Archives : July-2016
Im using Asterisk 13.9.1 (compiled from source) in Ubuntu 14.04.4.Im testing webrtc with jscommunicator (www.jscommunicator.org), and Im using wss. My problem is as follows:If I setdtls_cipher=ALLI can get audio with Firefox 47.0.1, but with Chrome 52.0.2743..
We use Asterisk and as per book we use MAC addresses as user names. So, when call coming in from outside (SIP trunk) – caller id is good.But when users calling each other on extensions – they see MAC addresses. How would I make it so we see actual na..
folks, I hope this is simple issue because it seems like something with registration expiration, etc. We use Asterisk (plain setup) with Cisco SPA phones (also nothing changed on settings, just proxy/UN/Password Everything on same LAN So, what we obse..
!Im facing a problem with the CPU consumption in Asterisk 11.22.0.I could decrease a lot of load, migrating both the astdb.sqlite3 and call recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and nodiratime flags), manually spread e..
Id like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8with chan_sip.If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packe..
Im debating between a cloud PBX or, perhaps, rasberry pi.For a SOHO, maybe three hardphones, rasberry pi would suffice?I would be amazed, but, if so, great.tha..
Generally, what am I looking for when turning SIP debug on?More specifically, the provider says that Im returning a 404 when they try to call me.Now, I had inbound working, literally, the other day.Outbound works fine.I may have broken it either thro..
I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration(telling me where the con..
Hello!Sometimes, I can see here the following scene:Asterisk sends 11 SIP OPTIONS-packages (qualify0) and they are all ignored by the peers – but the 12. package is answered immediately as expected (Im sure there is no network related problem).I ..
all,I am getting the following error when starting asterisk:pbx_functions.c: Function SHELL not registeredSome of my conf files use a SHELL command, which used to work with an older version of asterisk, but now with version 13.9.1 I see this warn..