Re: [asterisk-users] mysql CDRs in web based tool
i have uploaded two files for you one is the php script that reads Master.csv and loads data in the database you have to change $locale_db_name = 'databasename'; $locale_db_login = 'user'; $locale_db_pass = 'password'; this how you run the script php /path/cdrmysql.php /var/log/asterisk/cdr-csv/Master.csv path is the location where your cdrmysql.php is located second is the database already truncated for you hope this helps Regards. Kyeyune Bob VOIP Engineer +256 774 702 258 On Wed, Sep 25, 2013 at 5:02 PM, Doug Lytle supp...@drdos.info wrote: but i do not know how to interface the CDRs. has anyone used this tool or any other similar tool? It expects your CDR to be located in a mysql database. You'll either need to figure out how to import your .csv into mysql, or have Asterisk send the CDR directly to the mysql database. Doug -- _ -- 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 database.sql Description: Binary data attachment: cdrmysql.php -- _ -- 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] WebRTC softphone for Asterisk - any suggestion?
hello; hopefully u can help me i have asterisk vanilla installation 11 and i have also managed to install webrtc2sip how do i make asterisk to communicate with webrtc2sip c'se right now both run independently Regards. Kyeyune Bob Network IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Fri, May 31, 2013 at 11:36 PM, Adnan 112linuxstockh...@gmail.com wrote: Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.comwrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- 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 -- _ -- 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] Can't register to Asterisk 1.6 with old Aastra phones
am also stuck with Alcatel lucent IP Touch 4018 any one connected them to Asterisk thanks Regards. Kyeyune Bob Network IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Sun, Apr 28, 2013 at 11:56 PM, Carlos Alvarez car...@televolve.comwrote: We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from 'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found Typically of course we'd expect to see: sip:abc123@server We're running the latest available firmware, but it's from 2009. Any ideas on this before we just trash all the older phones? -- 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 -- _ -- 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] send record file to email
Hello; how do i embed and send the recorded file to email automagically exten = _1XXX,3,MixMonitor(${CALLFILENAME}|b|/usr/sbin/wav2mp3 ${CALLFILENAME} ${peeremail} ${EXTEN} ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} ) Regards. Kyeyune Bob Network IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Fri, Feb 1, 2013 at 4:39 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 01 February 2013, Joseph wrote: When recording the filename I use: exten = _NXX,n,Set(recordfilename=${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y_%m_%d_ %H%M)}.wav) This works OK but I would like to add to the name the local extension from which the calls is originating, what is the variable called? ${CALLERID(num)} has the originating extension number, unless you changed it e.g. to show your direct line on an outgoing call (but you can always do that *after* you generate the filename for the recording). -- AJS Answers come *after* questions. -- _ -- 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