We found that the 10 seconds relates to the "rtpkeepalive =10" in our
sip.conf. If the rtpkeepalive is reduced then the delay reduces as well. If
rtpkeepalive is removed from sip.conf then audio never starts flowing.
Does that help anyone make sense of what's happening?
We have DAHDI running on
Hello,
We are running an Asterisk 13 server which is having a strange problem,
where on calls which are received from the PSTN and then forwarded out to
the PSTN again there is no audio for the first 10 seconds of the call. At
the 10 second mark audio starts flowing fine, and in a PCAP we see
