Stefan at WPF
2012-Jun-20 18:04 UTC
[asterisk-users] Proactive problem monitoring on SIP on Asterisk
Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. Thanks for any hints! Best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120620/8daecb68/attachment.htm>
Tim Nelson
2012-Jun-20 18:14 UTC
[asterisk-users] Proactive problem monitoring on SIP on Asterisk
----- Original Message -----> Hello,> 1) I am wondering what is the best practice to monitor if there are > or were problems with SIP calls on my Asterisk box. E.g. how about a > software that extracts all calls from the /var/log/asterisk/full (I > have permanently enabled verbose 10 and sip debug) log and tells me > on which of them were problems? Checking the logs manually is very > hard, but as SIP is a standardized protocoll, there should be tools > doing that for you? As an example, a person calling me recently got > a 488 Not acceptable error as reply from my Asterisk box. Nothing > came through to my SIP phone, so I didn't know anything about the > call or the problems (which were on his phone btw). I would like to > be informed about such cases, know that there was a call to my > Asterisk box that made problems.> 2) How about monitoring speech quality? E.g. sometimes it seems like > a packet is missing (I then have a short pause during the call), how > to monitor such things and create statistics out of this data?> So basically I want to monitor my Asterisk installation proactively > for reliability/problems and (speech) quality.Have a look at VQmonitor: http://www.manageengine.com/products/vqmanager/ It works very well. --Tim
Ishfaq Malik
2012-Jun-21 07:52 UTC
[asterisk-users] Proactive problem monitoring on SIP on Asterisk
On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote:> Hello, > > 1) I am wondering what is the best practice to monitor if there are or > were problems with SIP calls on my Asterisk box. E.g. how about a > software that extracts all calls from the /var/log/asterisk/full (I > have permanently enabled verbose 10 and sip debug) log and tells me on > which of them were problems? Checking the logs manually is very hard, > but as SIP is a standardized protocoll, there should be tools doing > that for you? As an example, a person calling me recently got a 488 > Not acceptable error as reply from my Asterisk box. Nothing came > through to my SIP phone, so I didn't know anything about the call or > the problems (which were on his phone btw). I would like to be > informed about such cases, know that there was a call to my Asterisk > box that made problems. > > 2) How about monitoring speech quality? E.g. sometimes it seems like a > packet is missing (I then have a short pause during the call), how to > monitor such things and create statistics out of this data? > > So basically I want to monitor my Asterisk installation proactively > for reliability/problems and (speech) quality. > > Thanks for any hints! > > Best regards > Stefan > --I've not used this myself but had a look at the site and I think it's pretty much what you're after... http://www.voipmonitor.org/ Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at 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
Jamie A. Stapleton
2012-Jun-22 18:58 UTC
[asterisk-users] Proactive problem monitoring on SIP on Asterisk
ADTRAN has some interesting Voice Quality Monitoring built into their switches, routers, etc: http://adtran.com/web/url/vqm From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan at WPF Sent: Wednesday, June 20, 2012 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Proactive problem monitoring on SIP on Asterisk Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. Thanks for any hints! Best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120622/5868d5b7/attachment.htm>