similar to: Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

Are you using NFS? Any ODBC connections? On Tue, Apr 21, 2020 at 10:23 AM Stefan Viljoen <viljoens at verishare.co.za> wrote: > Hi all > > I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - > 32GB RAM. > When I approach about 320 channels, I -sometimes- get thousands of these > messages suddenly streamed in the CLI / Asterisk log: > >

Hi Joshua HTTP is used on in our setup on 127.0.0.1/mxml?<command> to send commands to the server, such as http://127.0.0.1/mxml?action=login&username=myuser&secret=thesecret to log in and then http://127.0.0.1/mxml?ActionID=123&Action=BlindTransfer&Channel=Channel&Context=local&Exten=123&Priority=1 etc. to control transfers, for example. ARI is not being

Hi Joshua No back-off, but I am caching the last 5000 results and and first hitting the cache to see if a recent command already provided the information I'm seeking for a particular request. I'll see if I can do some simulation and see if I'm effectively DDOSing the local HTTP interface. I'll have to see if I maybe have a resource leak in my code that makes the HTTP request,

Hi all I'm running an Asterisk 13.22.0 instance on Centos 7 - I7-8700 12 core HT with 16GB of RAM. The server maintains a total active call count of approx 285 calls with 440 channels at any one time. The totals never go below 200 calls concurrently active. For the kernel, top reports load levels at around 5.0 to 6.0 constantly. I'm having issues where at random intervals, the CLI is

Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script:

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

Hi all I need to pass a parameter in a thread-safe manner to the Queue pickup macro. This is to know when (and who) picked up an incoming call to a queue and log that to my back-office system with a CURL to a HTTP endpoint. However, the Queue application does not appear to allow passing of parameters to the called queue pickup macro. E. g. non-working code is: [queuetest] timeout = 60 retry =

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

Hi all I'm trying to limit the maximum concurrent calls on my Asterisk to try and mitigate another problem I posted about earlier. I've edited /etc/asterisk/asterisk.conf And uncommented this line, and put a value of 60 in there: maxcalls = 60 in an effort to limit my Asterisk to 60 simultaneous calls. I did a core reload in the CLI after doing that.

Hi all We've had a very strange failure on an Asterisk 1.8 install that has been running for about a year at a customer site. The physical hardware is fine, all other services off the Centos 6.5 server are running. Only Asterisk is not working... The first symptom was that no calls can be made over the SIP phones used with it, and no calls could be received over the SIP trunk connected to

Hi all I have a strange issue with 1.8.11.0 on a production Asterisk machine at our head office, and the same issue with a production machine at a branch office. Every now and then, on the head office machine, ODBC CEL and CDR logging will stop working. On examination in the CLI, Asterisk behaves as if the config files for ODBC in the /etc directory are just gone. Repeated tests have then

Hi All I'm using this extension to try and get Asterisk 1.8.11.0 to run a bash script: exten=>802,n,System(/bin/sh -f /root/wireless.sh) This file is -rwxr-xr-x 1 root root 171 Mar 2 16:23 wireless.sh e.g. root owns the file, and it has execute permissions for all users. Asterisk runs as root as well. Asterisk executes the command without any errors at max verbosity. The file

Hi all Asterisk 13 instance - I've got a situation in an agent queue that an agent will be talking to one person, then suddenly the same agent will be talking to another person who was talking to another agent. The calls do not switch around between the two agents, the "losing" agent will just suddenly have silence in his handset and the other agent will now be talking to

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would

On Wed, Jul 1, 2020 at 9:49 AM Stefan Viljoen <viljoens at verishare.co.za> wrote: > Hi all > > I'm running an Asterisk 13.22.0 instance on Centos 7 - I7-8700 12 core HT > with 16GB of RAM. > > The server maintains a total active call count of approx 285 calls with 440 > channels at any one time. The totals never go below 200 calls concurrently > active. >

On Wed, Jul 1, 2020 at 10:32 AM Stefan Viljoen <viljoens at verishare.co.za> wrote: > Hi Joshua > > HTTP is used on in our setup on > > 127.0.0.1/mxml?<command> > > to send commands to the server, such as > > http://127.0.0.1/mxml?action=login&username=myuser&secret=thesecret > > to log in and then > > >

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to