is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?or is chan_pjsip better supported?or the recommended way for asterisk is use respoke.io?my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)chan_sip.c:10496 process_s..
Author : Marek Cervenka
Home » Articles posted by Marek Cervenka (paged 3)
im facing problem with multiple calls to one agent when Local channels are used wireshark shows multiple invites to the agents phoneused versions asterisk 1.8/asterisk 13agents are logged dynamically. interface state based on hintsqueue configuration..
bounty offer prolonged to 31.4.2015 (end of april)Dne 3.3.2015 v 16:22 Marek Cervenka ..
im searching second BOUNTY donor ($250)for https://issues.asterisk.org/jira/browse/ASTERISK-22708if you want participate, please contact me..
is it possible use asterisk static realtime and config files simultaneously in asterisk 11?i want [globals] from extensions.conf in database, but dialplan in extensions.conf config filei sawthis can be configured in stasis.conf in asteris..
im converting extensions.conf to DB routing. can you help me with regexp or something which converts dialplan to single numbers like _3X0 to 310,320,330,340,… ?i found only https://pypi.python.org/pypi/asterisk_dialplan/0.1.2 but i need the oppos..
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 a..
i upgraded few asterisk systems from last asterisk 1.8 to 11 and i see in graph that cpu usage is ~50% higherany ideas? configuration, modules, .. i..
hi.i have dialplan with 2 simultaneous calls – dial(sip/phone1&sip/phone2).when i cancel call on phone1 (push reject button), the call is still ringing on phone2can i cancel call on both phones from one place(one pho..
i want convert mixmonitor recorded speech audio from wav to mp3 or aac can you recommend your settings for speech audio? filters, noise elimination, compression ratio, …i will probably use lam..