial a softphone from the other one, the second rings well, and I can pickup the call and establish the connection.

The problem is that at this state I have no sound between the phones.

If I put the call on hold from one of the phones, the second one has the music on hold, and when I quit the hold, then the sound works fine between the two phones.

Surely I made something wrong in my installation process, for my two implementations have the same behaviour, but, after a lot of forum digging, I do not see where to look.

Any idea would be appreciated.


I would look around the setting of directmedia. Note that some soft phones will violate the protocol, causing the call to drop, if you try to use direct media.

Hi David,

You pointed it exactly.

I just changed the canreinvite setting in my sip.conf file from no to yes, and now everything is ok.

Thank you so much for saving me time.

Now I will take time to RTFM, and understand the meaning of the parameters.

Thanks again,


Actually I was expecting the opposite change to help! If directmedia = no doesn’t work, there is a fairly serious problem somewhere. The canreinvite name for this parameter is depreacted.