Hello, I just recently added a new VoIP account in my Asterisk 15.3 configuration. The problem is that the calling party or called party don’t hear the voice. My carrier said after analysis that RE-INVITE changes RTP ports but sends them on a previously negotiated ports. It is best not to send the last INVITE or prevent from changing its port.
I am a bit confused how to correct this?
Depending on the channel driver in use it would be the “directmedia” (in the case of chan_sip) or the “direct_media” (in the case of chan_pjsip) to stop reinviting. If this doesn’t work you’ll need to provide the configuration and a SIP trace (sip set debug on or pjsip set logger on).
1 Like