[asterisk-users] how to hangup this channel "Message/ast_msg_queu
omega*CLI> core show channels Channel Location State Application(Data) Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u) "Message/ast_msg_queu" it's been up for the last day, how to hangup this channel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alphabet character in destination number (CDR)
J Montoya or A J Stiles wrote: On Thursday 30 Mar 2017, Ikka Tirtawidjaja wrote: Dear all, I have PBX with asterisk 13.x a couple of IPPhone that connect to that asterisk PBX send an alphanumeric dialed phone number. for example, in my CDR table, field DST, it show dialed phone number like - 0C81318304632C (it should be 081318304632) - 08D11157112 (it should be 0811157112). Why it's happening ? and how can I prevent it to happen ? A, B, C and D are actually valid DTMF digits (they belong in a column to the right of 3, 6, 9 and # respectively, and have the "high" frequency 1633 Hz). TTBOMK they were never actually used for anything in practice, they just keep kicking around like a vestigial organ (compare how computer software for the UK financial industry still includes code to deal with mediaeval pounds, shillings and pence). Is it possible for a 1633 Hz tone, loud enough to swamp the "high" frequency of a dialled digit, to be finding its way somehow into the microphone of the affected phone and confusing it when digits are dialled? A,BC & D were used in the US in the AutoVon Military system, and telephones that have that keypad often bring pretty good money. Some external Voice mail systems use one or more of the codes as well ( Toshiba systems for one ) Probably because they are an easy way to prevent most users from messing around in the system It was and in some cases still used for network signalling. Not that computer telephony designers believe in standards but the 16 button tone pad is an ITU-T standard. I believe in the UK it is known as MF-4 Asterisk DTMF detection leaves a lot to be desired. It certainly is not Exchange grade. There may be a way to diddle the code to remove the false detection but this should only happen with in-band signalling.I thought all SIP data was normally sent as data though sip.conf does allow in-band John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any way to clear ALL gosub stacks without knowing what they are?
On Sat, 1 Apr 2017, Jonathan H wrote: Any way of clearing ALL gosub stacks in dialplan? 1) rm -f /etc/asterisk/extensions.conf? 2) hangup()? (It is April fools...) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with Asterisk 13, multiple CDR per queue and arbitrary upper limit
On Fri, Mar 31, 2017, at 10:55 PM, Michaël Gaudette wrote: > > > Hi, > > > > I`ve recently upgraded a server from 1.8 to Asterisk 13. While > everything > is under control, I have one issue with the way CDRs are kept for queues. > And I don`t mean “I don`t like it”. I mean it crashes the server. > > > > I realize there are multiple CDRs per queue call – one per ring/per > phone, > basically. The issue is that whenever the number of CDRs “to be > recorded” for a call exceeds 5000, Asterisk becomes unresponsive for a > few > minute. I get this message in the console: > > “taskprocessor_push: The 'subm:cdr_engine-0003' task processor queue > reached 5000 scheduled tasks again.” > > > > This scenario is trivial to reproduce: a queue, with simultaneous ring, > 20 > phones, all unreachable, 1 second between attempts. After 250 (5000 > divided by 20) seconds of waiting asterisk partially breaks down. > > > > This seems to be because while multiple CDR`s are written per queue call, > it`s only done at the end of the call, so CDRs accumulate in > memory/cacher/whatever and break some limit. > > > > So, my question is: is there any way to force the CDR`s to be written as > the queue app is working it`s magic, instead of at the very end of the > call? Or anyway to work around this limit? Or any fix for this? There is not. If you are running the latest version I'd suggest filing an issue[1] as we definitely should not crash under the scenario. [1] https://issues.asterisk.org/jira -- 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any way to clear ALL gosub stacks without knowing what they are?
Any way of clearing ALL gosub stacks in dialplan? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
