search for: cervenka

.../var/lib/xen/test.img,hda1,w'', ''tap:aio:/var/lib/xen/test.swap,hda2,w'' ] blktapctrl do nothing domU cannot start, fail when mounting root what i''m doing wrong? i dont have any kernel module for blktap, is this ok? --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz ======================================= _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-u...

Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a): > Hello, > > Le 2015-08-06 09:24, Marek Cervenka a ?crit : >> hi, >> >> there is updated skills based routing patch for asterisk queue >> please test if you have time >> >> https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 >> >> > > You can find the latest ver...

...i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek Cervenka =======================================

Hi all, For anyone that's interested in setting up TLS with libvirt, but hasn't gotten around to it yet, there's a decent guide on the wiki now: http://wiki.libvirt.org/page/TLSSetup It's user focused (ie SysAdmin's), and has lots of pics explaining concepts, plus shows what needs to be done, and shows how to do it. Please note, it doesn't cover SASL, it's just

...rver = localhost UserName = top Password = secret Port = 3306 Socket = /var/lib/mysql/mysql.sock #SSLCIPHER = AES256-SHA Charset = utf8 -- --------------------------------------- Marek Cervenka ======================================= -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160527/cf8dda5b/attachment.html>

...x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287 if (t->msg_magic == msg->hdr.magic && it was only once after 2 days. i dont know how to repeat it now :( any similiar experience? -- --------------------------------------- Marek Cervenka ======================================= -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160526/0e31b760/attachment.html>

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

hi, can you recommend "clean&simple&stable" solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --------------------------------------- Marek Cervenka =======================================

...y=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --------------------------------------- Marek Cervenka =======================================

bounty offer prolonged to 31.4.2015 (end of april) Dne 3.3.2015 v 16:22 Marek Cervenka napsal(a): > hi, > > i'm offering bounty[1] $500 (five hundred) US dollars for resolving > https://issues.asterisk.org/jira/browse/ASTERISK-22708 > > fix must be available for asterisk 11.x and asterisk 13.x and accepted > to upstream > As part of this fix we should see...

hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 -- --------------------------------------- Marek Cervenka =======================================

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cervajs2 at gmail.com> wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3 /var/lib/aste...

...ret >> Port = 3306 >> Socket = /var/lib/mysql/mysql.sock >> #SSLCIPHER = AES256-SHA >> Charset = utf8 >> >> >> >> -- >> --------------------------------------- >> Marek Cervenka >> ======================================= >> >> > > > -- > --------------------------------------- > Marek Cervenka > ======================================= > > -- --------------------------------------- Marek Cervenka ================================...

.../ASTERISK-25957 - a > solution is to change the order in modules.conf to: > > preload => res_odbc.so > preload => res_config_odbc.so > preload => chan_local.so > preload => cdr_adaptive_odbc.so > > /niklas > -- --------------------------------------- Marek Cervenka ======================================= -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160529/675bc809/attachment.html>

The time appears to be perfect inside dom0, however all the domU''s tend to have a slightly faster date which gets further out of sync every day. I''m currently using Xen 3.0.3 with Gentoo Linux, under 3.0.2 I had no problems with domU clocks. Are there any known issues which could cause this? I''d strongly prefer not to run ntpd in every domU, having all domU clocks in

hi, is it possible connect call to queue to specified agent? like Mr. Neo called helpdesk queue, call picked by agent Smith Mr. Neo is calling again and i want connect him with agent Smith -- --------------------------------------- Marek Cervenka =======================================

...LS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? or howto debug this problem? asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox) -- --------------------------------------- Marek Cervenka =======================================

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --------------------------------------- Marek Cervenka LCNA - http://lcna.slu.cz =======================================

...because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) * uncompressed PCM * ADPCM * AAC can you someone recommend solution/combination which works? tnx --------------------------------------- Marek Cervenka =======================================