Dear David, I'm using G729 pass though mode...No transcoding is used here Regarding concurrent calls, I have 3 asterisk servers working in load balancing mode...The issue that the same problem appear on 3 asterisk...each asterisk handle around 150 calls...
I'll use tcpdump next time I face such issue Regards On Sat, Feb 28, 2009 at 7:21 PM, michel freiha <[email protected]> wrote: > 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 >
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