! Running Asterisk 20.0.0 on CentOS 7, logging CDRs using cdr_adaptive_odbc to mariadb-server-5.5.68 (via mariadb-connector-odbc-3.1.7-ga-rhel7) Using chan_sip. Im facing the problem when there is a sudden spike of calls, some of the calls that are be..
Author : Markus
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, Im using Asterisk 11.25.0 and would like to set anonymous@anonymous.invalid as outgoing caller ID via SIP: Set(CALLERID(num)=anonymous@anonymous.invalid) However, when I look at the outgoing packet with tcpdump I see that the @ is not being transmit..
! Im getting this error frequently: ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed Right now, as a workaround, I reload Asterisk ..
, imagine a ConfBridge conference with 10 participants. Now, one of them suddenly starts to yell and scream. Is there any built-in functionality (maybe not in ConfBridge, but in Asterisk itself) to identify the loudest caller? Or maybe already built..
Asterisk 11.25.0 user here. Im trying to set up failing over to a second SIP peer if the first SIP peer doesnt answer on our SIP INVITE within 2 seconds. In sip.conf I set timerb=2000 for this peer, but it doesnt seem to have any effect. The time..
,Im using Asterisk2Billing (v2.0.16) and it appears to have an annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 (UK premium) and a fraudster manages to somehow dial 44-870 instead of 44870 the rate for 44 will match, not the ..
,n00b question, but I cant figure it out:[callthrough]exten => _+X.,1,NoOp(nothing here)#include blockedall.confexten => _+X.,n(hangup),Hangup exten => _+X.,n(nohangup),GotoIf($[${CALLERID(num)} anonymous]?nocli:cli)… more stuff that is handling ..
Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) ——– Weitergeleitete Nachricht ——– Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus An: universe@truemetal…
,probably this is a FAQ but I cant seem to find it. How to find the RTP IP address of an ongoing SIP call?sip show channels seems to list the RTP address under the very left column called Peer. And it also lists the associated Call ID which I could associ..
,I have a fresh install of Asterisk 12.0.0 and Im going to use it only as a client. Im trying to SIP REGISTER with a remote SIP provider.The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of ..