Archives : August-2017
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Let me provide the details first:* Asterisk 1.8.32 on CentOS behind the NAT firewall* Two (2) SIP trunks with canreinvite=no and directmedia=noIf a call comes from either trunk and is bridged to a local extension there is never a problem with aud..
folks.I have a couple of questions regarding RTP.The background of my inquiry is that I have packet captures of SIP and RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has a time stamp that rewinds by 480 using g.711u. ..
since al long time I have used UNIQUEID for identify calls in my dialplan, statistics…Now I have had an problem, after I have checked log file I found out following:calls same time ( hours:seconds) came in.CallID, DID, channel name (00003cf9 to 00003c..
Ive had two Asterisk crashes today that seem to be caused by errors where chan->tech_pvt is pointing to something that cant be deallocated and I think I see a reference count bug in the above function.It contains:if (data->chan_old_vsrc) {ast_channel_unref(data->chan_old_vsrc);}Shoul..
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..