Re: [asterisk-users] moh reload not reloading/reading new musiconhold files
Hi Jonas Wouldn't this do the job? touch /etc/asterisk/musiconhold.conf ; asterisk -rx 'module reload res_musiconhold.so' Pete > On 31/03/2017, at 8:55 am, Jonas Kellenswrote: > > > I would not know how to automate this through script... > smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] CDR reporting solution
I installed CDR-Stats on Debian 8.7 http://cdr-stats.readthedocs.io/en/latest/installation/install-cdr-stats.html I am trying to figure out how to import flat CSV file to CDR-Stats On Wed, Mar 22, 2017 at 6:28 PM, Bruce Ferrellwrote: > How about CDR to either MySQL or cdrlite and a quickie sql query? I could > have added postgres, but I'm a DB bigot. That would work too. > > > On 03/22/2017 01:46 PM, Motty Cruz wrote: > >> >> Hello, I am looking for CDR reporting solution? Any suggestions? I am >> using Asterisk 13.13.1 >> >> I would like a report on number of calls per extension. >> >> Thanks, >> Motty >> >> >> >> > > -- > _ > -- 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 > -- Thanks for your support, Motty -- _ -- 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] moh reload not reloading/reading new musiconhold files
Hello I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open with vi and close again) and then issuing a 'module reload res_musiconhold.so' on the Asterisk CLI makes the new files load into Asterisk. Very strange !! I would not know how to automate this through script... Kind regards. On 24-03-17 12:29, Daniel Journo wrote: > Hello > as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. > Only at restart the new files are available. > Is this a bug ?? How can I get more debugging for this problem ?? I think there is currently a bug with MOH. For now, if you add a file to a moh folder, ‘touch musiconhold.conf’ and then reload moh. Please let me know how it goes. Kind regards Dan Journo -- _ -- 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] gcontacts to asterisk
hello everyone. i am looking to automate the management of contacts to my system (debian 8, with asterisk 11). at the moment i do create the astdb with database put cidname. I have searched a bit i have found the google contacts integration https://zmonkey.org/blog/content/google-contacts-asterisk-caller-id the problem is that it does not run at all. my system also has mysql running that i could use. has anyone any working solution for google contacts in the system, please? -- _ -- 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)
Hi Ikka, The last time I had this kind of problem the numbers in DB were altogether different and the reason for that was inappropriate columns data-types in DB. You can also print out some CDR(${variable}) AFAIK in the dialplan and verify that those are in original condition there or not. Regards, Sammy On Thu, Mar 30, 2017 at 10:41 AM, Ikka Tirtawidjajawrote: > 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 ? > > > Thanks in advance, > > > Ikka > Jakarta - Indonesia > > > -- > _ > -- 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 > -- _ -- 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] Alphabet character in destination number (CDR)
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 ? Thanks in advance, Ikka Jakarta - Indonesia -- _ -- 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] Asterisk 13.14.0. Debugging DTMF issues
Hello, I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing issues with DTMF. Installed version is 13.14.0. 1. In outbound calls SDP, I'm seeing these kind of lines: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 I would expect events to range from 0 to 15, not to 16, as seen in rfc 4733 examples. What is this event 16 for ? Is there a way to configure this ? 2. With channel originated calls using Local prefix, I can read this on console: [2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4103 __ast_read: DTMF begin '#' received on PJSIP/Foo-006f [2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4114 __ast_read: DTMF begin passthrough '#' on PJSIP/Foo-006f [2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4103 __ast_read: DTMF begin '#' received on Local/2@from-originate-0024;1 [2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4114 __ast_read: DTMF begin passthrough '#' on Local/2@from-originate-0024;1 [2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4017 __ast_read: DTMF end '#' received on PJSIP/Foo-006f, duration 180 ms With pure inbound-outbound calls (calls coming in from PJSIP and leaving through PJSIP), I get this: [2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4017 __ast_read: DTMF end '9' received on PJSIP/Bar-IPO-0093, duration 100 ms [2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4044 __ast_read: DTMF begin emulation of '9' with duration 100 queued on PJSIP/Bar-IPO-0093 [2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4181 __ast_read: DTMF end emulation of '9' queued on PJSIP/Bar-IPO-0093 Can I get both inbound and outbound DTMF on console ? How ? 3. Looking at DTMF duration (as logged by Asterisk console), I can see that some (from mobile phone) have a 180ms duration while some, from an other SIP trunk, have a 100ms duration. Can I configure tone duration ? How ? Should I configure this ? Does this duration any real relation with the way a user presses keys ? Best regards -- _ -- 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] Logging endpoint IP address when PJSIP registration fails
Hi! I am seeing a lot of warnings of these types: res_pjsip_registrar.c: AOR '31' not found for endpoint 'anonymous' I am guessing these are coming from a scanner trying to scan for the extensions on the asterisk server. Is there any way to print the IP address of the endpoint trying to register an extension using PJSIP in asterisk 13? I can then configure fail2ban to temporarily hinder the scan. Regards, Sree -- _ -- 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