Le 26/03/2020 à 21:50, Kai Herlemann a écrit : Kai Hangup is h extension. your macro will never be executed. Solution: same = n,Dial(whatever) same = n,[…]) same = n,Hangup exten = h,1,1,DumpChan()  same = n,System(/home/asterisk/bash_test)..
Author : Administrator TOOTAI
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2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I cant compile asterisk having error   [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cbâ..
we just installed the latest 13 and 16 version of asterisk and face problem on incoming calls: they are ended like in Asterisk 16 [2020-02-04 19:19:48] ERROR[3768][C-00000001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization requi..
all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we dont know if they run kamailio- registration..
Hi
Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :
[…]
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[…]
—
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, on asterisk 13 I use same => n,Set(__myCpt=$[${myCpt} + 1]) which is working well. On an Asterisk 16 I get, for this same command [2019-10-01 16:15:01] WARNING[28197][C-00000008]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():syntax error: syntax err..
, I would like to now what is the sense of such type of entry in security.log [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: SecurityEvent=ChallengeSent,EventTV=2019-09-27T15:12:24.181+0200,Severity=Informational,Servic e=PJSIP,EventVersion=1,AccountI..
list, Im looking for a solution that can be applied to a stock asterisk 16 (pjsip if it matter) running Debian 9 (php7.0). Statistics should be available for normal calls and queues using a WEB interface. Open source better but not necessary, Any feedb..
I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk 13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also with PJSIP. Both LAN Asteriks are also connected via IAX. Everything is working fine except SIP call f..
all, I switched an old asterisk 1.8 to a new 13 version, stock version from Ubuntu 18.04 server. I did some modification in dialplan but after a reload they are not taken in account :(, even after restarting asterisk. I checked logs and found lots..