<div dir="ltr">Hi all....<br>I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found..<br>
[Feb 28 11:34:14] NOTICE[19197] chan_sip.c: Disconnecting call 'SIP/netcafe2-b7da99b8' for lack of RTP activity in 301 seconds<br>After that the log show:<br>[Feb 28 11:41:12] WARNING[19197] chan_sip.c: Remote host can't match request CANCEL to call '6697777b27bb46ca01dc42b526adf7bd@Asterisk_IP_Address'. Giving up.<br>
<br>Did someone faced this issue before?<br><br>Thanks for help<br><br>Regards<br></div>