Archives : November-2019
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I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013]pbx..
With Debian Busters asterisk package, what can you use instead of Digiums contrib/scripts/ast_tls_cert ?If that matters, this is for using WebRTC and Cyber Mega Phone 2K (both on the same box) in a private LAN environment.My intent was to use easy-..
Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)}TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity..
All.I have an interesting situation.I have two Asterisk servers and one Asterisk Voicemail Server, which serves voicemail to both boxes.My question is – how do I get the MWI to work for the end users since the two Asterisk servers do not have voicem..
how would one set up SIP messaging from one server to another server and based on the reply back from the server – perhaps do something else .I think I can do receiving the SIP message – but how about the response back. I was looking around and h..
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Gang I have stumbled over a strange issue with Asterisk 13.18.3 I have two interfaces, two different IP Addresses. One facing to the internet, and one facing to am internal voice lan. Therefore I defined two different transports and endpoints: [transport-udp-intern..
We have a box running Asterisk 11. A call comes in and the caller wants to use INFO (and the peer is set as INFO). We send the call out to a carrier were we specify rfc2833 and negotiate it correctly. In theory Asterisk should see the DTMF in rfc2..
we want to extract the information when the most callers are entering our phone system based on an interval of 15 minutes. this is quite simple (although not perfect) with select calldate, count(*) as anzahl from cdr where calldate > 2019-10-12 gr..
when using some non dynamichost eg. host2.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or featur..