Re: [asterisk-users] nagios asterisk check SIP
On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote: > Iam loocking for an programm to check the SIP port of an Asterisk asterisk. > > Ome time ago I have used > #/usr/bin/sipsak > but it seemed that it is not working anymore? Hi Thomas, Maybe this links help you: http://fabian-affolter.ch/blog/nmap-scripts-for-voip-analyses/ Not for sipsak, but for great nmap. Cheers -- GnuPG Key ID: 0x39BCA9D8 https://www.github.com/mefhigoseth ...:::[ God Rulz ! ]:::... -- _ -- 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] Setting realm=blabla in sip.conf ignored ?
Hello no matter what I set in sip.conf for the param "realm=blablabla" , I notice in a wireshark trace file that the realm is completely ignored. I see that realm value is still 'asterisk', being the default. Why is this ? (I would like to add a printscreen of the wiresharl trace but then this thread is rejected due to message size) So how can I really change the realm value ? Thanks. Jonas. -- _ -- 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] nagios asterisk check SIP
On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote: > Iam loocking for an programm to check the SIP port of an Asterisk asterisk. > > Ome time ago I have used > #/usr/bin/sipsak > but it seemed that it is not working anymore? What is the problem with sipsak? /usr/bin/sipsak -s 'sip:10@somehostname' -r 5060 -vv -S -- Stefan Tichy ( asterisk3 at pi4tel dot de ) -- _ -- 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] Is it possible to set a timeout when querying a calendar ?
Hello, Asterisk offers several ways to query an external calendar. I'm wondering what could happen if an external calendar was down or very slow to respond. Looking at wiki.asterisk.org, I didn't find any timeout parameter in calendar functions, nor in calendar.conf (but I'm not too sure for the later case) ? What would you suggest to protect your dialplan from a mis-behaving external calendar ? Best regards -- _ -- 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] update from Asterisk 12 to 13
Saint Michael wrote: I am using Asterisk 12 and PJSIP. Last night I tried to upgrade to Asterisk 13, and it did compile just fine, but PJSIP would not load, and no error was shown on the screen when I did "pjsip reload". Do I have to erase some objects before compiling Asterisk 13? Is there a document that shows steps to a successful migration? You should always clear your modules directory to ensure there's no old leftover things, besides that it should be fine. Were there any errors output in the console when loading? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] update from Asterisk 12 to 13
I am using Asterisk 12 and PJSIP. Last night I tried to upgrade to Asterisk 13, and it did compile just fine, but PJSIP would not load, and no error was shown on the screen when I did "pjsip reload". Do I have to erase some objects before compiling Asterisk 13? Is there a document that shows steps to a successful migration? -- _ -- 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
