Hi folks,<br><br>I'm having an issue with an asterisk 1.4.36 with an E1 card that is not forwarfing the early media a remote SIP end-point is creating.<br><br>--incoming E1 call-->asterisk 1.4.36---->SIP endpoint (which happens to be an asterisk 1.6.20).<br>
<br>I've checked signalling and the remote end-point returns 183 with the correct SDP. In the asterisk 1.4.36 I have progressinband=yes to precissely enable this feature and debugging from the asterisk console (both sip and rtp), the asterisk gateway gets the 183, create the remote peer and starts receiving the RTP (RTP From...blabnlabal).<br>
The only missing part is that the asterisk 1.4.36 instance gets the RTP audio from the SIP endpoint and forwards it to the E1 card.<br><br>I've also played with prematuremedia parameter but got no change in the behaviour.<br>
<br>Can anyone provide any hint about this issue? Both links to documentation and help debugging this issue will be highly appreciated.<br><br>Thank you very much in advance,<br>Samuel.<br>