Re: [asterisk-users] Logging SIP connection status for review
On Wed, 2013-04-10 at 11:06 -0700, Carlos Alvarez wrote: On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards asterisk@sedwards.com wrote: dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. That's completely different. We already run a good packet capture system. What I want to see is SIP registration statuses and latency logged about once a minute. We do that now by doing a 'sip show peers like x' and putting it in a text file. I can then correlate issues with times of high latency or unreachable phones. I'd just like to see more reporting and the ability to correlate times and such. How about using your current scripts and then pushing the data into Graphite? http://kaivanov.blogspot.co.uk/2012/02/how-to-install-and-use-graphite.html Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging SIP connection status for review
Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging SIP connection status for review
On Wed, 10 Apr 2013, Carlos Alvarez wrote: Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging SIP connection status for review
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards asterisk@sedwards.comwrote: dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. That's completely different. We already run a good packet capture system. What I want to see is SIP registration statuses and latency logged about once a minute. We do that now by doing a 'sip show peers like x' and putting it in a text file. I can then correlate issues with times of high latency or unreachable phones. I'd just like to see more reporting and the ability to correlate times and such. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging SIP connection status for review
http://www.artifact-software.com/?page_id=1666 Would this help? Put a JasperReport graph or two in a report step. Ron On 10/04/2013 2:02 PM, Steve Edwards wrote: On Wed, 10 Apr 2013, Carlos Alvarez wrote: Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. dumpcap can capture all of the SIP (and RTP) packets into a series of files without a huge performance hit. A cron job can pbzip2 the files and delete if over x days old. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging SIP connection status for review
On On Wed, 10 Apr 2013, Carlos Alvarez wrote: Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within a range, etc. How about munin http://munin-monitoring.org/ With these plugins http://www.venturevoip.com/news.php?rssid=2322 You could easily adapt one to do registrations. I find the sip peers and calls in the last hour quite interesting Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
