I have an Asterisk box which has an IVR that plays random gsm files. The box has SSDs and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPUusage along with the load seems to jump around. With about 500 callers it hovers between 250-400% CPU ..
Author : Dovid Bender
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All,Does anyone have stats on how many callers they can get on a single ConfBridge per core? I was testing on Digital Ocean with 30 users in a room and I was shocked by how well it kept up with 40 callers. I am trying to figure out how scalable it..
I am using the ICES application and one issue we are having is the carrier is timing out because we are not sending RTP. I did try RTP keepalive and that did not help. Is anyone aware of a way to have Asterisk send a fake RTP packet (as in a real ..
We have a requirement to build a cluster that can handle 30k calls. The system is going to play one of 15,000 sound files. In the past we had no issue with Asterisk doing a few hundred calls. When we went above that Asterisk melted (this is going b..
Its been forever since I dealt with POTS lines. We have a client that needs FXS and FXO support. If memory serves correct we used the TDM400Pwith fxs_gs/fxo_gs. Whats the equivalent of that card today..
I have some dialplan code that is trying to convert 12 hour time with AM/PMto 24 hour format. The code has something like this:Exten =>2,1,ExecIf(${MATH(${HOUR..
Is there any way in Asterisk to have multiple forms of real time for the same object? For instance my main source for real time is MySQL. I want a fail over that if a mailbox is say not in the MySQL database for Asterisk to try via curl..
Disclaimer: I know I should not be using chan_sipThat being said I am trying to force Asterisk to use tcp by doing Dial(SIP/1234::::tcp@1.1.1.1//2.2.2.2)I want that Asterisk should send out the packet using TCP via the SIP proxy2.2.2.2. When I do a de..
We have a carrier that is using an upstream that we know has issues to a specific country. Is there any way in Asterisk that if we see a certain IPcome back in the SDP that we can then cancel the..
We have a box running Asterisk 11. A call comes in and the caller wants to use INFO (and the peer is set as INFO). We send the call out to a carrier were we specify rfc2833 and negotiate it correctly. In theory Asterisk should see the DTMF in rfc2..