Stressing asterisk with SIPP generating calls with audio


I’m trying to test an asterisk system using sipp with rtp media.

I teste sipp making it call-me in my extension with this command:

It instruct sipp do call, play the audio and press a button (sending a dtmf tone)

It’s working…kinda.

After picking up I can hear the a man talking (generated by sipp) but the sound is interrupted (the voice, not the call) and all I can hear is silence until the DTMF sound.

Anyone knows what may be happening?