I use a python AGI which pulls some info from a web service, which should take half a second.Sometimes, it takes 5-10 seconds which blocks the dialplan execution, but the dialplan should continue immediately as its not dependent on the AGI/web serv..
all Im trying to limit the maximum concurrent calls on my Asterisk to try and mitigate another problem I posted about earlier. Ive edited /etc/asterisk/asterisk.conf And uncommented this line, and put a value of 60 in there: maxcalls = 60 in an eff..
guys Does anybody have any opinion on what causes tens of thousands of these messages per hour to pop up in the CLI: [Jun 30 14:24:59] WARNING: chan_sip.c:4057 __sip_autodestruct:Autodestruct on dialog email@example.com:5060w..
Cant find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how t..
While trying to use direct_media Im seeing RTP payload mismatches after succesful reinvites.Initial INVITE from endpoint A to asterisk has rfc4733 DMTFm=audio 35648 RTP/AVP 9 8 111 96a=rtpmap:96 telephone-event/8000a=fmtp:96 0-16From asterisk to upstr..
do you have someone example ofhttp://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/in node.js asterisk-ari ?t..
all,We packaged phpagi for CentOS 7 and Debian 8 (though nothing version-specific in those packages, I suppose).Packaging:http://git.xorcom.com/cgit/rpm/phpagi.git/Packages:* RPM: http://updates.xorcom.com/servers/ombutel/* Deb: should soon be in http://updates.xorcom.com/servers/spark/T..
Hi.I have some Nagios / Icinga monitoring plugins Ive created for Asterisk, and one of them checks the percentage of SIP accounts which are currently registered on an Asterisk server.It does this by running sip show peers via AMI and analysing the summ..
i have mix of realtime and static configuration of pjsiphttps://pastebin.com/YVFwVsMDpjsip.conf[global]endpoint_identifier_order=username,ip,anonymous user_agent=ipbx… transport definition extconfig.conf[settings]ps_endpoints => odbc,configDb ps_au..
I am using Asterisk 13.9 and using originate with early_media=true.I redirect these calls to an app that I wrote and it just write down audio before the answer. When I detect frame->subclass.integer =AST_CONTROL_ANSWER, app returns to asterisk nor..