Archives : November-2017
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When a call is starting, Asterisk starts sending and receiving RTP packets. Each packet has a sequence number.1. Instead of logging everything as rtp set debug is currently doing, is there a way to only log:- the sequence number of the first recei..
AllI have an Asterisk 1.8.32.3 instance that will at random intervals stop logging CDR data to MSSQL via FreeTDS.On investigation Ill find that the FreeTDS module has been unloaded somehow. It is not listed in cdr show status or show module like.Try..
To troubleshoot Voice Quality issues and automate testing, Im planning to use Asterisk as a softphone alternative (with better scripting capabilities) on a Linux 4.10-enabled Ubuntu laptop. This laptop has two audio jacks with a compatible headset..
Quick one here: is it possible to setup my dialplan on such a way that the MoH while waiting to be answered by an agent on a Queue be different than the one that I will hear when that agent puts me manually on hold for a few minutes AFTER I finally ..
Asterisk Project Security Advisory – AST-2017-011 ProductAsterisk SummaryMemory leak in pjsip session resource Nature of AdvisoryMemory leak SusceptibilityRemote Sessions Severity Minor Exploits KnownNo Reported OnOctober 15, 2017 Reported ByCorrey FarrellPos..
The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15.The available security releases are released as versions 13.13-cert7, 13.18.1,14.7.1 and 15.1.1.These releases are available for immedi..
I am trying to block/hide outgoing caller id on a PRI E1.It seems that it should be fairly simple, but it is defeating me.https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says:to hide your caller id, use: Set(CALLERID(num-pres)=prohib)T..
I just tried to compile the latest Asterisk 13.8.0 and it stopped with several errors on pjsip.So FYI if you run the install_prereq script and then use ./configure –with-pjproject-bundled you will have the same problem because the prereq script insta..
Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this :- I want to limit the number of calls on a given SIP peer to 10- on the other hand, some calls have higher priority than others- w..
Im running Asterisk 15.1.0 and in the process of converting my various SIP endpoints to use PJSIP.My Zoiper client causes the messages quoted below to show up on the CLI once per minute.Things seem to work OK, but I am curious because there seems..