<br><br><div class="gmail_quote">2009/3/1 michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com">michofr@gmail.com</a>></span><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">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<div><div></div><div class="Wj3C7c"><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" target="_blank">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>
</blockquote></div><br></div></div></div>
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<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote><div><br>Check your opensips config, besides network check, you need to be sure that your sip messages are not redirected to wrong servers, sometimes, if it the case some call legs are messed up, and wrong cancels go to wrong servers... I saw that with bad NAT config on opensips it was passible.<br>
</div></div>Good Day<br>