<div dir="ltr">Dear David,<br>I'm using G729 pass though mode...No transcoding is used here<br>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...<br>
<br>I'll use tcpdump next time I face such issue<br><br>Regards<br><br><br><div class="gmail_quote">On Sat, Feb 28, 2009 at 7:21 PM, michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com">michofr@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><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>
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