Archives : May-2020
Home » 2020 (paged 18)
In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law.We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are ..
I have an endpoint with multiple phones registered as aor contacts.When I attempt to originate a call it will only ring one of the phones.Is it possible to ring multiple phones as a single endpoint.First phone to answer wins the call and all others terminated?Aga..
all,We would like to pull the RTT of registered endpoints from MySQL for use in a webportal. However it doesnt appear asterisk tables this by default like chan_sip did.Ive found some information [1] that a modification of sorcery.conf can get it writ..
I use Asterisk 13 with FreePBX. When I try to connect my Softphone via VPN to Asterisk Im registered and Its show via pjsip list contacts Then I try to call an internal number / other extension I get the following: SIP/2.0 401 Unauthorized. The VPN ..
Ive seen that Asterisk 17 supports Prometheus but beside [1], Ive not much about how to use this.Can someone shed some light on this ?1. If Im not mistaken, Prometheus favors a pull model over HTTP. So basically, a Prometheus instance should be a..
Everybody,Ive been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didnt requite installing connectors on anything or adding tables on the DB server.Its based off of PHP5 and the only reaso..
there I have a pbx (v16.10) on AWS (Ubuntu 18.04) with Freepbx (14) that I am trying to set up the proxy reSIProcate on the same host as pbx. I can make it all work when the proxy is on a different host but when the proxy is on the same host aster..
We have an Asterisk 11.3 server where we want log rotation handled purely by Linuxs logrotate, and not by Asterisk. To this end weve configured the[general] action of /etc/asterisk/logger.conf with:rotatestrategy = noneHowever, an asterisk -rx log..
My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below:force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=inviteBut when I send a call I see the RTP being sent..
I want to see the help when I type core show application xxxx, and its not available. This is asterisk 16 from sources. I have libxml2-devinstalled. Ubuntu 19What am I miss..