similar to: playing media(moh,prompts) from flash player

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hi, I am using a 64-bit RHEL 5 machine. I built Asterisk latest 1.6 branch. The system has ODBC and Postgres installed. psql, isql and odbc work fine. Asterisk "make menuselect" for some reason does not see the installed packages and refuses to build res_odbc and other packages. How do I force it to do that? Is there a way to modify the output file from menuselect and make it

hi, i''m trying move from file: based disk to tap:aio but things don''t work i have centos4 dom0 with centos4 domU xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled [root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config CONFIG_XEN_BLKDEV_TAP=m config disk = [ ''file:/var/lib/xen/test.img,hda1,w'',

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

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

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

hello, is there some way to notify people in the same pickup group about call from caller to callee? 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

hi, i have the same problems as in https://issues.asterisk.org/jira/browse/ASTERISK-25833 my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1 i tried update to mysql-connector-odbc 5.3.6 from oracle but it segfaults every time can you share your working mysql+odbc combination? thanks odbcinst.ini

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

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 =======================================

Hi, I''m struggling to get the validates_associated to work as I think it should be. I''m using: JRuby 1.1 rails-2.0.2 activerecord-2.0.2 activerecord-jdbc-adapter-0.8.2 My tables in MySQL: CREATE TABLE area_codes ( id INT UNSIGNED auto_increment primary key ... ); CREATE TABLE markets ( id INT UNSIGNED auto_increment primary key, ... ); CREATE TABLE

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

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 =======================================

doesnt work for me Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a): > Hi, > > > On 2016-05-27 18:28, Marek ?ervenka wrote: >> after downgrade to 13.8.2 >> May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip >> b49162cd sp bfac0940 error 4 in >> libmysqlclient.so.16.0.0[b48f1000+12e000] >> >> after downgrade to 13.7.2 >>

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

I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor

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

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/asterisk/realtime.sqlite3 > > CREATE TABLE