Hello.Voice quality when calling – this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.Because the overall quality of a call can only be determined upon completion, I did it in the HangUp hand..
, this is a bug?ERROR: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect clie..
Kindly guide with debugging TLS issue in asterisk 11.16. Compiled from source and works all ok !Added the below to sip.conftlsenable=yes tlsbindaddr=0.0.0.0:5061However asterisk doesnt even listen to port 5061sudo netstat -anpKindly guideThanks Be..
Hey guys, have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtpin endpoints table “direct_media..
Guys I have a 4 port PRI card that I need to setup each port in their own group.In chan_dahdi.conf I have the following which works for one portHow do I add the rest of the ports in their own groups so that I can have different signaling on each?[channels]language=enswitchtype=euroisdnpridialplan=unknownresetinterval`0echocancel=yesechotraining=yes;echocancelwhenbridged=no;rxgain=0;txgain=0callerid=asreceivedm..
All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provi..
I am in need of information about how to configure the sip.conf and extensions.conf for subscribers to support the dialog-info event package rfc 4235. I am using Asterisk 126.96.36.199 version. Also please inform if the phone must have the support for t..