Hello!It there way to limit script execution time ?I did something wrong writing my script yesterday , finally got it working,but found that there are busy ISDN channels, looks like these are with hang scripts…..
Hello!I run two asterisks 13.13.1.Here is how they are connected:me—PBX–isdn pri–asterisk1–sip–asterisk2.If I call something from asterisk1 and have in dial plan:Lets sayexten => 6000,n,Set(CONNECTEDLINE(name)=Conf. 6000)exten => 6000,n,Meetme(6000,TL(10800000:60000))T..
Hello!Upgraded 13.10 to 13.11.1 today and now I see messages in log:[Sep9 12:23:16] NOTICE res_pjsip/pjsip_distributor.c: Request REGISTER from 3563 failed for 192.168.32.116:5060 (callid: email@example.com) – No matching endpoint foundor[S..
Hi!Installing new asterisk server and decided to use chan_pjsip.While module reload I get:y 12 15:33:04] ERROR: config_options.c:715 aco_process_var: Could not find option suitable for category 3567 named inband_progress at line 867 of[May..
Hello!I need to use n-way call as it described here:http://habrahabr.ru/sandbox/52259/It is in russian, but dial plan is quite clear. It works in asterisk 11: — Blind transferring OOH323/7272-6385 to 0 (context fromtransfer) priority 1 — Execut..
Hello!We have asterisk connected over PRI no our phone network, so Im avaya PBX user. Asterisk connects to another avaya system over h323.Connection can be shown asavaya–PRI-asterisk–h323-avayaWhen I do call as avaya user I see name of remote end a..
Hello!I see large enough amount of such messages on one of our asterisks. There are no complains from users, so I may be they are harmless. Could you tell me what can it be?..
Hello!As I see there isdsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence?Is it possible to change silence level, so, lets say some not loud enough background noises will be recognized as silence and only loud eno..
Hello!Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: — Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6, SIP/6166@asterisk) in new stack == Using SIP RTP TOS b..
Hello!I want to change call files, which has caller id in them, to call originate from dial plan. But I dont see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_OriginateHow can I pass callerid to following:ex..