Hi all.... 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.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c: Disconnecting call 'SIP/netcafe2-b7da99b8' for lack of RTP activity in 301 seconds After that the log show: [Feb 28 11:41:12] WARNING[19197] chan_sip.c: Remote host can't match request CANCEL to call '6697777b27bb46ca01dc42b526adf...@asterisk_ip_address'. Giving up.
Did someone faced this issue before? Thanks for help Regards
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
