Asterisk,Ive been running CentOS since 2006 or so and support for the 32bit version recently ended. CentOS no longer offers a 32 bit version so I thought Id try Fedora 26 as they have 32 bit and support. Got it installed, then downloaded Asterisk 14…
Ive recently setup a small load test against an instance of Asterisks.Ive tested on asterisk 13.5 and 14.6 with the same results.I am using PJSIP.My dial plan is,[test]exten => 1001,1,Answerexten => 1001,n,MusicOnHold(15)exten => 1001,n,HangupI am us..
According to the instructions given at https://www.asterisksounds.org/deI converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time.But unlikily, some..
Weve had dozens of Polycom 3.x firmware phones deployed and working great for years. Now Ive finally been charged with the long-overdue task of figuring out why newer Polycom devices with 4.x firmware register fine but do not respond to SIP OPTIONS requ..
Im using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity ar..
all,Lately, Ive seen an increase in the number of attacks against my system from the so-called Friendly Scanner.When one of these script kiddies targets my server, all I see for symptoms is a few of my trunks become lagged due to server load and a str..
Hello.Is someone here using VoIPmonitor?I am using just the sniffer and I found some pcap files that contain some odd streams.For example, I have a file with 3 streams, but the weird stuff is that 2streams are the same (e.g., have the same source addr..
all,I had a box with CentOS 6… I backed up, installed C7. restored my backups, put back on asterisk 11.25.1 put back my configs and ran a test… All seems good, my device activates like audio is ready to come out – but no audio. CLI looks like everyth..
While debugging a SIP trunk with an Avaya IPO, I noticed that wikis PJSIPdtmf_mode at  includes:This setting allows to choose the DTMF mode for endpoint communication.rfc4733 – DTMF is sent out of band of the main audio stream. This supercedes ..
It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. Its a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or..