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 '6697777b27bb46ca01dc42b526adf7bd at Asterisk_IP_Address'. Giving up. Did someone faced this issue before? Thanks for help Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090228/f0f548c8/attachment.htm
hi wich codec? once i had a similar problem it was a bandwith problem. how many calls in peak hours? i recomend you tcpdump and then analice the file using wireshark, you will be able to see if the rtp is coming too late or if it inst coming. you can also do the tcpdump in both side so you can see waht is goin out and what is arribing. if you are transcoding TOO much calls it can be a procesor problem. David 2009/2/28 michel freiha <michofr at gmail.com>> 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 '6697777b27bb46ca01dc42b526adf7bd at 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 >-- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090301/e42b26f2/attachment.htm
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 <michofr at gmail.com> 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 '6697777b27bb46ca01dc42b526adf7bd at Asterisk_IP_Address'. > Giving up. > > Did someone faced this issue before? > > Thanks for help > > Regards >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090301/df43fe0c/attachment.htm